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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-05-14 14:44:20 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-05-14 14:44:20 +0000
commit0d0750b1b200901f2c49061aae1f5ade9953dde3 (patch)
tree6c8aba2263c629769b63b1a94c4c0ea3120f948a
parent82562a032ea5e706e5753d296b00331031a5c457 (diff)
parent192468bae9997f24dd33ec6309f9d56f8de86fc0 (diff)
Creating tag for the release of asterisk-1.6.0-beta9
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0-beta9@116245 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--.lastclean1
-rw-r--r--.version1
-rw-r--r--ChangeLog40542
-rw-r--r--apps/app_queue.c4
-rw-r--r--channels/chan_alsa.c3
-rw-r--r--channels/chan_iax2.c5
-rw-r--r--channels/chan_local.c25
-rw-r--r--channels/chan_sip.c4
-rw-r--r--channels/chan_zap.c7
-rw-r--r--include/asterisk/lock.h9
-rw-r--r--main/asterisk.c2
-rw-r--r--main/autoservice.c14
-rw-r--r--main/channel.c20
-rw-r--r--main/utils.c78
-rw-r--r--res/res_smdi.c2
15 files changed, 103 insertions, 40614 deletions
diff --git a/.lastclean b/.lastclean
deleted file mode 100644
index 8f92bfdd4..000000000
--- a/.lastclean
+++ /dev/null
@@ -1 +0,0 @@
-35
diff --git a/.version b/.version
deleted file mode 100644
index 64462a5a2..000000000
--- a/.version
+++ /dev/null
@@ -1 +0,0 @@
-1.6.0-beta9
diff --git a/ChangeLog b/ChangeLog
deleted file mode 100644
index 73b5a5939..000000000
--- a/ChangeLog
+++ /dev/null
@@ -1,40542 +0,0 @@
-2008-05-12 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta9 released.
-
-2008-05-11 03:27 +0000 [r115599-115601] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 115600 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115600 |
- mattf | 2008-05-10 22:23:05 -0500 (Sat, 10 May 2008) | 1 line Add
- Zap MTP2 support to chan_zap ........
-
- * channels/chan_zap.c, /: Merged revisions 115598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115598 |
- mattf | 2008-05-10 21:19:21 -0500 (Sat, 10 May 2008) | 1 line
- Open up audio channel when we get ACM on SS7 event ........
-
-2008-05-10 14:22 +0000 [r115597] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 115596 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115596 |
- tilghman | 2008-05-10 09:19:41 -0500 (Sat, 10 May 2008) | 2 lines
- Ensure that "calldate" is acceptable for a column name. ........
-
-2008-05-09 16:38 +0000 [r115581] Joshua Colp <jcolp@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 115580 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115580 | file | 2008-05-09 13:36:58 -0300 (Fri, 09 May 2008) |
- 10 lines Merged revisions 115579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2
- lines Improve res_ninit and res_ndestroy autoconf logic on the
- Darwin platform. ........ ................
-
-2008-05-08 19:21 +0000 [r115553-115570] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 115569 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115569 | russell | 2008-05-08 14:20:35 -0500
- (Thu, 08 May 2008) | 10 lines Merged revisions 115568 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 May 2008)
- | 2 lines Remove debug output. ........ ................
-
- * /, channels/chan_iax2.c: Merged revisions 115566 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115566 | russell | 2008-05-08 14:17:04 -0500
- (Thu, 08 May 2008) | 41 lines Merged revisions 115565 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r115565 | russell | 2008-05-08 14:15:25 -0500
- (Thu, 08 May 2008) | 33 lines Merged revisions 115564 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008)
- | 25 lines Fix a race condition that bbryant just found while
- doing some IAX2 testing. He was running Asterisk trunk running
- IAX2 calls through a few Asterisk boxes, however, the audio was
- extremely choppy. We looked at a packet trace and saw a storm of
- INVAL and VNAK frames being sent from one box to another. It
- turned out that what had happened was that one box tried to send
- a CONTROL frame before the 3 way handshake had completed. So,
- that frame did not include the destination call number, because
- it didn't have it yet. Part of our recent work for security
- issues included an additional check to ensure that frames that
- are supposed to include the destination call number have the
- correct one. This caused the frame to be rejected with an INVAL.
- The frame would get retransmitted for forever, rejected every
- time ... This race condition exists in all versions that got the
- security changes, in theory. However, it is really only likely
- that this would cause a problem in Asterisk trunk. There was a
- control frame being sent (SRCUPDATE) at the _very_ beginning of
- the call, which does not exist in 1.2 or 1.4. However, I am
- fixing all versions that could potentially be affected by the
- introduced race condition. These changes are what bbryant and I
- came up with to fix the issue. Instead of simply dropping control
- frames that get sent before the handshake is complete, the code
- attempts to wait a little while, since in most cases, the
- handshake will complete very quickly. If it doesn't complete
- after yielding for a little while, then the frame gets dropped.
- ........ ................ ................
-
- * /, channels/chan_sip.c: Merged revisions 115562 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115562 | russell | 2008-05-08 11:14:08 -0500 (Thu, 08 May 2008)
- | 11 lines Merged revisions 115561 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008)
- | 3 lines Don't give up on attempting an outbound registration if
- we receive a 408 Timeout. (closes issue #12323) ........
- ................
-
- * /, contrib/scripts/postgres_cdr.sql (removed): Merged revisions
- 115558 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115558 | russell | 2008-05-08 10:38:27 -0500 (Thu, 08 May 2008)
- | 11 lines Merged revisions 115557 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115557 | russell | 2008-05-08 10:37:49 -0500 (Thu, 08 May 2008)
- | 3 lines remove postgres_cdr.sql, as the CDR schema is in
- realtime_pgsql.sql, as well (closes issue #9676) ........
- ................
-
- * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115555 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115555 | russell | 2008-05-08 10:32:48 -0500
- (Thu, 08 May 2008) | 11 lines Merged revisions 115554 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115554 | russell | 2008-05-08 10:32:08 -0500 (Thu, 08 May 2008)
- | 3 lines Don't exit the script if Asterisk is not running.
- (closes issue #12611) ........ ................
-
- * main/pbx.c, /: Merged revisions 115552 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115552 | russell | 2008-05-08 10:26:49 -0500 (Thu, 08 May 2008)
- | 12 lines Merged revisions 115551 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115551 | russell | 2008-05-08 10:24:54 -0500 (Thu, 08 May 2008)
- | 4 lines Don't use a channel before checking for channel
- allocation failure. (closes issue #12609) Reported by: edantie
- ........ ................
-
-2008-05-08 15:08 +0000 [r115549] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 115548 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115548 |
- mattf | 2008-05-08 10:04:45 -0500 (Thu, 08 May 2008) | 1 line
- Remove unused code as well as demote an error message to a debug
- message ........
-
-2008-05-08 14:41 +0000 [r115538-115547] Russell Bryant <russell@digium.com>
-
- * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115546 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115546 | russell | 2008-05-08 09:41:12 -0500
- (Thu, 08 May 2008) | 12 lines Merged revisions 115545 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115545 | russell | 2008-05-08 09:40:53 -0500 (Thu, 08 May 2008)
- | 4 lines Use the same method for executing Asterisk as the rest
- of the script. (closes issue #12611) Reported by: b_plessis
- ........ ................
-
-2008-05-07 18:35 +0000 [r115514-115524] Russell Bryant <russell@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 115523 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r115523 | russell | 2008-05-07 13:33:50 -0500 (Wed, 07 May 2008)
- | 6 lines Only save a password if a username exists. (closes
- issue #12600) Reported By: suretec Patch by me ........
-
- * /, res/res_config_ldap.c: Merged revisions 115521 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r115521 | russell | 2008-05-07 13:30:12 -0500 (Wed, 07 May 2008)
- | 7 lines Use the default that the log output claims will be used
- for the basedn (closes issue #12599) Reported by: suretec
- Patches: 12599.patch uploaded by juggie (license 24) ........
-
- * /, channels/chan_h323.c: Merged revisions 115519 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r115519 | russell | 2008-05-07 13:24:51 -0500 (Wed, 07 May 2008)
- | 2 lines Let chan_h323 build in dev mode ........
-
- * /, include/asterisk/dlinkedlists.h (removed),
- channels/chan_iax2.c: Merged revisions 115513 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115513 | russell | 2008-05-07 12:28:19 -0500 (Wed, 07 May 2008)
- | 19 lines Merged revisions 115512 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r115512 | russell | 2008-05-07 11:24:09 -0500
- (Wed, 07 May 2008) | 11 lines Merged revisions 115511 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008)
- | 3 lines Remove remnants of dlinkedlists. I didn't actually use
- them in the final version of my IAX2 improvements. ........
- ................ ................
-
-2008-05-07 13:49 +0000 [r115510] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/asterisk.ldap-schema,
- contrib/scripts/asterisk.ldif, /: Merged revisions 115509 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r115509 | tilghman | 2008-05-07 08:49:15 -0500 (Wed, 07
- May 2008) | 2 lines Update typos in description fields (closes
- issue #12598) Reported by: suretec Patches:
- asterisk_schema_changes.patch uploaded by suretec (license 70)
- ........
-
-2008-05-06 19:56 +0000 [r115420-115424] Jason Parker <jparker@digium.com>
-
- * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115423
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115423 | qwell | 2008-05-06 14:55:45 -0500
- (Tue, 06 May 2008) | 23 lines Merged revisions 115422 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r115422 | qwell | 2008-05-06 14:55:29 -0500
- (Tue, 06 May 2008) | 15 lines Merged revisions 115421 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
- 7 lines read requires an argument on some non-bash shells (closes
- issue #12593) Reported by: bkruse Patches:
- getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
- ........ ................ ................
-
- * /, res/res_musiconhold.c: Merged revisions 115419 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115419 | qwell | 2008-05-06 14:38:44 -0500
- (Tue, 06 May 2008) | 15 lines Merged revisions 115418 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May 2008) |
- 7 lines Switch to using ast_random() rather than just rand().
- This does not fix the bug reported, but I believe it is correct.
- (from issue #12446) Patches: bug_12446.diff uploaded by snuffy
- (license 35) ........ ................
-
-2008-05-06 19:33 +0000 [r115417] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 115416 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115416 | tilghman | 2008-05-06 14:32:29 -0500 (Tue, 06 May 2008)
- | 10 lines Merged revisions 115415 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115415 | tilghman | 2008-05-06 14:31:39 -0500 (Tue, 06 May 2008)
- | 2 lines Don't print the terminating NUL. (Closes issue #12589)
- ........ ................
-
-2008-05-06 13:57 +0000 [r115343] Joshua Colp <jcolp@digium.com>
-
- * /, configure, configure.ac: Merged revisions 115342 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115342 | file | 2008-05-06 10:55:44 -0300 (Tue,
- 06 May 2008) | 10 lines Merged revisions 115341 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115341 | file | 2008-05-06 10:54:15 -0300 (Tue, 06 May 2008) | 2
- lines Add in missing argument. ........ ................
-
-2008-05-05 23:01 +0000 [r115335] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /, main/logger.c: Merged revisions 115334 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115334 | tilghman | 2008-05-05 18:00:31 -0500
- (Mon, 05 May 2008) | 15 lines Merged revisions 115333 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115333 | tilghman | 2008-05-05 17:50:31 -0500 (Mon, 05 May 2008)
- | 7 lines Separate verbose output from CLI output, by using a
- preamble. (closes issue #12402) Reported by: Corydon76 Patches:
- 20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76
- (license 14) 20080501__no_verbose_in_rx_output__1.4.diff.txt
- uploaded by Corydon76 (license 14) ........ ................
-
-2008-05-05 22:17 +0000 [r115331] Joshua Colp <jcolp@digium.com>
-
- * /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
- configure.ac: Merged revisions 115328 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115328 | file | 2008-05-05 19:13:57 -0300 (Mon, 05 May 2008) |
- 10 lines Merged revisions 115327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2
- lines Make sure that either the main speex library contains
- preprocess functions or that speexdsp does. If both fail then
- speex stuff can not be built. ........ ................
-
-2008-05-05 22:14 +0000 [r115330] Mark Michelson <mmichelson@digium.com>
-
- * main/config.c, /: Merged revisions 115329 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115329 |
- mmichelson | 2008-05-05 17:14:06 -0500 (Mon, 05 May 2008) | 15
- lines #execing the same file multiple times led to warning
- messages saying that the same file was being #included twice.
- This was due to the fact that #exec created a temporary file
- which was then #included. The name of the temporary file was the
- name of the #exec'd file, with the Unix timestamp and thread ID
- concatenated. The issue was that if multiple #exec statements of
- the same file were reached in the same second, then the result
- was that the temporary files would have duplicate names. To
- resolve this, the temporary file now has microsecond resolution
- for the timestamp portion. (closes issue #12574) Reported by:
- jmls Patches: 12574.patch uploaded by putnopvut (license 60)
- Tested by: jmls, putnopvut ........
-
-2008-05-05 21:44 +0000 [r115322] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 115321 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115321 | mmichelson | 2008-05-05 16:43:21 -0500 (Mon, 05 May
- 2008) | 21 lines Merged revisions 115320 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115320 | mmichelson | 2008-05-05 16:41:34 -0500 (Mon, 05 May
- 2008) | 13 lines Don't consider a caller "handled" until the
- caller is bridged with a queue member. There was too much of an
- opportunity for the member to hang up (either during a delay,
- announcement, or overly long agi) between the time that he
- answered the phone and the time when he actually was bridged with
- the caller. The consequence of this was that if the member hung
- up in that interval, then proper abandonment details would not be
- noted in the queue log if the caller were to hang up at any point
- after the member hangup. (closes issue #12561) Reported by:
- ablackthorn ........ ................
-
-2008-05-05 20:28 +0000 [r115316] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 115315 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r115315 | russell | 2008-05-05 15:28:17 -0500 (Mon, 05 May 2008)
- | 2 lines Remove my rant, since I have now replaced the rant with
- code. ........
-
-2008-05-05 19:58 +0000 [r115310] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h, /: Merged revisions 115309 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115309 | tilghman | 2008-05-05 14:57:28 -0500
- (Mon, 05 May 2008) | 10 lines Merged revisions 115308 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115308 | tilghman | 2008-05-05 14:55:55 -0500 (Mon, 05 May 2008)
- | 2 lines Err, the documentation on the return value of
- ast_odbc_backslash_is_escape is exactly backwards. ........
- ................
-
-2008-05-05 19:50 +0000 [r115306] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 115305 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115305 | russell | 2008-05-05 14:50:24 -0500 (Mon, 05 May 2008)
- | 13 lines Merged revisions 115304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008)
- | 5 lines Avoid putting opaque="" in Digest authentication. This
- patch came from switchvox. It fixes authentication with Primus in
- Canada, and has been in use for a very long time without causing
- problems with any other providers. (closes issue AST-36) ........
- ................
-
-2008-05-05 19:43 +0000 [r115303] Tilghman Lesher <tlesher@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 115302 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115302 |
- tilghman | 2008-05-05 14:42:36 -0500 (Mon, 05 May 2008) | 2 lines
- Note change for ExecIf syntax (caught by jmls on IRC) ........
-
-2008-05-05 10:55 +0000 [r115289] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 115288 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r115288 |
- kpfleming | 2008-05-05 05:55:09 -0500 (Mon, 05 May 2008) | 2
- lines clarify wording ........
-
-2008-05-05 03:26 +0000 [r115287] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
- contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk, /,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.slackware.asterisk: Merged revisions 115286 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115286 | tilghman | 2008-05-04 22:25:35 -0500
- (Sun, 04 May 2008) | 15 lines Merged revisions 115285 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115285 | tilghman | 2008-05-04 22:22:25 -0500 (Sun, 04 May 2008)
- | 7 lines When starting Asterisk, bug out if Asterisk is already
- running. (closes issue #12525) Reported by: explidous Patches:
- 20080428__bug12525.diff.txt uploaded by Corydon76 (license 14)
- Tested by: mvanbaak ........ ................
-
-2008-05-04 02:12 +0000 [r115278-115284] Joshua Colp <jcolp@digium.com>
-
- * /, configure, acinclude.m4: Merged revisions 115283 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115283 | file | 2008-05-03 23:11:01 -0300 (Sat,
- 03 May 2008) | 10 lines Merged revisions 115282 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115282 | file | 2008-05-03 23:09:44 -0300 (Sat, 03 May 2008) | 2
- lines Expand the test function for GCC attributes so that more
- complex attributes are properly recognized. ........
- ................
-
- * /, include/asterisk/compiler.h: Merged revisions 115280 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115280 | file | 2008-05-03 22:52:00 -0300 (Sat,
- 03 May 2008) | 10 lines Merged revisions 115279 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115279 | file | 2008-05-03 22:50:59 -0300 (Sat, 03 May 2008) | 2
- lines For my next trick I will make these work with what our
- autoconf header file gives us. ........ ................
-
- * /, configure, acinclude.m4: Merged revisions 115277 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115277 | file | 2008-05-03 22:45:21 -0300 (Sat,
- 03 May 2008) | 10 lines Merged revisions 115276 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115276 | file | 2008-05-03 22:43:26 -0300 (Sat, 03 May 2008) | 2
- lines Treat warnings as errors when checking if a GCC attribute
- exists. We have to do this as GCC will just ignore the attribute
- and pop up a warning, it won't actually fail to compile. ........
- ................
-
-2008-05-03 04:25 +0000 [r115269-115275] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /: block voicemail mwi notification subscriptions taskprocessor
-
- * /: block pbx taskprocessor
-
- * /: block app_queue taskprocessor
-
- * /: blocked taskprocessors
-
-2008-05-02 14:55 +0000 [r115198-115200] Mark Michelson <mmichelson@digium.com>
-
- * /, include/asterisk/sched.h: Merged revisions 115197 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115197 | mmichelson | 2008-05-02 09:28:55 -0500
- (Fri, 02 May 2008) | 14 lines Merged revisions 115196 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, 02 May
- 2008) | 6 lines Clarify a comment that was, well, just wrong. It
- turns out that ignoring the way that macros expand. Instead, I
- have clarified in the comment why the macro will work even if the
- scheduler id for the task to be deleted changes during the
- execution of the macro. ........ ................
-
-2008-05-02 02:57 +0000 [r115107-115160] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h, /: Merged revisions 115104 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r115104 | tilghman | 2008-05-01 18:21:13 -0500
- (Thu, 01 May 2008) | 10 lines Merged revisions 115102 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115102 | tilghman | 2008-05-01 18:20:25 -0500 (Thu, 01 May 2008)
- | 2 lines Change the comment of deprecated to an actual compiler
- deprecation ........ ................
-
-2008-05-01 19:01 +0000 [r115020] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/utils.c: Merged revisions 115018 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115018 | tilghman | 2008-05-01 14:00:18 -0500 (Thu, 01 May 2008)
- | 14 lines Merged revisions 115017 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115017 | tilghman | 2008-05-01 13:59:08 -0500 (Thu, 01 May 2008)
- | 6 lines '#' is another reserved character for URIs that also
- needs to be escaped. (closes issue #10543) Reported by: blitzrage
- Patches: 20080418__bug10543.diff.txt uploaded by Corydon76
- (license 14) ........ ................
-
-2008-05-01 17:28 +0000 [r114932] Russell Bryant <russell@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 114931 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114931 |
- russell | 2008-05-01 12:28:25 -0500 (Thu, 01 May 2008) | 4 lines
- Clarify the deprecation notice about Macro() to note that it will
- not be removed for the sake of backwards compatibility, since it
- is a non-trivial task to convert existing large dialplans that
- depend on Macro() to use GoSub(), instead. ........
-
-2008-05-01 16:52 +0000 [r114923] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 114922 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114922 |
- qwell | 2008-05-01 11:49:24 -0500 (Thu, 01 May 2008) | 10 lines
- Allow dringXrange to properly default to 10, as was done in 1.4.
- dringXrange is a new feature that was added, and it attempted to
- default, but only when the option was specified. (closes issue
- #12536) Reported by: bjm Patches: 12536-dringXrange.diff uploaded
- by qwell (license 4) Tested by: bjm ........
-
-2008-04-30 20:20 +0000 [r114909] Russell Bryant <russell@digium.com>
-
- * include/asterisk/dlinkedlists.h (added): Add the dlinkedlists
- implementation from trunk
-
-2008-04-30 20:17 +0000 [r114907-114908] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Make 1.6.0 compile
-
-2008-04-30 17:06 +0000 [r114900] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 114899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114899 | oej | 2008-04-30 18:55:49 +0200 (Ons, 30 Apr 2008) | 15
- lines Merged revisions 114890 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7
- lines Don't crash on bad SIP replys. Fix created in Huntsville
- together with Mark M (putnopvut) (closes issue #12363) Reported
- by: jvandal Tested by: putnopvut, oej ........ ................
-
-2008-04-30 16:41 +0000 [r114893] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_console.c, channels/chan_iax2.c: Merged
- revisions 114892 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008)
- | 36 lines Merged revisions 114891 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008)
- | 28 lines Merge changes from team/russell/iax2_find_callno and
- iax2_find_callno_1.4 These changes address a critical performance
- issue introduced in the latest release. The fix for the latest
- security issue included a change that made Asterisk randomly
- choose call numbers to make them more difficult to guess by
- attackers. However, due to some inefficient (this is by far, an
- understatement) code, when Asterisk chose high call numbers,
- chan_iax2 became unusable after just a small number of calls. On
- a small embedded platform, it would not be able to handle a
- single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
- more than about 16 IAX2 channels. Ouch. These changes address
- some performance issues of the find_callno() function that have
- bothered me for a very long time. On every incoming media frame,
- it iterated through every possible call number trying to find a
- matching active call. This involved a mutex lock and unlock for
- each call number checked. So, if the random call number chosen
- was 20000, then every media frame would cause 20000 locks and
- unlocks. Previously, this problem was not as obvious since
- Asterisk always chose the lowest call number it could. A second
- container for IAX2 pvt structs has been added. It is an astobj2
- hash table. When we know the remote side's call number, the pvt
- goes into the hash table with a hash value of the remote side's
- call number. Then, lookups for incoming media frames are a very
- fast hash lookup instead of an absolutely insane array traversal.
- In a quick test, I was able to get more than 3600% more IAX2
- channels on my machine with these changes. ........
- ................
-
-2008-04-30 16:15 +0000 [r114889] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_console.c: Merged revisions 114888 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114888 | jpeeler | 2008-04-30 11:14:43 -0500 (Wed, 30 Apr 2008)
- | 3 lines Fixes a bug where if a stream monitor thread was not
- created (caused from failure of opening or starting the stream)
- pthread_cancel was called with an invalid thread ID. ........
-
-2008-04-30 14:55 +0000 [r114877-114886] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/iax2.h, channels/chan_iax2.c: Merged revisions 114884
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114884 | kpfleming | 2008-04-30 09:49:51 -0500
- (Wed, 30 Apr 2008) | 10 lines Merged revisions 114880 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr
- 2008) | 2 lines use the ARRAY_LEN macro for indexing through the
- iaxs/iaxsl arrays so that the size of the arrays can be adjusted
- in one place, and change the size of the arrays from 32768 calls
- to 2048 calls when LOW_MEMORY is defined ........
- ................
-
- * /, Makefile.rules: Merged revisions 114876 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114876 | kpfleming | 2008-04-30 07:15:43 -0500 (Wed, 30 Apr
- 2008) | 10 lines Merged revisions 114875 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114875 | kpfleming | 2008-04-30 07:14:07 -0500 (Wed, 30 Apr
- 2008) | 2 lines pay attention to *all* header files for
- dependency tracking, not just the local ones (inspired by r578 of
- asterisk-addons by tilghman) ........ ................
-
-2008-04-29 22:55 +0000 [r114867] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/iax2-provision.c: Merged revisions 114866 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r114866 | jpeeler | 2008-04-29 17:54:14 -0500 (Tue, 29
- Apr 2008) | 2 lines Fixes a problem where all the templates were
- marked as dead no matter what. The templates should only be
- marked as dead if a configuration file has been successfully
- loaded and has changes. Bug found while making API documentation
- for 1.6.0. ........
-
-2008-04-29 21:09 +0000 [r114850-114858] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 114849 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114849 | mmichelson | 2008-04-29 14:42:04 -0500 (Tue, 29 Apr
- 2008) | 22 lines Merged revisions 114848 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr
- 2008) | 14 lines Use the MACRO_CONTEXT and MACRO_EXTEN channel
- variables instead of the channel's macrocontext and macroexten
- fields. This is needed because if macros are daisy-chained, the
- incorrect context and extension are placed on the new channel. I
- also added locking to the channel prior to accessing these
- variables as noted in trunk's janitor project file. (closes issue
- #12549) Reported by: darren1713 Patches:
- app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
- (with modifications from me) Tested by: putnopvut ........
- ................
-
-2008-04-29 17:11 +0000 [r114831] Jason Parker <jparker@digium.com>
-
- * res/res_config_pgsql.c, /: Merged revisions 114830 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114830 | qwell | 2008-04-29 12:10:55 -0500
- (Tue, 29 Apr 2008) | 9 lines Merged revisions 114829 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr
- 2008) | 1 line Change warning message to debug, since there are
- cases where 0 results is perfectly fine. ........
- ................
-
-2008-04-29 12:55 +0000 [r114825] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114824
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114824 | kpfleming | 2008-04-29 07:54:31 -0500
- (Tue, 29 Apr 2008) | 18 lines Merged revisions 114823 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r114823 | kpfleming | 2008-04-29 07:53:12 -0500
- (Tue, 29 Apr 2008) | 10 lines Merged revisions 114822 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
- 2008) | 2 lines stop script from appending source code if run
- multiple times ........ ................ ................
-
-2008-04-28 17:04 +0000 [r114777] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 114776 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114776 |
- mattf | 2008-04-28 12:00:38 -0500 (Mon, 28 Apr 2008) | 1 line Fix
- deadlock issue in chan_zap with libss7 due to channel variables
- being set with the channel pvt lock being held. #12512 ........
-
-2008-04-28 13:44 +0000 [r114714] Joshua Colp <jcolp@digium.com>
-
- * /, configure, configure.ac: Merged revisions 114713 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114713 | file | 2008-04-28 10:42:13 -0300 (Mon, 28 Apr 2008) | 2
- lines Update autoconf logic with latest API change for libss7.
- ........
-
-2008-04-28 04:54 +0000 [r114707-114710] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
- revisions 114709 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008)
- | 13 lines Merged revisions 114708 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008)
- | 5 lines When modules are embedded, they take on a different
- name, without the ".so" extension. Specifically check for this
- name, when we're checking if a module is loaded. (Closes issue
- #12534) ........ ................
-
-2008-04-27 15:20 +0000 [r114701] Michiel van Baak <michiel@vanbaak.info>
-
- * /, channels/chan_skinny.c: Merged revisions 114700 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk Merged to 1.6
- because it fixes a crash. ........ r114700 | mvanbaak |
- 2008-04-27 17:17:18 +0200 (Sun, 27 Apr 2008) | 8 lines Make MWI
- in chan_skinny event based modeled after chan_zap and chan_mgcp.
- (closes issue #12214) Reported by: DEA Patches:
- chan_skinny-vm-events-v3.txt uploaded by DEA (license 3) Tested
- by: DEA and me ........
-
-2008-04-27 01:30 +0000 [r114697] Sean Bright <sean.bright@gmail.com>
-
- * /, configure, configure.ac: Merged revisions 114696 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114696 | seanbright | 2008-04-26 21:28:32 -0400
- (Sat, 26 Apr 2008) | 13 lines Merged revisions 114695 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, 26 Apr
- 2008) | 5 lines When we don't explicitly pass a path to the
- --with-tds configure option, we may end up finding tds.h in
- /usr/local/include instead of /usr/include. If this happens, the
- grep that looks for the version (from tdsver.h) will fail and
- we'll have some problems during the build. ........
- ................
-
-2008-04-26 15:09 +0000 [r114684-114693] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/vmail.cgi: Merged revisions 114690 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114690 | tilghman | 2008-04-26 08:17:19 -0500
- (Sat, 26 Apr 2008) | 14 lines Merged revisions 114689 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114689 | tilghman | 2008-04-26 08:15:21 -0500 (Sat, 26 Apr 2008)
- | 6 lines Clicking forward without selecting a message leaves an
- errant .lock file. (closes issue #12528) Reported by: pukepail
- Patches: patch.diff uploaded by pukepail (license 431) ........
- ................
-
-2008-04-25 22:05 +0000 [r114671-114677] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_lua.c: Merged revisions 114676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114676 |
- russell | 2008-04-25 17:04:46 -0500 (Fri, 25 Apr 2008) | 7 lines
- Lock the channel around datastore access (closes issue #12527)
- Reported by: mnicholson Patches: pbx_lua4.diff uploaded by
- mnicholson (license 96) ........
-
- * /, channels/chan_iax2.c: Merged revisions 114674 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114674 | russell | 2008-04-25 17:00:35 -0500
- (Fri, 25 Apr 2008) | 11 lines Merged revisions 114673 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008)
- | 3 lines Use consistent logic for checking to see if a call
- number has been chosen yet. Also, remove some redundant logic I
- recently added in a fix. ........ ................
-
-2008-04-25 19:34 +0000 [r114664] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 114663 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114663 | mmichelson | 2008-04-25 14:33:27 -0500 (Fri, 25 Apr
- 2008) | 12 lines Merged revisions 114662 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114662 | mmichelson | 2008-04-25 14:32:02 -0500 (Fri, 25 Apr
- 2008) | 4 lines Move the unlock of the spyee channel to outside
- the start_spying() function so that the channel is not unlocked
- twice when using whisper mode. ........ ................
-
-2008-04-25 16:26 +0000 [r114652] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 114651 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114651 | mmichelson | 2008-04-25 11:25:17 -0500 (Fri, 25 Apr
- 2008) | 4 lines Fix a memory leak and protect against potential
- dereferences of a NULL pointer. ........
-
-2008-04-24 22:14 +0000 [r114636] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114635 |
- file | 2008-04-24 19:11:46 -0300 (Thu, 24 Apr 2008) | 4 lines Hey
- look, it builds. (closes issue #12519) Reported by: falves11
- ........
-
-2008-04-24 21:36 +0000 [r114626-114634] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114633 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr
- 2008) | 19 lines Merged revisions 114632 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr
- 2008) | 11 lines Re-invite RTP during a masquerade so that, for
- instance, an AMI redirect of two channels which are natively
- bridged will preserve audio on both channels. This prevents a
- problem with Asterisk not re-inviting due to one of the channels
- having being a zombie. (closes issue #12513) Reported by:
- mneuhauser Patches:
- asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
- mneuhauser (license 425) ........ ................
-
- * /, apps/app_queue.c: Merged revisions 114629 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114629 | mmichelson | 2008-04-24 15:43:52 -0500 (Thu, 24 Apr
- 2008) | 16 lines Merged revisions 114628 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114628 | mmichelson | 2008-04-24 15:43:03 -0500 (Thu, 24 Apr
- 2008) | 8 lines Output of channel variables when
- eventwhencalled=vars was set was being truncated two characters.
- This patch corrects the problem. (closes issue #12493) Reported
- by: davidw ........ ................
-
- * channels/chan_local.c, /: Merged revisions 114625 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114625 | mmichelson | 2008-04-24 15:06:06 -0500
- (Thu, 24 Apr 2008) | 18 lines Merged revisions 114624 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr
- 2008) | 10 lines Resolve a deadlock in chan_local by releasing
- the channel lock temporarily. (closes issue #11712) Reported by:
- callguy Patches: 11712.patch uploaded by putnopvut (license 60)
- Tested by: acunningham ........ ................
-
-2008-04-24 19:55 +0000 [r114619-114623] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 114622 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114622 | tilghman | 2008-04-24 14:54:57 -0500
- (Thu, 24 Apr 2008) | 12 lines Merged revisions 114621 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008)
- | 4 lines Ensure that when we set the accountcode, it actually
- shows up in the CDR. (Fix for AMI Originate) (Closes issue
- #12007) ........ ................
-
- * /, apps/app_meetme.c: Merged revisions 114617 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114617 |
- tilghman | 2008-04-24 14:24:31 -0500 (Thu, 24 Apr 2008) | 6 lines
- Fix DST calculation, and fix bug in calculation of whether conf
- has started yet or not (Closes issue #12292) Reported by: DEA
- Patches: app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license
- 3) ........
-
-2008-04-24 16:48 +0000 [r114613] Jason Parker <jparker@digium.com>
-
- * channels/chan_misdn.c, /: Merged revisions 114612 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114612 | qwell | 2008-04-24 11:47:01 -0500
- (Thu, 24 Apr 2008) | 17 lines Merged revisions 51989 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #12496) Reported by: daniele Patches:
- misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471)
- Tested by: daniele Technically, I didn't use the patch above
- except to find out what revision to merge - but it's the same
- thing as this revision. ........ r51989 | crichter | 2007-01-24
- 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line added fix from #8899
- ........ ................
-
-2008-04-24 15:57 +0000 [r114610] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 114609 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114609 | russell | 2008-04-24 10:56:55 -0500
- (Thu, 24 Apr 2008) | 12 lines Merged revisions 114608 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008)
- | 4 lines Fix a silly mistake in a change I made yesterday that
- caused chan_iax2 to blow up very quickly. (issue #12515) ........
- ................
-
-2008-04-24 15:00 +0000 [r114607] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Merged revisions 114606 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114606 | oej | 2008-04-24 16:59:05 +0200 (Tor, 24 Apr 2008) | 11
- lines Merged revisions 114603 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3
- lines Only have one max-forwards header in outbound REFERs.
- Discovered in the Asterisk SIP Masterclass in Orlando. Thanks
- Joe! ........ ................
-
-2008-04-24 14:56 +0000 [r114599-114605] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114604 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114604 |
- russell | 2008-04-24 09:55:21 -0500 (Thu, 24 Apr 2008) | 3 lines
- Change a verbose message to debug. (closes issue #12514) ........
-
- * /, main/http.c: Merged revisions 114601 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114601 | russell | 2008-04-23 17:53:20 -0500 (Wed, 23 Apr 2008)
- | 14 lines Merged revisions 114600 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008)
- | 6 lines Improve some broken cookie parsing code. Previously,
- manager login over HTTP would only work if the mansession_id
- cookie was first. Now, the code builds a list of all of the
- cookies in the Cookie header. This fixes a problem observed by
- users of the Asterisk GUI. (closes AST-20) ........
- ................
-
- * apps/app_chanspy.c, /: Merged revisions 114598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114598 | russell | 2008-04-23 15:53:05 -0500 (Wed, 23 Apr 2008)
- | 18 lines Merged revisions 114597 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114597 | russell | 2008-04-23 15:49:18 -0500 (Wed, 23 Apr 2008)
- | 10 lines Fix an issue that caused getting the correct next
- channel to not always work. Also, remove setting the amount of
- time to wait for a digit from 5 seconds back down to 1/10 of a
- second. I believe this was so the beep didn't get played over and
- over really fast, but a while back I put in another fix for that
- issue. (closes issue #12498) Reported by: jsmith Patches:
- app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license
- 15) ........ ................
-
-2008-04-23 18:34 +0000 [r114596] Jason Parker <jparker@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 114595 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114595 | qwell | 2008-04-23 13:33:28 -0500
- (Wed, 23 Apr 2008) | 16 lines Merged revisions 114594 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114594 | qwell | 2008-04-23 13:28:44 -0500 (Wed, 23 Apr 2008) |
- 8 lines Fix reload/unload for res_musiconhold module. (closes
- issue #11575) Reported by: sunder Patches: M11575_14_rev3.diff
- uploaded by junky (license 177) bug11575_trunk.diff.txt uploaded
- by jamesgolovich (license 176) ........ ................
-
-2008-04-23 18:01 +0000 [r114589-114593] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /, include/asterisk/manager.h: Merged revisions
- 114592 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114592 | russell | 2008-04-23 13:01:00 -0500 (Wed, 23 Apr 2008)
- | 13 lines Merged revisions 114591 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008)
- | 5 lines Store the manager session ID explicitly as 4 byte ID
- instead of a ulong. The mansession_id cookie is coded to be
- limited to 8 characters of hex, and this could break logins from
- 64-bit machines in some cases. (inspired by AST-20) ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 114588 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114588 | russell | 2008-04-23 12:18:29 -0500
- (Wed, 23 Apr 2008) | 10 lines Merged revisions 114587 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008)
- | 2 lines Fix find_callno_locked() to actually return the callno
- locked in some more cases. ........ ................
-
-2008-04-23 16:57 +0000 [r114586] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Merged revisions 114585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114585 | oej | 2008-04-23 18:53:34 +0200 (Ons, 23 Apr 2008) | 10
- lines Merged revisions 114584 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2
- lines Add 502 support for both directions, not only one... (see
- r114571) ........ ................
-
-2008-04-23 14:56 +0000 [r114581] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, /: Merged revisions 114580 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114580 | file | 2008-04-23 11:55:03 -0300 (Wed, 23 Apr 2008) |
- 12 lines Merged revisions 114579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4
- lines Instead of stopping dialplan execution when SayNumber
- attempts to say a large number that it can not print out a
- message informing the user and continue on. (closes issue #12502)
- Reported by: bcnit ........ ................
-
-2008-04-23 01:00 +0000 [r114576-114578] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 114575 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114575 | mmichelson | 2008-04-22 19:40:30 -0500 (Tue, 22 Apr
- 2008) | 10 lines Round 1 of IMAP_STORAGE-related app_voicemail
- changes This makes IMAP_STORAGE include the proper headers if you
- have specified the "system" option for --with-imap when running
- the configure script and your IMAP-related headers exist in
- /usr/include/c-client. This change is due to a hasty merge of a
- 1.4 change I made. ........
-
-2008-04-22 23:59 +0000 [r114573] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114572 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114572 | tilghman | 2008-04-22 18:58:19 -0500 (Tue, 22 Apr 2008)
- | 10 lines Merged revisions 114571 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008)
- | 2 lines Treat a 502 just like a 503, when it comes to
- processing a response code ........ ................
-
-2008-04-22 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta8 released.
-
-2008-04-22 22:18 +0000 [r114560] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 114559 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114559 | russell | 2008-04-22 17:17:31 -0500
- (Tue, 22 Apr 2008) | 13 lines Merged revisions 114558 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008)
- | 5 lines When we receive a full frame that is supposed to
- contain our call number, ensure that it has the correct one.
- (closes issue #10078) (AST-2008-006) ........ ................
-
-2008-04-22 22:04 +0000 [r114556] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 114553 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114553 |
- murf | 2008-04-22 15:57:57 -0600 (Tue, 22 Apr 2008) | 14 lines
- (closes issue #12469) Reported by: triccyx I had a bit a problem
- reproducing this in my setup (trying not to disturb my other
- stuff) but finally, I got it. The problem appears to be that the
- extension is being added in replace mode, which kinda assumes
- that the pattern trie has been formed, when in fact, in this
- case, it was not. The checks being done are not nec. when the
- tree is not yet formed, as changes like this will be summarized
- when the trie is formed in the future. I tested the fix, and the
- crash no longer happens. Feel free to open the bug again if this
- fix doesn't cure the problem. ........
-
-2008-04-22 21:16 +0000 [r114544-114552] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 114548 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114548 |
- russell | 2008-04-22 15:25:56 -0500 (Tue, 22 Apr 2008) | 2 lines
- re-add a fix that got lost with a recent change ........
-
-2008-04-22 18:14 +0000 [r114541] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, /, include/asterisk/pbx.h, apps/app_queue.c: Merged
- revisions 114540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114540 |
- qwell | 2008-04-22 13:14:09 -0500 (Tue, 22 Apr 2008) | 8 lines
- Allow setqueuevar=yes (et al) to work, after changes to
- pbx_builtin_setvar() (closes issue #12490) Reported by: bcnit
- Patches: 12490-queuevars-3.diff uploaded by qwell (license 4)
- Tested by: qwell ........
-
-2008-04-22 18:06 +0000 [r114534-114539] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 114538 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114538 | russell | 2008-04-22 13:04:39 -0500
- (Tue, 22 Apr 2008) | 17 lines Merged revisions 114537 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008)
- | 9 lines If the dial string passed to the call channel callback
- does not indicate an extension, then consider the extension on
- the channel before falling back to the default. (closes issue
- #12479) Reported by: darren1713 Patches:
- exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
- 116) ........ ................
-
-2008-04-22 15:46 +0000 [r114524-114528] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 114527 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114527 |
- russell | 2008-04-22 10:46:01 -0500 (Tue, 22 Apr 2008) | 8 lines
- Correct action_ping() and action_events() with regards to Manager
- 1.1 documentation. Also, fix a bug in xml_translate(). (closes
- issue #11649) Reported by: ys Patches: trunk_manager.c.diff
- uploaded by ys (license 281) ........
-
-2008-04-21 20:23 +0000 [r114422] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 114389 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114389 |
- mattf | 2008-04-21 13:44:35 -0500 (Mon, 21 Apr 2008) | 1 line Add
- support for generic name transmission (#12484) on SS7 in chan_zap
- ........
-
-2008-04-21 15:38 +0000 [r114328] Jeff Peeler <jpeeler@digium.com>
-
- * /, apps/app_authenticate.c: Merged revisions 114327 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114327 | jpeeler | 2008-04-21 10:34:37 -0500 (Mon, 21 Apr 2008)
- | 2 lines This removes an invalid warning message for an
- incorrectly entered pin, but more importantly removes an
- inapplicable check. If the first argument passed to
- app_authenticate does not contain a '/', the argument should be
- treated as the sole fixed "password" to match against and that is
- all. (Previous behavior was attempting to open a file based on
- the pin.) ........
-
-2008-04-21 14:42 +0000 [r114321-114324] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114323 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114323 | file | 2008-04-21 11:40:33 -0300 (Mon, 21 Apr 2008) |
- 12 lines Merged revisions 114322 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4
- lines Only drop audio if we receive it without a progress
- indication. We allow other frames through such as DTMF because
- they may be needed to complete the call. (closes issue #12440)
- Reported by: aragon ........ ................
-
- * /, res/res_config_ldap.c: Merged revisions 114320 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114320 | file | 2008-04-21 11:34:06 -0300 (Mon, 21 Apr 2008) | 6
- lines Only print out the error message if ldap_modify_ext_s
- actually returns an error, and not success. (closes issue #12438)
- Reported by: gservat Patches: res_config_ldap.c-patch-code
- uploaded by gservat (license 466) ........
-
-2008-04-19 17:00 +0000 [r114304] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: SS7:Added - Generic Name / Access Transport
- / Redirecting Number handling. #12425
-
-2008-04-18 21:51 +0000 [r114277-114286] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 114285 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114285 | russell | 2008-04-18 16:51:05 -0500 (Fri, 18 Apr 2008)
- | 10 lines Merged revisions 114284 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114284 | russell | 2008-04-18 16:48:06 -0500 (Fri, 18 Apr 2008)
- | 2 lines Don't destroy a manager session if poll() returns an
- error of EAGAIN. ........ ................
-
- * Makefile, /: Merged revisions 114279 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114279 | russell | 2008-04-18 15:01:47 -0500 (Fri, 18 Apr 2008)
- | 10 lines Merged revisions 114278 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114278 | russell | 2008-04-18 15:01:09 -0500 (Fri, 18 Apr 2008)
- | 2 lines ensure directories are created before we try to install
- stuff into them ........ ................
-
- * Makefile, /: Merged revisions 114276 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114276 | russell | 2008-04-18 14:59:17 -0500 (Fri, 18 Apr 2008)
- | 10 lines Merged revisions 114275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114275 | russell | 2008-04-18 14:58:55 -0500 (Fri, 18 Apr 2008)
- | 2 lines SUBDIRS_INSTALL is already listed as a subtarget for
- bininstall ........ ................
-
-2008-04-18 19:36 +0000 [r114262-114272] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_unistim.c, /: Merged revisions 114271 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114271 | file | 2008-04-18 16:35:33 -0300 (Fri, 18 Apr 2008) | 4
- lines Make sure ADSI is marked as unavailable on Unistim channels
- so voicemail does not try to do some ADSI jazz. (closes issue
- #12460) Reported by: PerryB ........
-
-2008-04-18 18:04 +0000 [r114260] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c, /, main/callerid.c: Merged revisions 114259
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114259 | mmichelson | 2008-04-18 13:03:06 -0500
- (Fri, 18 Apr 2008) | 14 lines Merged revisions 114257 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr
- 2008) | 6 lines Clearing up error messages so they make a bit
- more sense. Also removing a redundant error message. Issue AST-15
- ........ ................
-
-2008-04-18 16:12 +0000 [r114255] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 114254 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114254 | file | 2008-04-18 13:11:27 -0300 (Fri, 18 Apr 2008) | 4
- lines If the parsing of the config file fails make sure we unlock
- ldap_lock. (closes issue #12477) Reported by: IgorG ........
-
-2008-04-18 13:40 +0000 [r114247] Sean Bright <sean.bright@gmail.com>
-
- * channels/chan_sip.c: Merged revisions 114246 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114246 | seanbright | 2008-04-18 09:38:07 -0400 (Fri, 18 Apr
- 2008) | 9 lines Merged revisions 114245 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr
- 2008) | 1 line Only complete the SIP channel name once for 'sip
- show channel <channel>' ........ ................
-
-2008-04-18 06:54 +0000 [r114244] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_setcallerid.c, /: Merged revisions 114243 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114243 | tilghman | 2008-04-18 01:53:47 -0500
- (Fri, 18 Apr 2008) | 11 lines Merged revisions 114242 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114242 | tilghman | 2008-04-18 01:49:16 -0500 (Fri, 18 Apr 2008)
- | 3 lines For consistency sake, ensure that the values that
- ${CALLINGPRES} returns are valid as an input to SetCallingPres.
- (Closes issue #12472) ........ ................
-
-2008-04-17 23:09 +0000 [r114232-114241] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114151 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114151 | oej | 2008-04-15 15:39:29 -0500 (Tue, 15 Apr 2008) | 10
- lines Merged revisions 114148 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2
- lines Handle subscribe queues in all situations... Thanks to
- festr_ on irc for telling me about this bug. ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 114150 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114150 |
- oej | 2008-04-15 15:31:08 -0500 (Tue, 15 Apr 2008) | 2 lines
- Adding chanvar to SIPPEER from 1.4 branch ........
-
- * main/autoservice.c, /: Merged revisions 114233 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114233 | russell | 2008-04-17 17:24:00 -0500 (Thu, 17 Apr 2008)
- | 14 lines Merged revisions 114230 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114230 | russell | 2008-04-17 17:15:43 -0500 (Thu, 17 Apr 2008)
- | 6 lines Remove redundant safety net. The check for the
- autoservice channel list state accomplishes the same goal in a
- better way. (issue #12470) Reported By: atis ........
- ................
-
-2008-04-17 21:05 +0000 [r114228] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 114227 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114227 | mmichelson | 2008-04-17 16:04:40 -0500 (Thu, 17 Apr
- 2008) | 17 lines Merged revisions 114226 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114226 | mmichelson | 2008-04-17 16:03:29 -0500 (Thu, 17 Apr
- 2008) | 9 lines Declaration of the peer channel in this scope was
- making it so the peer variable defined in the outer scope was
- never set properly, therefore making iterating through the
- channel list always restart from the beginning. This bug would
- have affected anyone who called chanspy without specifying a
- first argument. (closes issue #12461) Reported by: stever28
- ........ ................
-
-2008-04-17 16:51 +0000 [r114210-114213] Mark Michelson <mmichelson@digium.com>
-
- * main/dsp.c, main/frame.c, /, include/asterisk/dsp.h,
- include/asterisk/frame.h: Merged revisions 114208 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114208 | mmichelson | 2008-04-17 11:40:12 -0500
- (Thu, 17 Apr 2008) | 20 lines Merged revisions 114207 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr
- 2008) | 12 lines It was possible for a reference to a frame which
- was part of a freed DSP to still be referenced, leading to memory
- corruption and eventual crashes. This code change ensures that
- the dsp is freed when we are finished with the frame. This change
- is very similar to a change Russell made with translators back a
- month or so ago. (closes issue #11999) Reported by: destiny6628
- Patches: 11999.patch uploaded by putnopvut (license 60) Tested
- by: destiny6628, victoryure ........ ................
-
-2008-04-17 16:26 +0000 [r114206] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 114205 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114205 | russell | 2008-04-17 11:25:29 -0500 (Thu, 17 Apr 2008)
- | 11 lines Merged revisions 114204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114204 | russell | 2008-04-17 11:23:45 -0500 (Thu, 17 Apr 2008)
- | 3 lines Fix the bininstall target to install from subdirs, as
- well. (closes issue AST-8, patch from bmd at switchvox) ........
- ................
-
-2008-04-17 15:17 +0000 [r114203] Tilghman Lesher <tlesher@digium.com>
-
- * doc/CODING-GUIDELINES, /: Merged revisions 114202 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114202 | tilghman | 2008-04-17 10:12:52 -0500 (Thu, 17 Apr 2008)
- | 2 lines fileio.h does not exist; io.h does, though. ........
-
-2008-04-17 13:55 +0000 [r114200] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, res/res_jabber.c: Merged revisions 114199 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114199 | phsultan | 2008-04-17 15:46:17 +0200 (Thu, 17 Apr 2008)
- | 10 lines Merged revisions 114198 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114198 | phsultan | 2008-04-17 15:42:23 +0200 (Thu, 17 Apr 2008)
- | 2 lines Use keepalives effectively in order diagnose bug
- #12432. ........ ................
-
-2008-04-17 12:59 +0000 [r114197] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 114196 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114196 | tilghman | 2008-04-17 07:59:04 -0500 (Thu, 17 Apr 2008)
- | 16 lines Merged revisions 114195 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114195 | tilghman | 2008-04-17 07:56:38 -0500 (Thu, 17 Apr 2008)
- | 8 lines Add special case for when the agi cannot be executed,
- to comply with the documentation that we return failure in that
- case. (closes issue #12462) Reported by: fmueller Patches:
- 20080416__bug12462.diff.txt uploaded by Corydon76 (license 14)
- Tested by: fmueller ........ ................
-
-2008-04-17 10:56 +0000 [r114193] Sean Bright <sean.bright@gmail.com>
-
- * apps/app_chanspy.c, /: Merged revisions 114192 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114192 | seanbright | 2008-04-17 06:55:05 -0400 (Thu, 17 Apr
- 2008) | 9 lines Merged revisions 114191 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114191 | seanbright | 2008-04-17 06:51:20 -0400 (Thu, 17 Apr
- 2008) | 1 line Make sure we have enough room for the recording's
- filename. ........ ................
-
-2008-04-16 20:48 +0000 [r114186] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 114185 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114185 | kpfleming | 2008-04-16 15:47:30 -0500 (Wed, 16 Apr
- 2008) | 14 lines Merged revisions 114184 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr
- 2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by
- initializing the structure to all zeroes in case it contains
- fields that we don't write values into (which it does as of
- Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian
- ........ ................
-
-2008-04-15 20:53 +0000 [r114153] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 114152 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114152 |
- tilghman | 2008-04-15 15:51:08 -0500 (Tue, 15 Apr 2008) | 2 lines
- Oops, buffer wasn't long enough for query ........
-
-2008-04-15 20:09 +0000 [r114147] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 114146 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114146 |
- murf | 2008-04-15 13:59:50 -0600 (Tue, 15 Apr 2008) | 8 lines
- These changes: a. fix a self-found problem with SPAWN-ing an
- extension, where matches were not being found b. correct some
- wording in a comment c. Add some debug for future debugging.
- ........
-
-2008-04-15 17:22 +0000 [r114132-114142] Jason Parker <jparker@digium.com>
-
- * channels/chan_unistim.c, /: Merged revisions 114141 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114141 | qwell | 2008-04-15 12:21:58 -0500 (Tue, 15 Apr 2008) |
- 8 lines Shorten the mac address pattern, since some phones use
- different identifiers (such as the i2050 softphone). (closes
- issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff
- uploaded by c (license 460) Tested by: c_hans ........
-
- * contrib/scripts/autosupport, /: Merged revisions 114139 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114139 | qwell | 2008-04-15 12:17:37 -0500
- (Tue, 15 Apr 2008) | 15 lines Merged revisions 114138 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114138 | qwell | 2008-04-15 12:17:18 -0500 (Tue, 15 Apr 2008) |
- 7 lines Update Digium autosupport script, for more useful
- information. (closes issue #12452) Reported by: angler Patches:
- autosupport.diff uploaded by angler (license 106) ........
- ................
-
- * /, apps/app_queue.c: Merged revisions 114134 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114134 | qwell | 2008-04-15 11:18:38 -0500 (Tue, 15 Apr 2008) |
- 16 lines Merged revisions 114133 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114133 | qwell | 2008-04-15 11:18:08 -0500 (Tue, 15 Apr 2008) |
- 8 lines Allow autofill to work in the general section of
- queues.conf. Additionally, don't try to (re)set options when they
- have empty values in realtime (all unset columns would have an
- empty value). (closes issue #12445) Reported by: atis Patches:
- 12445-autofill.diff uploaded by qwell (license 4) ........
- ................
-
-2008-04-14 18:34 +0000 [r114122] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_h323.c: Merged revisions 114121 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114121 | qwell | 2008-04-14 13:34:17 -0500
- (Mon, 14 Apr 2008) | 15 lines Merged revisions 114120 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) |
- 7 lines The call_token on the pvt can occasionally be NULL,
- causing a crash. If it is NULL, we can skip this channel, since
- it can't the one we're looking for. (closes issue #9299) Reported
- by: vazir ........ ................
-
-2008-04-14 17:42 +0000 [r114119] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 114118 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114118 | mmichelson | 2008-04-14 12:42:20 -0500 (Mon, 14 Apr
- 2008) | 19 lines Merged revisions 114117 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr
- 2008) | 11 lines Increase the retry count when attempting to show
- channels. This apparently cleared an issue someone was seeing
- when attempting to show channels when the load was high. (closes
- issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
- by russell (license 2) Tested by: falves11 ........
- ................
-
-2008-04-14 16:33 +0000 [r114116] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 114115 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114115 | tilghman | 2008-04-14 11:32:59 -0500 (Mon, 14 Apr 2008)
- | 2 lines Make tab-completion work for all cases ........
-
-2008-04-14 16:25 +0000 [r114114] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 114113 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114113 | mmichelson | 2008-04-14 11:25:09 -0500
- (Mon, 14 Apr 2008) | 17 lines Merged revisions 114112 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr
- 2008) | 9 lines If the datastore has been moved to another
- channel due to a masquerade, then freeing the datastore here
- causes an eventual double free when the new channel hangs up. We
- should only free the datastore if we were able to successfully
- remove it from the channel we are referencing (i.e. the datastore
- was not moved). (closes issue #12359) Reported by: pguido
- ........ ................
-
-2008-04-14 15:02 +0000 [r114108] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 114107 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114107 | mmichelson | 2008-04-14 10:01:36 -0500 (Mon, 14 Apr
- 2008) | 13 lines Merged revisions 114106 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr
- 2008) | 5 lines Save a local copy of the generate callback prior
- to unlocking the channel in case the generate callback goes NULL
- on us after the channel is unlocked. Thanks to Russell for
- pointing this need out to me. ........ ................
-
-2008-04-14 14:54 +0000 [r114102-114105] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114104 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114104 | file | 2008-04-14 11:53:33 -0300 (Mon, 14 Apr 2008) |
- 12 lines Merged revisions 114103 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4
- lines It is possible for the remote side to say they want T38 but
- not give any capabilities. (closes issue #12414) Reported by: MVF
- ........ ................
-
- * main/rtp.c, /: Merged revisions 114101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114101 | file | 2008-04-14 10:53:33 -0300 (Mon, 14 Apr 2008) |
- 12 lines Merged revisions 114100 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4
- lines Don't change the SSRC when a new source comes into play,
- this might happen quite often and depending on the remote side...
- they might not like this. (closes issue #12353) Reported by:
- dimas ........ ................
-
-2008-04-14 02:59 +0000 [r114097-114099] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 114098 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114098 | tilghman | 2008-04-13 21:55:41 -0500 (Sun, 13 Apr 2008)
- | 3 lines Add tab command-line completion (Closes issue #12428)
- ........
-
- * /, apps/app_meetme.c: Merged revisions 114096 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114096 |
- tilghman | 2008-04-13 09:35:43 -0500 (Sun, 13 Apr 2008) | 3 lines
- Use ast_mkdir instead of mkdir (Closes issue #12430) ........
-
-2008-04-12 16:22 +0000 [r114094-114095] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure linkset is locked exiting
- ss7_start_call
-
- * channels/chan_zap.c: Make sure we start incoming calls on SS7
- with echo cancellation enabled. Also make sure when completing a
- COT we call ss7_start_call with the proper locks held. Lastly,
- make sure if we fail to get a channel from zt_new that we don't
- assume it's there.
-
-2008-04-11 23:27 +0000 [r114089-114091] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 114090 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114090 |
- tilghman | 2008-04-11 18:26:56 -0500 (Fri, 11 Apr 2008) | 3 lines
- If any field is not null, but has no default, then it must be set
- or the insert will fail. (Closes issue #12285) ........
-
- * /, configs/res_ldap.conf.sample: Merged revisions 114088 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r114088 | tilghman | 2008-04-11 18:21:54 -0500 (Fri, 11
- Apr 2008) | 3 lines Make the sample config match the contributed
- LDAP schema (Closes issue #12421) ........
-
-2008-04-11 23:21 +0000 [r114087] Terry Wilson <twilson@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 114084 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r114084 | twilson | 2008-04-11 17:48:52 -0500
- (Fri, 11 Apr 2008) | 15 lines Merged revisions 114083 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008)
- | 7 lines Several places in the code called find_callno() (which
- releases the lock on the pvt structure) and then immediately
- locked the call and did things with it. Unfortunately, the call
- can disappear between the find_callno and the lock, causing Bad
- Stuff(tm) to happen. Added find_callno_locked() function to
- return the callno withtout unlocking for instances that it is
- needed. (issue #12400) Reported by: ztel ........
- ................
-
-2008-04-11 23:13 +0000 [r114086] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 114085 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114085 | tilghman | 2008-04-11 18:12:16 -0500 (Fri, 11 Apr 2008)
- | 7 lines Use the correct function for free'ing objects, and
- maybe we won't crash. (closes issue #12163) Reported by: gservat
- Patches: 20080411__bug12163.diff.txt uploaded by Corydon76
- (license 14) Tested by: gservat ........
-
-2008-04-11 15:51 +0000 [r114065] Mark Michelson <mmichelson@digium.com>
-
- * /, main/features.c: Merged revisions 114064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114064 | mmichelson | 2008-04-11 10:49:35 -0500 (Fri, 11 Apr
- 2008) | 19 lines Merged revisions 114063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr
- 2008) | 11 lines Fix a race condition that may happen between a
- sip hangup and a "core show channel" command. This patch adds
- locking to prevent the resulting crash. (closes issue #12155)
- Reported by: tsearle Patches: show_channels_crash2.patch uploaded
- by tsearle (license 373) Tested by: tsearle ........
- ................
-
-2008-04-11 14:56 +0000 [r114062] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 114061 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114061 | tilghman | 2008-04-11 09:54:22 -0500 (Fri, 11 Apr 2008)
- | 6 lines Errors are all greater than 0 (closes issue #12422)
- Reported by: nito Patches:
- res_config_ldap_result_check_patch.diff uploaded by nito (license
- 340) ........
-
-2008-04-10 22:23 +0000 [r114056] Mark Michelson <mmichelson@digium.com>
-
- * utils/conf2ael.c, utils/check_expr.c, utils/Makefile,
- main/manager.c, /, utils/astman.c, utils/hashtest.c,
- main/utils.c, include/asterisk/lock.h, utils/ael_main.c,
- utils/hashtest2.c: Merged revisions 114052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114052 | mmichelson | 2008-04-10 17:02:32 -0500 (Thu, 10 Apr
- 2008) | 11 lines Merged revisions 114051 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr
- 2008) | 3 lines Fix 1.4 build when LOW_MEMORY is enabled.
- ........ ................
-
-2008-04-10 19:59 +0000 [r114047] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 114046 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114046 | mmichelson | 2008-04-10 14:58:36 -0500 (Thu, 10 Apr
- 2008) | 14 lines Merged revisions 114045 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr
- 2008) | 6 lines Be sure that we're not about to set bridgepvt
- NULL prior to dereferencing it. (closes issue #11775) Reported
- by: fujin ........ ................
-
-2008-04-10 19:09 +0000 [r114043] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 114042 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114042 | tilghman | 2008-04-10 14:04:29 -0500 (Thu, 10 Apr 2008)
- | 7 lines The hydra grows yet another head... (closes issue
- #12401) Reported by: davevg Patches: astcli.diff2 uploaded by
- davevg (license 209) Tested by: davevg, Corydon76 ........
-
-2008-04-10 17:27 +0000 [r114037] Jason Parker <jparker@digium.com>
-
- * /, main/file.c: Merged revisions 114036 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114036 | qwell | 2008-04-10 12:27:16 -0500 (Thu, 10 Apr 2008) |
- 18 lines Merged revisions 114035 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) |
- 10 lines Only try to prefix language if we are not using an
- absolute path (suffix it otherwise).
- en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
- issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
- uploaded by qwell (license 4) Tested by: kuj, qwell ........
- ................
-
-2008-04-10 16:00 +0000 [r114023-114034] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 114030 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114030 | file | 2008-04-10 12:10:47 -0300 (Thu, 10 Apr 2008) |
- 14 lines Merged revisions 114029 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6
- lines Create the directory where name recordings will go if it
- does not exist. (closes issue #12311) Reported by: rkeene
- Patches: 12311-mkdir.diff uploaded by qwell (license 4) ........
- ................
-
- * apps/app_voicemail.c, /: Merged revisions 114027 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r114027 | file | 2008-04-10 11:53:19 -0300 (Thu, 10 Apr 2008) | 6
- lines Don't hardcode ru into the digits filename so that
- languageprefix can work. (closes issue #12404) Reported by: IgorG
- Patches: voicemail_ru_hardcoded-v1.patch uploaded by IgorG
- (license 20) ........
-
- * main/rtp.c, channels/chan_unistim.c, /, channels/chan_skinny.c:
- Merged revisions 114024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r114024 |
- file | 2008-04-10 10:45:45 -0300 (Thu, 10 Apr 2008) | 4 lines Fix
- spelling of existent in a few places. (closes issue #12409)
- Reported by: candlerb ........
-
- * /, channels/chan_sip.c: Merged revisions 114022 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r114022 | file | 2008-04-10 10:28:30 -0300 (Thu, 10 Apr 2008) |
- 14 lines Merged revisions 114021 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6
- lines Don't add custom URI options if they don't exist OR they
- are empty. (closes issue #12407) Reported by: homesick Patches:
- uri_options-1.4.diff uploaded by homesick (license 91) ........
- ................
-
-2008-04-09 22:34 +0000 [r113929-113982] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 113980 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r113980 |
- mmichelson | 2008-04-09 17:32:32 -0500 (Wed, 09 Apr 2008) | 8
- lines Fix a crash that happened due to accessing free'd memory
- (closes issue #12396) Reported by: tcalosi Patches: 12396.patch
- uploaded by putnopvut (license 60) Tested by: tcalosi ........
-
- * /, channels/chan_sip.c: Merged revisions 113928 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr
- 2008) | 16 lines Merged revisions 113927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr
- 2008) | 8 lines We need to set the persistant_route [sic]
- parameter for the sip_pvt during the initial INVITE, no matter if
- we're building the route set from an INVITE request or response.
- (closes issue #12391) Reported by: benjaminbohlmann Tested by:
- benjaminbohlmann ........ ................
-
-2008-04-09 19:02 +0000 [r113876] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_csv.c, /, configs/cdr.conf.sample: Merged revisions
- 113875 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113875 | tilghman | 2008-04-09 14:00:40 -0500 (Wed, 09 Apr 2008)
- | 12 lines Merged revisions 113874 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008)
- | 4 lines If the [csv] section does not exist in cdr.conf, then
- an unload/load sequence is needed to correct the problem. Track
- whether the load succeeded with a variable, so we can fix this
- with a simple reload event, instead. ........ ................
-
-2008-04-09 17:56 +0000 [r113839] Jason Parker <jparker@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 113838 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r113838 | qwell | 2008-04-09 12:56:07 -0500 (Wed, 09 Apr 2008) |
- 2 lines Fix a small file handle "leak" pointed out by jjshoe on
- #asterisk. ........
-
-2008-04-09 17:50 +0000 [r113837] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, /: Merged revisions 113836 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r113836 |
- mmichelson | 2008-04-09 12:48:33 -0500 (Wed, 09 Apr 2008) | 14
- lines There was a subtle logical difference between 1.4 and trunk
- with regards to how timeouts were handled. In 1.4, if the
- absolute timeout were reached on a call, no matter what the
- return value of ast_spawn_extension was, the pbx would attempt to
- go to the 'T' extension or hangup otherwise. The rearrangement of
- this function in trunk made this check only happen in the case
- that ast_spawn_extension returned 0. If ast_spawn_extension
- returned 1, then the fact that the timeout expired resulted in a
- no-op, and would cause an infinite loop to occur in
- __ast_pbx_run. This change fixes this problem. Now timeouts will
- behave as they did in 1.4 (closes issue #11550) Reported by: pj
- Tested by: putnopvut ........
-
-2008-04-09 16:53 +0000 [r113786] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 113785 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113785 | file | 2008-04-09 13:52:04 -0300 (Wed,
- 09 Apr 2008) | 12 lines Merged revisions 113784 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4
- lines If we receive an AUTHREQ from the remote server and we are
- unable to reply (for example they have a secret configured, but
- we do not) then queue a hangup frame on the Asterisk channel.
- This will cause the channel to hangup and a HANGUP to be sent via
- IAX2 to the remote side which is the proper thing to do in this
- scenario. (closes issue #12385) Reported by: viraptor ........
- ................
-
-2008-04-09 14:42 +0000 [r113683] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 113682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113682 | mmichelson | 2008-04-09 09:41:58 -0500 (Wed, 09 Apr
- 2008) | 17 lines Merged revisions 113681 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr
- 2008) | 9 lines If Asterisk receives a 488 on an INVITE (not a
- reinvite), then we should not send a BYE. (closes issue #12392)
- Reported by: fnordian Patches: chan_sip.patch uploaded by
- fnordian (license 110) with small modification from me ........
- ................
-
-2008-04-09 13:56 +0000 [r113648-113650] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli: Merged revisions 113647 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r113647 | tilghman | 2008-04-09 08:23:44 -0500 (Wed, 09 Apr 2008)
- | 6 lines Additional enhancements (closes issue #12390) Reported
- by: tzafrir Patches: astcli_fixes.diff uploaded by tzafrir
- (license 46) ........
-
-2008-04-09 01:40 +0000 [r113598] Terry Wilson <twilson@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 113597 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113597 | twilson | 2008-04-08 20:36:58 -0500
- (Tue, 08 Apr 2008) | 10 lines Merged revisions 113596 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008)
- | 2 lines Initialize fr->cacheable to make valgrind happy
- ........ ................
-
-2008-04-08 21:34 +0000 [r113560] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astcli (added): Merged revisions 113559 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r113559 | tilghman | 2008-04-08 16:33:11 -0500 (Tue, 08
- Apr 2008) | 6 lines Add commandline tool for doing CLI commands
- through AMI (instead of using asterisk -rx) (closes issue #12389)
- Reported by: davevg Patches: astcli uploaded by davevg (license
- 209) ........
-
-2008-04-08 18:49 +0000 [r113404-113506] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 113505 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113505 | qwell | 2008-04-08 13:49:21 -0500
- (Tue, 08 Apr 2008) | 9 lines Merged revisions 113504 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr
- 2008) | 1 line Add a little more that is required for previously
- added devices. ........ ................
-
- * /, channels/chan_skinny.c: Merged revisions 113455 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113455 | qwell | 2008-04-08 13:08:35 -0500
- (Tue, 08 Apr 2008) | 12 lines Merged revisions 113454 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) |
- 4 lines Add support for several new(ish) devices - most notably,
- 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver for providing
- me the required information. ........ ................
-
- * main/asterisk.c, /: Merged revisions 113403 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113403 | qwell | 2008-04-08 12:00:55 -0500 (Tue, 08 Apr 2008) |
- 9 lines Merged revisions 113402 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) |
- 1 line Work around some silliness caused by sys/capability.h -
- this should fix compile errors a number of users have been
- experiencing. ........ ................
-
-2008-04-08 16:56 +0000 [r113350-113401] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/astgenkey.8: Merged revisions 113400 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113400 | tilghman | 2008-04-08 11:54:21 -0500
- (Tue, 08 Apr 2008) | 14 lines Merged revisions 113399 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008)
- | 6 lines Add security note on astgenkey's manpage. (closes issue
- #12373) Reported by: lmamane Patches: 20080406__bug12373.diff.txt
- uploaded by Corydon76 (license 14) ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 113349 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113349 | tilghman | 2008-04-08 10:48:58 -0500 (Tue, 08 Apr 2008)
- | 15 lines Merged revisions 113348 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008)
- | 7 lines Move check for still-bridged channels out a little
- further, to avoid possible deadlocks. (Closes issue #12252)
- Reported by: callguy Patches: 20080319__bug12252.diff.txt
- uploaded by Corydon76 (license 14) Tested by: callguy ........
- ................
-
-2008-04-08 15:10 +0000 [r113298-113299] Joshua Colp <jcolp@digium.com>
-
- * /, main/audiohook.c: Merged revisions 113297 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113297 | file | 2008-04-08 12:05:35 -0300 (Tue, 08 Apr 2008) |
- 12 lines Merged revisions 113296 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4
- lines If audio suddenly gets fed into one side of a channel after
- a lapse of frames flush the other factory so that old audio does
- not remain in the factory causing the sync code to not execute.
- (closes issue #12296) Reported by: jvandal ........
- ................
-
-2008-04-07 22:17 +0000 [r113246] Tilghman Lesher <tlesher@digium.com>
-
- * /, configs/manager.conf.sample: Merged revisions 113245 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r113245 | tilghman | 2008-04-07 17:16:46 -0500 (Mon, 07
- Apr 2008) | 2 lines Additional note ........
-
-2008-04-07 21:49 +0000 [r113244] Jason Parker <jparker@digium.com>
-
- * /, configs/manager.conf.sample: Merged revisions 113243 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r113243 | qwell | 2008-04-07 16:49:27 -0500 (Mon, 07 Apr
- 2008) | 1 line Document 'originate' permission in manager sample
- config. ........
-
-2008-04-07 21:36 +0000 [r113242] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 113241 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113241 | jpeeler | 2008-04-07 16:35:48 -0500 (Mon, 07 Apr 2008)
- | 23 lines Merged revisions 113013 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008)
- | 15 lines Merged revisions 113012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008)
- | 7 lines (closes issue #12362) (closes issue #12372) Reported
- by: vinsik Tested by: tecnoxarxa This one line change makes an if
- inside a for loop (in realtime_peer) check all the ast_variables
- the loop was intending to test rather than just the first one.
- ........ ................ ................
-
-2008-04-07 19:10 +0000 [r113174] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged
- revisions 113119 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113119 | qwell | 2008-04-07 13:02:51 -0500 (Mon, 07 Apr 2008) |
- 16 lines Merged revisions 113118 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) |
- 8 lines Allow playback with noanswer (and add earlyrtp option).
- (closes issue #9077) Reported by: pj Patches: earlyrtp.diff
- uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA,
- wedhorn ........ ................
-
-2008-04-07 19:08 +0000 [r113173] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 113172 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r113172 | tilghman | 2008-04-07 14:06:46 -0500
- (Mon, 07 Apr 2008) | 11 lines Merged revisions 113117 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008)
- | 3 lines Force ast_mktime() to check for DST, since strptime(3)
- does not. (Closes issue #12374) ........ ................
-
-2008-04-07 16:13 +0000 [r113067] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 113066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113066 | mmichelson | 2008-04-07 11:12:30 -0500 (Mon, 07 Apr
- 2008) | 21 lines Merged revisions 113065 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr
- 2008) | 13 lines This fix prevents a deadlock that was
- experienced in chan_local. There was deadlock prevention in place
- in chan_local, but it would not work in a specific case because
- the channel was recursively locked. By unlocking the channel
- prior to calling the generator's generate callback in
- ast_read_generator_actions(), we prevent the recursive locking,
- and therefore the deadlock. (closes issue #12307) Reported by:
- callguy Patches: 12307.patch uploaded by putnopvut (license 60)
- Tested by: callguy ........ ................
-
-2008-04-07 15:28 +0000 [r113042] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 113013 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008)
- | 15 lines Merged revisions 113012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008)
- | 7 lines (closes issue #12362) (closes issue #12372) Reported
- by: vinsik Tested by: tecnoxarxa This one line change makes an if
- inside a for loop (in realtime_peer) check all the ast_variables
- the loop was intending to test rather than just the first one.
- ........ ................
-
-2008-04-05 13:30 +0000 [r112973-112975] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 112972 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r112972 |
- tilghman | 2008-04-05 08:24:12 -0500 (Sat, 05 Apr 2008) | 6 lines
- AsyncAGI should not close the manager session on error. (closes
- issue #12370) Reported by: srt Patches: asterisk-12370.diff
- uploaded by srt (license 378) ........
-
-2008-04-04 19:30 +0000 [r112786-112822] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 112821 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r112821 | phsultan | 2008-04-04 21:28:49 +0200
- (Fri, 04 Apr 2008) | 9 lines Merged revisions 112820 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04
- Apr 2008) | 1 line Free newly allocated channel before returning
- ........ ................
-
- * /, channels/chan_gtalk.c: Merged revisions 112785 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r112785 | phsultan | 2008-04-04 19:32:46 +0200
- (Fri, 04 Apr 2008) | 15 lines Merged revisions 112766 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008)
- | 7 lines Prevent call connections when codecs don't match.
- (closes issue #10604) Reported by: keepitcool Patches:
- branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
- by: phsultan ........ ................
-
-2008-04-04 01:08 +0000 [r112715] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/asterisk.c, /: Merged revisions 112653,112656,112714 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r112653 | dhubbard | 2008-04-03 17:13:11 -0500 (Thu, 03
- Apr 2008) | 1 line add a Zaptel timer check to verify the timer
- is responding when Zaptel support is compiled into Asterisk and
- Zaptel drivers are loaded. This will help people not waste their
- valuable time debugging side effects. ........ r112656 | dhubbard
- | 2008-04-03 17:19:43 -0500 (Thu, 03 Apr 2008) | 1 line satisfy
- buildbot ........ r112714 | dhubbard | 2008-04-03 19:57:33 -0500
- (Thu, 03 Apr 2008) | 1 line sleep long enough for the zaptel
- timer error message to display before exit ........
-
-2008-04-04 00:54 +0000 [r112713] Joshua Colp <jcolp@digium.com>
-
- * /, main/Makefile: Merged revisions 112712 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112712 | file | 2008-04-03 21:53:19 -0300 (Thu, 03 Apr 2008) |
- 10 lines Merged revisions 112711 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2
- lines Pass in the path to Zaptel for systems that install Zaptel
- headers in a separate location. ........ ................
-
-2008-04-03 14:42 +0000 [r112601] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 112600 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112600 | mmichelson | 2008-04-03 09:35:47 -0500 (Thu, 03 Apr
- 2008) | 17 lines Merged revisions 112599 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr
- 2008) | 9 lines Fix the testing of the "res" variable so that it
- is more logically correct and makes the correct warning and debug
- messages print. (closes issue #12361) Reported by: one47 Patches:
- chan_zap_deferred_digit.patch uploaded by one47 (license 23)
- ........ ................
-
-2008-04-02 17:37 +0000 [r112470] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 112469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112469 | mmichelson | 2008-04-02 12:36:49 -0500 (Wed, 02 Apr
- 2008) | 21 lines Merged revisions 112468 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr
- 2008) | 13 lines Fix a race condition in the manager. It is
- possible that a new manager event could be appended during a
- brief time when the manager is not waiting for input. If an event
- comes during this period, we need to set an indicator that there
- is an event pending so that the manager doesn't attempt to wait
- forever for an event that already happened. (closes issue #12354)
- Reported by: bamby Patches: manager_race_condition.diff uploaded
- by bamby (license 430) (comments added by me) ........
- ................
-
-2008-04-02 15:27 +0000 [r112436] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 112431 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r112431 |
- file | 2008-04-02 12:26:51 -0300 (Wed, 02 Apr 2008) | 7 lines
- Since the SIP request structure gets reused multiple times with
- TCP handling we have to clear the debug state or else we will
- keep spitting out debug even after it has been turned off.
- (closes issue #12169) Reported by: pj Patches:
- 12169-debugoff-2.diff uploaded by qwell (license 4) Tested by: pj
- ........
-
-2008-04-02 14:33 +0000 [r112395] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 112394 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112394 | mmichelson | 2008-04-02 09:32:43 -0500 (Wed, 02 Apr
- 2008) | 14 lines Merged revisions 112393 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112393 | mmichelson | 2008-04-02 09:32:00 -0500 (Wed, 02 Apr
- 2008) | 6 lines Ensure that there is no timeout if none is
- specified. (closes issue #12349) Reported by: johnlange ........
- ................
-
-2008-04-01 22:48 +0000 [r112359] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 112357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r112357 |
- murf | 2008-04-01 16:45:10 -0600 (Tue, 01 Apr 2008) | 1 line
- Bumped across another test set for the new exten pattern matcher,
- which revealed a problem with the CANMATCH/MATCHMORE modes.
- Direct matches were getting in the way. Fixed. ........
-
-2008-04-01 20:20 +0000 [r112299] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 112289 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r112289 |
- murf | 2008-04-01 14:02:19 -0600 (Tue, 01 Apr 2008) | 21 lines
- (closes issue #12298) Reported by: falves11 Patches: 12298.patch1
- uploaded by murf (license 17) Tested by: murf I have hopes that
- the changes made over the last few days will finalize and
- solidify this code. While there are bound to be small tweaks
- still needed, I feel that the job (at last) is somewhat
- completed. Finally, I had a chance to comprehend how the scoring
- of extension patterns was done in the previous version, and I've
- come very close to using the exact same criteria in the new
- pattern matching code. The left-right sorting is now replicated
- in the trie structure itself, such that the first match found
- will the 'best' match. Compared the results against 1.4 for
- several extensions. Replicated falves11's setup and it works.
- Used some devious patterns provided by jsmith, supplemented with
- a few of my own. Looks good. ........
-
-2008-04-01 18:09 +0000 [r112211] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 112210 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112210 | file | 2008-04-01 15:06:13 -0300 (Tue, 01 Apr 2008) |
- 12 lines Merged revisions 112209 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4
- lines Disable Packet2Packet bridging when we need to feed DTMF
- frames into the core. Some implementations do not like how we
- switch between things. (closes issue #12212) Reported by: bamby
- ........ ................
-
-2008-04-01 17:52 +0000 [r112170-112206] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 112205 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) |
- 12 lines Merged revisions 112204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4
- lines Do not pass audio until the remote side has indicated they
- are providing early media, or if the channel has been answered.
- (closes issue #11823) Reported by: SDamm ........
- ................
-
-2008-04-01 17:25 +0000 [r112157] Mark Michelson <mmichelson@digium.com>
-
- * main/dns.c, /: Merged revisions 112148 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112148 | mmichelson | 2008-04-01 12:23:19 -0500 (Tue, 01 Apr
- 2008) | 18 lines Merged revisions 112138 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr
- 2008) | 10 lines Initialize the __res_state structure used for
- dns purposes to all 0's prior to using it. This is due to
- valgrind's complaints on issue #12284 as well as an excerpt found
- in "Description" portion of the online man page found here:
- http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV
- (pertains to issue #12284 but does not necessarily close it)
- ........ ................
-
-2008-04-01 16:57 +0000 [r112127] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/slinfactory.h, /, main/slinfactory.c: Merged
- revisions 112126 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r112126 | file | 2008-04-01 13:50:37 -0300 (Tue, 01 Apr 2008) |
- 13 lines Merged revisions 112125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5
- lines Ensure that we do not exceed the hold's maximum size with a
- single frame. (closes issue #12047) Reported by: fabianoheringer
- Tested by: fabianoheringer ........ ................
-
-2008-03-31 22:17 +0000 [r112070-112072] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 112069 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r112069 | qwell | 2008-03-31 16:48:30 -0500
- (Mon, 31 Mar 2008) | 13 lines Merged revisions 112068 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r112068 | qwell | 2008-03-31 16:48:05 -0500 (Mon, 31 Mar 2008) |
- 5 lines Fix a silly infinite loop when choosing an invalid
- option. (closes issue #12315) Reported by: jmls ........
- ................
-
-2008-03-31 21:03 +0000 [r112034-112036] Terry Wilson <twilson@digium.com>
-
- * /, main/http.c: Merged revisions 112033 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r112033 |
- twilson | 2008-03-31 15:45:05 -0500 (Mon, 31 Mar 2008) | 2 lines
- Handle blank prefix= in http.conf ........
-
-2008-03-31 17:15 +0000 [r111997-111999] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 111998 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r111998 |
- russell | 2008-03-31 12:14:58 -0500 (Mon, 31 Mar 2008) | 7 lines
- Ensure configure gets run on a clean checkout. (closes issue
- #12197) Reported by: juggie Patches: 12197.diff uploaded by
- juggie (license 24) ........
-
-2008-03-31 14:22 +0000 [r111962] Joshua Colp <jcolp@digium.com>
-
- * res/res_config_sqlite.c, /: Merged revisions 111961 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r111961 | file | 2008-03-31 11:20:39 -0300 (Mon, 31 Mar 2008) | 4
- lines Initialize all these here tmp pointers at declaration. They
- confused some compilers a wee bit. (closes issue #12333) Reported
- by: ovi ........
-
-2008-03-29 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta7.1 released.
-
- Asterisk 1.6.0-beta7 was tagged against trunk, instead of the 1.6.0 branch.
-
-2008-03-28 21:46 +0000 [r111858] Jason Parker <jparker@digium.com>
-
- * codecs/gsm/inc/private.h, /: Merged revisions 111857 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111857 | qwell | 2008-03-28 16:46:02 -0500
- (Fri, 28 Mar 2008) | 20 lines Merged revisions 111856 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) |
- 12 lines Allow gsm to compile correctly on x86 with gcc4
- optimizations. (closes issue #11243) Reported by: whiskerp
- Patches: 11243-maybe-asm.diff uploaded by qwell (license 4)
- Tested by: Seggy (IRC) Note: While I did write this patch, I
- would not have found this if fossil had not reported and fixed
- issue #12253. A huge thanks to him for helping to (indirectly)
- find the problem here. ........ ................
-
-2008-03-28 19:11 +0000 [r111722-111776] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 111721 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111721 | qwell | 2008-03-28 12:57:12 -0500
- (Fri, 28 Mar 2008) | 9 lines Merged revisions 111720 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar
- 2008) | 1 line Remove unimplemented softkeys. Prompted by issue
- #12325. ........ ................
-
-2008-03-28 16:21 +0000 [r111660] Jason Parker <jparker@digium.com>
-
- * /, formats/format_wav_gsm.c: Merged revisions 111659 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111659 | qwell | 2008-03-28 11:20:59 -0500
- (Fri, 28 Mar 2008) | 16 lines Merged revisions 111658 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) |
- 8 lines The file size of WAV49 does not need to be an even
- number. (closes issue #12128) Reported by: mdu113 Patches:
- 12128-noevenlength.diff uploaded by qwell (license 4) Tested by:
- qwell, mdu113 ........ ................
-
-2008-03-28 14:43 +0000 [r111607-111608] Tilghman Lesher <tlesher@digium.com>
-
- * doc/valgrind.txt, /: Merged revisions 111606 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111606 | tilghman | 2008-03-28 09:37:28 -0500 (Fri, 28 Mar 2008)
- | 11 lines Merged revisions 111605 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008)
- | 3 lines Update debugging text, since Valgrind eliminated the
- --log-file-exactly option. (Closes issue #12320) ........
- ................
-
-2008-03-28 00:56 +0000 [r111566] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 111565 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r111565 |
- file | 2008-03-27 21:55:47 -0300 (Thu, 27 Mar 2008) | 2 lines
- Forgetting to unregister a manager action is bad, mmmk? ........
-
-2008-03-28 00:17 +0000 [r111534] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 111533 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r111533 |
- mmichelson | 2008-03-27 19:12:52 -0500 (Thu, 27 Mar 2008) | 10
- lines Fix a crash that would happen when attempting to unload the
- app_queue module. The problem was that when the refcount on the
- queue hit 0, the destructor was called, and inside the
- destructor, another function was called which would increase the
- refcount back to 1 again and then decrease it again back to 0 for
- every member in the queue. This meant that the destructor was
- being recursively called, leading to a double free of the queue.
- This is now fixed by making sure to unlink the queue from the
- queues container prior to the final unref of the queue. ........
-
-2008-03-27 21:28 +0000 [r111498] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 111497 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r111497 |
- murf | 2008-03-27 15:25:55 -0600 (Thu, 27 Mar 2008) | 1 line
- comment cleanup and iron out a really dumb mistake in handling
- the '.'-wildcard in the new exten pattern matcher. ........
-
-2008-03-27 19:30 +0000 [r111444] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/acl.c: Merged revisions 111443 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111443 | tilghman | 2008-03-27 14:26:45 -0500 (Thu, 27 Mar 2008)
- | 14 lines Merged revisions 111442 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008)
- | 6 lines For FreeBSD, at least, the ifa_addr element could be
- NULL. (closes issue #12300) Reported by: festr Patches:
- acl.c.patch uploaded by festr (license 443) ........
- ................
-
-2008-03-27 13:42 +0000 [r111361-111411] Steve Murphy <murf@digium.com>
-
- * apps/app_playback.c, main/pbx.c, /: Merged revisions 111410 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111410 | murf | 2008-03-27 07:29:41 -0600 (Thu,
- 27 Mar 2008) | 17 lines Merged revisions 111391 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9
- lines These small documentation updates made in response to a
- query in asterisk-users, where a user was using Playback, but
- needed the features of Background, and had no idea that
- Background existed, or that it might provide the features he
- needed. I thought the best way to avert these kinds of queries
- was to provide "See Also" references in all three of
- "Background", "Playback", "WaitExten". Perhaps a project to do
- this with all related apps is in order. ........ ................
-
- * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c,
- include/asterisk/ael_structs.h: Merged revisions 111360 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111360 | murf | 2008-03-26 22:47:12 -0600 (Wed,
- 26 Mar 2008) | 23 lines Merged revisions 111341 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) |
- 15 lines (closes issue #12302) Reported by: pj Tested by: murf
- These changes will set a channel variable ~~EXTEN~~ just before
- generating code for a switch, with the value of ${EXTEN}. The
- exten is marked as having a switch, and ever after that, till the
- end of the exten, we substitute any ${EXTEN} with ${~~EXTEN~~}
- instead in application arguments; (and the ${EXTEN: also). The
- reason for this, is that because switches are coded using
- separate extensions to provide pattern matching, and jumping
- to/from these switch extensions messes up the ${EXTEN} value,
- which blows the minds of users. ........ ................
-
-2008-03-27 00:36 +0000 [r111247-111339] Jason Parker <jparker@digium.com>
-
- * main/frame.c, /: Merged revisions 111285 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111285 | qwell | 2008-03-26 19:25:56 -0500 (Wed, 26 Mar 2008) |
- 9 lines Merged revisions 111280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) |
- 1 line Put this flag back so we don't change the API. ........
- ................
-
- * main/frame.c, /: Merged revisions 111246 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111246 | qwell | 2008-03-26 18:27:33 -0500 (Wed, 26 Mar 2008) |
- 17 lines Merged revisions 111245 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) |
- 9 lines Remove excessive smoother optimization that was causing
- audio glitches (small "pops") after (about 200ms later) an
- "incorrectly" sized frame was received. While it would be very
- nice to keep this as optimized as possible, it makes no sense for
- the smoother to be dropping random bits of audio like this. Isn't
- that the whole point of a smoother? Closes issue #12093. ........
- ................
-
-2008-03-26 19:57 +0000 [r111131] Joshua Colp <jcolp@digium.com>
-
- * contrib/scripts/autosupport, /: Merged revisions 111130 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111130 | file | 2008-03-26 16:56:40 -0300 (Wed,
- 26 Mar 2008) | 14 lines Merged revisions 111129 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6
- lines Update autosupport script. (closes issue #12310) Reported
- by: angler Patches: autosupport.diff uploaded by angler (license
- 106) ........ ................
-
-2008-03-26 19:53 +0000 [r111128] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 111127 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111127 | kpfleming | 2008-03-26 14:52:27 -0500 (Wed, 26 Mar
- 2008) | 18 lines Merged revisions 111126 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500
- (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
- 2008) | 2 lines update UPGRADE notes to document usage of the
- script ........ ................ ................
-
-2008-03-26 19:41 +0000 [r111124] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 111123 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111123 | mmichelson | 2008-03-26 14:39:23 -0500
- (Wed, 26 Mar 2008) | 12 lines Merged revisions 111121 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar
- 2008) | 4 lines This code change is made just for clarification.
- It does exactly the same thing as before. It just doesn't look as
- wrong. ........ ................
-
-2008-03-26 19:27 +0000 [r111072] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 111067 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111067 | mmichelson | 2008-03-26 14:26:23 -0500
- (Wed, 26 Mar 2008) | 17 lines Merged revisions 111049 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar
- 2008) | 9 lines Add a lock to the vm_state structure and use the
- lock around mail_open calls to prevent concurrent access of the
- same mailstream. This, along with trunk's ability to configure
- TCP timeouts for IMAP storage will help to prevent crashes and
- hangs when using voicemail with IMAP storage. (closes issue
- #10487) Reported by: ewilhelmsen ........ ................
-
-2008-03-26 19:08 +0000 [r111026] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
- Merged revisions 111025 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111025 | kpfleming | 2008-03-26 14:08:00 -0500 (Wed, 26 Mar
- 2008) | 18 lines Merged revisions 111024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r111024 | kpfleming | 2008-03-26 14:06:56 -0500
- (Wed, 26 Mar 2008) | 10 lines Merged revisions 111019 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
- 2008) | 2 lines add a script to make getting the iLBC source code
- simple for end users ........ ................ ................
-
-2008-03-26 19:06 +0000 [r111018-111023] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 111021 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r111021 | file | 2008-03-26 16:05:42 -0300 (Wed, 26 Mar 2008) |
- 12 lines Merged revisions 111020 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4
- lines If we are requested to authenticate a reinvite make sure
- that it contains T38 SDP if need be. (closes issue #11995)
- Reported by: fall ........ ................
-
- * /, channels/chan_iax2.c: Merged revisions 111017 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r111017 | file | 2008-03-26 15:42:52 -0300 (Wed,
- 26 Mar 2008) | 12 lines Merged revisions 110628 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4
- lines Add an option (transmit_silence) which transmits silence
- during both Record() and DTMF generation. The reason this is an
- option is that in order to transmit silence we have to setup a
- translation path. This may not be needed/wanted in all cases.
- (closes issue #10058) Reported by: tracinet ........
- ................
-
-2008-03-26 17:44 +0000 [r110964] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 110963 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110963 | kpfleming | 2008-03-26 12:44:09 -0500 (Wed, 26 Mar
- 2008) | 10 lines Merged revisions 110962 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar
- 2008) | 2 lines add note that the user will need to enable
- codec_ilbc to get it to build ........ ................
-
-2008-03-26 17:35 +0000 [r110959] Donny Kavanagh <donnyk@gmail.com>
-
- * /, doc/snmp.txt: Merged revisions 110911 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110911 |
- juggie | 2008-03-26 13:24:54 -0400 (Wed, 26 Mar 2008) | 8 lines
- update documentation to reflect the changes in the way configure
- detects net-snmp. (closes issue #12067) Reported by: juggie
- Patches: 12067_snmp_doc.patch uploaded by juggie (license 24)
- Tested by: juggie ........
-
-2008-03-26 17:15 +0000 [r110882] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/ilbc/constants.h (removed), codecs/ilbc/iLBC_decode.h
- (removed), codecs/ilbc/iCBSearch.c (removed), codecs/Makefile,
- codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
- codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
- (removed), codecs/ilbc/iCBSearch.h (removed),
- codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
- codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
- (removed), codecs/ilbc/hpOutput.h (removed),
- codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
- codecs/ilbc/LPCencode.h (removed), codecs/ilbc/iCBConstruct.c
- (removed), codecs/ilbc/StateSearchW.h (removed),
- codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
- (removed), codecs/ilbc/syntFilter.h (removed),
- codecs/ilbc/packing.c (removed), codecs/ilbc/StateConstructW.c
- (removed), codecs/ilbc/packing.h (removed),
- codecs/ilbc/libilbc.vcproj (removed),
- codecs/ilbc/StateConstructW.h (removed), codecs/ilbc/LPCdecode.c
- (removed), codecs/ilbc/getCBvec.c (removed),
- codecs/ilbc/enhancer.c (removed), codecs/ilbc/lsf.c (removed),
- codecs/ilbc/iLBC_encode.c (removed), codecs/ilbc/getCBvec.h
- (removed), codecs/ilbc/LPCdecode.h (removed),
- codecs/ilbc/enhancer.h (removed), codecs/ilbc/FrameClassify.c
- (removed), codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h
- (removed), codecs/ilbc/iLBC_encode.h (removed),
- codecs/ilbc/FrameClassify.h (removed), codecs/ilbc/helpfun.c
- (removed), codecs/ilbc/doCPLC.c (removed),
- codecs/ilbc/anaFilter.c (removed), codecs/ilbc/helpfun.h
- (removed), codecs/ilbc/createCB.c (removed), codecs/ilbc/doCPLC.h
- (removed), codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
- codecs/ilbc/constants.c (removed), codecs/ilbc/iLBC_decode.c
- (removed), codecs/ilbc/createCB.h (removed), CHANGES: Merged
- revisions 110881 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110881 | kpfleming | 2008-03-26 10:10:28 -0700 (Wed, 26 Mar
- 2008) | 18 lines Merged revisions 110880 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700
- (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
- 2008) | 2 lines due to licensing restrictions, we cannot
- distribute the source code for iLBC encoding and decoding... so
- remove it, and add instructions on how the user can obtain it
- themselves ........ ................ ................
-
-2008-03-26 15:33 +0000 [r110866-110868] Joshua Colp <jcolp@digium.com>
-
- * /: Merged revisions 110726 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110726 |
- jpeeler | 2008-03-25 17:02:57 -0300 (Tue, 25 Mar 2008) | 2 lines
- This one line change makes an if inside a for loop (in
- realtime_peer) check all the ast_variables the loop was intending
- to test rather than just the first one. ........
-
-2008-03-26 00:03 +0000 [r110832] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 110831 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110831 |
- mmichelson | 2008-03-25 19:02:31 -0500 (Tue, 25 Mar 2008) | 6
- lines This ensures that the manager interface is not enabled by
- default. Prior to this change, it was possible to start Asterisk
- with the manager interface enabled, then either comment out the
- enabled option or make manager.conf unopenable and the manager
- interface would still be enabled. ........
-
-2008-03-25 22:52 +0000 [r110781] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_custom.c, /: Merged revisions 110780 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110780 | qwell | 2008-03-25 17:51:55 -0500 (Tue, 25 Mar 2008) |
- 14 lines Merged revisions 110779 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) |
- 6 lines Make file access in cdr_custom similar to cdr_csv. Fixes
- issue #12268. Patch borrowed from r82344 ........
- ................
-
-2008-03-25 22:11 +0000 [r110778] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_sip.c: This one line change makes an if inside a
- for loop (in realtime_peer) check all the ast_variables the loop
- was intending to test rather than just the first one.
-
-2008-03-25 17:47 +0000 [r110690-110692] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.conf.sample, /, configs/voicemail.conf.sample:
- Merged revisions 110691 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110691 |
- tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines
- Update sample configurations to make virtual hosting more
- obvious. (closes issue #11969) Reported by: pprindeville Patches:
- acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)
- ........
-
- * configs/extensions.conf.sample, /: Merged revisions 110689 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25
- Mar 2008) | 6 lines Update the sample configuration, to use Macro
- less (since it's now deprecated). (closes issue #12293) Reported
- by: pprindeville Patches: bugid-0012293.1.6.patch uploaded by
- pprindeville (license 347) ........
-
-2008-03-25 15:43 +0000 [r110637-110638] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Oops.
-
- * /, channels/chan_sip.c: Merged revisions 110636 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110636 | mmichelson | 2008-03-25 10:41:33 -0500 (Tue, 25 Mar
- 2008) | 15 lines Merged revisions 110635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar
- 2008) | 7 lines When reverting a commit, I accidentally left in
- this bit which was an experiment to see what would happen. It
- passed the compile test, and I didn't notice I had left this
- change in too. So this is a revert of a revert...sort of.
- ........ ................
-
-2008-03-25 15:39 +0000 [r110630-110634] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/options.h, main/asterisk.c, Makefile, /,
- main/app.c: Merged revisions 110629 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110629 | file | 2008-03-25 11:39:45 -0300 (Tue, 25 Mar 2008) |
- 12 lines Merged revisions 110628 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4
- lines Add an option (transmit_silence) which transmits silence
- during both Record() and DTMF generation. The reason this is an
- option is that in order to transmit silence we have to setup a
- translation path. This may not be needed/wanted in all cases.
- (closes issue #10058) Reported by: tracinet ........
- ................
-
-2008-03-24 20:14 +0000 [r110620-110622] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 110619 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110619 | mmichelson | 2008-03-24 14:19:37 -0500 (Mon, 24 Mar
- 2008) | 23 lines Merged revisions 110618 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar
- 2008) | 15 lines This is a revert for revision 108288. The reason
- is that that revision was not for an actual bug fix per se, and
- so it really should not have been in 1.4 in the first place.
- Plus, people who compile with DO_CRASH are more likely to
- encounter a crash due to this change. While I think the usage of
- DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
- beyond the scope of 1.4 and should be done instead in a developer
- branch based on trunk so that all scheduler functions are fixed
- at once. I also am reverting the change to trunk and 1.6 since
- they also suffer from the DO_CRASH potential. (closes issue
- #12272) Reported by: qq12345 ........ ................
-
-2008-03-24 17:36 +0000 [r110616] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 110615 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r110615 | russell | 2008-03-24 12:36:04 -0500
- (Mon, 24 Mar 2008) | 10 lines Merged revisions 110614 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008)
- | 2 lines Turn a NOTICE into a DEBUG message. ........
- ................
-
-2008-03-24 15:29 +0000 [r110611] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 110610 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110610 |
- file | 2008-03-24 12:28:25 -0300 (Mon, 24 Mar 2008) | 6 lines
- Only print out the set_address_from_contact host verbose message
- if debugging is enabled on the dialog. (closes issue #12280)
- Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain
- (license 226) ........
-
-2008-03-21 21:52 +0000 [r110579] Jason Parker <jparker@digium.com>
-
- * /, sounds/Makefile: Merged revisions 110578 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110578 |
- qwell | 2008-03-21 16:52:06 -0500 (Fri, 21 Mar 2008) | 1 line
- Update to 1.4.11 core sounds. ........
-
-2008-03-21 15:25 +0000 [r110501] Russell Bryant <russell@digium.com>
-
- * /, configs/sip.conf.sample, CHANGES: Merged revisions 110499 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21
- Mar 2008) | 3 lines Note that the TCP and TLS support is
- currently considered experimental and is subject to change while
- we work out the remaining issues. ........
-
-2008-03-21 14:36 +0000 [r110476] Jason Parker <jparker@digium.com>
-
- * /, codecs/gsm/Makefile: Merged revisions 110475 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110475 | qwell | 2008-03-21 09:36:17 -0500 (Fri, 21 Mar 2008) |
- 15 lines Merged revisions 110474 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) |
- 7 lines Don't attempt to do optimizations of gsm on mips
- platforms either. (closes issue #12270) Reported by: zandbelt
- Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33)
- ........ ................
-
-2008-03-20 23:14 +0000 [r110304-110397] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 110396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110396 | russell | 2008-03-20 18:14:13 -0500 (Thu, 20 Mar 2008)
- | 17 lines Merged revisions 110395 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008)
- | 9 lines Shorten the ast_waitfor() timeout from 500 ms to 50 ms
- in the autoservice thread. This really should not make a
- difference except in very rare cases. That case would be that all
- of the channels in autoservice are not generating any frames. In
- that case, this change reduces the potential amount of time that
- a thread waits in ast_autoservice_stop() for the autoservice
- thread to wrap back around to the beginning of its loop. (closes
- issue #12266, reported by dimas) ........ ................
-
- * codecs/codec_g722.c, /: Merged revisions 110339 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110339 |
- russell | 2008-03-20 17:02:20 -0500 (Thu, 20 Mar 2008) | 3 lines
- Use the correct buffer for g722tolin16_sample. This shouldn't
- have caused any problems, but Qwell noticed the typo here.
- ........
-
- * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
- 110337 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110337 | russell | 2008-03-20 16:55:50 -0500 (Thu, 20 Mar 2008)
- | 22 lines Merged revisions 110336 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r110336 | russell | 2008-03-20 16:54:58 -0500
- (Thu, 20 Mar 2008) | 14 lines Merged revisions 110335 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
- | 6 lines Fix some very broken code that was introduced in 1.2.26
- as a part of the security fix. The dnsmgr is not appropriate
- here. The dnsmgr takes a pointer to an address structure that a
- background thread continuously updates. However, in these cases,
- a stack variable was passed. That means that the dnsmgr thread
- would be continuously writing to bogus memory. ........
- ................ ................
-
- * /, main/file.c: Merged revisions 110303 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110303 |
- russell | 2008-03-20 15:08:26 -0500 (Thu, 20 Mar 2008) | 8 lines
- Fix a bug when using zaptel timing for playing back files that
- have a sample rate other than 8 kHz. The issue here is that
- format modules give a "whennext" sample value, which is used to
- calculate when to set a timer for to retrieve the next frame.
- However, the zaptel timer operates on 8 kHz samples, so this must
- be taken into account. (another part of issue #12164, reported by
- milazzo and jsmith, patch by me) ........
-
-2008-03-20 18:02 +0000 [r110273] Mark Michelson <mmichelson@digium.com>
-
- * main/dial.c, /: Merged revisions 110272 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r110272 |
- mmichelson | 2008-03-20 13:01:36 -0500 (Thu, 20 Mar 2008) | 3
- lines Add missing unlock ........
-
-2008-03-20 17:45 +0000 [r110269-110271] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /, res/res_musiconhold.c: Merged revisions 110268
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r110268 | russell | 2008-03-20 12:41:22 -0500 (Thu, 20
- Mar 2008) | 27 lines Add some fixes that I made in regards to
- wideband codec handling to get G.722 music on hold working for
- me. (issue #12164, reported by milazzo and jsmith, patches by me)
- res/res_musiconhold.c: - I moved a single line so that the sample
- queue update happened before ast_write(). The reason that this
- was a bug is that the G.722 frame originally says it has 320
- samples in it (which is correct). However, when the frame is
- written to a channel that uses RTP, main/rtp.c modifies the frame
- to cut the number of samples in half before it sends it on the
- wire. This is to account for the stupid incorrect G.722 spec that
- makes it so we have to lie about the number of samples with RTP.
- I should probably go and re-work the RTP code so it doesn't
- modify the frame so that a bug like this won't happen in the
- future. However, this change to MOH is harmless. main/channel.c:
- - I made two fixes in regards to generator timing. Generators use
- samples for timing. However, this code assumed 8 kHz samples. In
- one case, it was a hard coded 160 samples, that is now written as
- the sample rate / 50. The other place was dealing with timing a
- generator based on frames coming from the other direction.
- However, that would have only worked if the sample rates for the
- formats in both directions were the same. The code now takes into
- account that the sample rates may differ, and scales the
- generator samples accordingly. ........
-
-2008-03-19 23:00 +0000 [r110165] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 110164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110164 | russell | 2008-03-19 17:58:33 -0500 (Wed, 19 Mar 2008)
- | 13 lines Merged revisions 110163 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008)
- | 5 lines Fix a bug where when calls on the trunk side hang up
- while on hold, the state is not properly reflected. (closes issue
- #11990, reported by anakaoka, patched by me) ........
- ................
-
-2008-03-19 21:06 +0000 [r110088] Jeff Peeler <jpeeler@digium.com>
-
- * /: marking rev 110087 from trunk as not applying
-
-2008-03-19 20:37 +0000 [r110085] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 110084 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110084 | mmichelson | 2008-03-19 15:34:13 -0500 (Wed, 19 Mar
- 2008) | 12 lines Merged revisions 110083 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar
- 2008) | 4 lines Add a missing unlock in the case that memory
- allocation fails in app_chanspy. Thanks to Russell for confirming
- that this was an issue. ........ ................
-
-2008-03-19 19:14 +0000 [r110037] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 110036 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r110036 | file | 2008-03-19 16:13:39 -0300 (Wed,
- 19 Mar 2008) | 12 lines Merged revisions 110035 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4
- lines Add sanity checking for position resuming. We *have* to
- make sure that the position does not exceed the total number of
- files present, and we have to make sure that the position's
- filename is the same as previous. These values can change if a
- music class is reloaded and give unpredictable behavior. (closes
- issue #11663) Reported by: junky ........ ................
-
-2008-03-19 19:00 +0000 [r110024-110032] Russell Bryant <russell@digium.com>
-
- * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
- (added), /: Merged revisions 109974 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109974 | qwell | 2008-03-19 12:15:14 -0500 (Wed, 19 Mar 2008) |
- 13 lines Merged revisions 109973 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) |
- 5 lines People report bugs about Asterisk crashing with DO_CRASH
- enabled was getting a little silly... Now we only show certain
- cflags when you run configure with --enable-dev-mode
- (corresponding menuselect change to follow) ........
- ................
-
-2008-03-19 18:26 +0000 [r109971-110021] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 110020 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r110020 | file | 2008-03-19 15:25:33 -0300 (Wed, 19 Mar 2008) |
- 14 lines Merged revisions 110019 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6
- lines Make sure that the mark bit does not incorrectly cause
- video frame timestamps to be calculated as if they are audio
- frames. (closes issue #11429) Reported by: sperreault Patches:
- 11429-frametype.diff uploaded by qwell (license 4) ........
- ................
-
-2008-03-19 16:46 +0000 [r109969] Steve Murphy <murf@digium.com>
-
- * main/config.c, /: Merged revisions 109942 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109942 | murf | 2008-03-19 10:24:51 -0600 (Wed, 19 Mar 2008) |
- 80 lines Merged revisions 109908 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) |
- 72 lines (closes issue #11442) Reported by: tzafrir Patches:
- 11442.patch uploaded by murf (license 17) Tested by: murf I
- didn't give tzafrir very much time to test this, but if he does
- still have remaining issues, he is welcome to re-open this bug,
- and we'll do what is called for. I reproduced the problem, and
- tested the fix, so I hope I am not jumping by just going ahead
- and committing the fix. The problem was with what file_save does
- with templates; firstly, it tended to print out multiple options:
- [my_category](!)(templateref) instead of
- [my_category](!,templateref) which is fixed by this patch.
- Nextly, the code to suppress output of duplicate declarations
- that would occur because the reader copies inherited declarations
- down the hierarchy, was not working. Thus: [master-template](!)
- mastervar = bar [template](!,master-template) tvar = value
- [cat](template) catvar = val would be rewritten as: ;! ;!
- Automatically generated configuration file ;! Filename:
- experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
- Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
- [master-template](!) mastervar = bar
- [template](!,master-template) mastervar = bar tvar = value
- [cat](template) mastervar = bar tvar = value catvar = val This
- has been fixed. Since the config reader 'explodes' inherited vars
- into the category, users may, in certain circumstances, see
- output different from what they originally entered, but it should
- be both correct and equivalent. ........ ................
-
-2008-03-19 04:06 +0000 [r109834-109840] Russell Bryant <russell@digium.com>
-
- * /, main/utils.c: Merged revisions 109839 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109839 | russell | 2008-03-18 23:06:31 -0500 (Tue, 18 Mar 2008)
- | 10 lines Merged revisions 109838 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008)
- | 2 lines Tweak spacing in a recent change because I'm very
- picky. ........ ................
-
- * apps/app_chanspy.c, /: Merged revisions 109764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109764 | russell | 2008-03-18 17:36:02 -0500 (Tue, 18 Mar 2008)
- | 11 lines Merged revisions 109763 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008)
- | 3 lines Fix one place where the chanspy datastore isn't removed
- from a channel. (issue #12243, reported by atis, patch by me)
- ........ ................
-
-2008-03-18 23:23 +0000 [r109779] Tilghman Lesher <tlesher@digium.com>
-
- * /, configs/res_ldap.conf.sample, res/res_config_ldap.c: Merged
- revisions 109775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109775 |
- tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines
- Change back to using ldap_initialize() and let the user specify a
- URL directly, instead of trying to piece it together, badly.
- ........
-
-2008-03-18 21:03 +0000 [r109716] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 109714 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109714 | mmichelson | 2008-03-18 15:59:02 -0500 (Tue, 18 Mar
- 2008) | 20 lines Merged revisions 109713 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar
- 2008) | 12 lines This patch makes it so that all queue member
- status changes are handled through device state code. This
- removes several problems people were seeing where their queue
- members would get into an "unknown" state. Huge props go to atis
- on this one since he was the one who found the code section that
- was causing the problem and proposed the solution. I just wrote
- what he suggested :) (closes issue #12127) Reported by: atis
- Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested
- by: atis, jvandal ........ ................
-
-2008-03-18 20:14 +0000 [r109684] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 109683 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109683 | tilghman | 2008-03-18 15:13:40 -0500 (Tue, 18 Mar 2008)
- | 4 lines Set protocol version, port number correctly. (closes
- issue #12211, closes issue #12209) Reported by: sylvain ........
-
-2008-03-18 19:24 +0000 [r109654] Jason Parker <jparker@digium.com>
-
- * /, codecs/log2comp.h: Merged revisions 109651 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109651 | qwell | 2008-03-18 14:24:15 -0500 (Tue, 18 Mar 2008) |
- 15 lines Merged revisions 109648 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109648 | qwell | 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) |
- 7 lines Allow codecs that use log2comp (g726) to compile
- correctly on x86 with gcc4 optimizations. (closes issue #12253)
- Reported by: fossil Patches: log2comp.patch uploaded by fossil
- (license 140) ........ ................
-
-2008-03-18 19:00 +0000 [r109546-109622] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 109576 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r109576 | mmichelson | 2008-03-18 12:59:18 -0500
- (Tue, 18 Mar 2008) | 14 lines Merged revisions 109575 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, 18 Mar
- 2008) | 6 lines Make sure an agent doesn't try to send dtmf to a
- NULL channel closes issue #12242 Reported by Yourname ........
- ................
-
- * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar
- 2008) | 3 lines Add format attribute to printf-style functions in
- astmm.h ........
-
-2008-03-18 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta6 released.
-
-2008-03-18 17:01 +0000 [r109546] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar
- 2008) | 3 lines Add format attribute to printf-style functions in
- astmm.h ........
-
-2008-03-18 16:26 +0000 [r109487] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /: Merged revisions 109475 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109475 | kpfleming | 2008-03-18 11:23:05 -0500 (Tue, 18 Mar
- 2008) | 2 lines fix up various warnings found via the addition of
- format string checking... some of these were really, really bad
- code ........
-
-2008-03-18 15:58 +0000 [r109454-109459] Russell Bryant <russell@digium.com>
-
- * Makefile, channels/chan_misdn.c, include/asterisk/strings.h,
- res/res_indications.c, utils/extconf.c, main/asterisk.c,
- apps/app_voicemail.c, utils/check_expr.c,
- cdr/cdr_sqlite3_custom.c, apps/app_meetme.c, /,
- res/res_phoneprov.c, main/utils.c, channels/chan_iax2.c,
- utils/frame.c, main/cli.c, funcs/func_enum.c, main/manager.c,
- include/asterisk/astobj.h, res/res_agi.c, main/features.c,
- apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c,
- include/asterisk/utils.h, channels/chan_sip.c,
- apps/app_festival.c, main/translate.c, main/jitterbuf.c,
- utils/astman.c, include/jitterbuf.h, apps/app_queue.c: Merged
- revisions 109447 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109447 |
- twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines
- Go through and fix a bunch of places where character strings were
- being interpreted as format strings. Most of these changes are
- solely to make compiling with -Wsecurity and -Wformat=2 happy,
- and were not actual problems, per se. I also added format
- attributes to any printf wrapper functions I found that didn't
- have them. -Wsecurity and -Wmissing-format-attribute added to
- --enable-dev-mode. ........
-
- * configs/sip_notify.conf.sample, /: Merged revisions 109111 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r109111 | qwell | 2008-03-17 11:37:31 -0500 (Mon, 17 Mar
- 2008) | 10 lines Add sample events for aastra phones.
- aastra-check-cfg is the same as the other check-cfg entries, and
- aastra-xml is to load a pre-configured xml script. (closes issue
- #12229) Reported by: gowen72 Patches: aastra.patch uploaded by
- gowen72 (license 432) ........
-
-2008-03-18 15:50 +0000 [r109453] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, acinclude.m4:
- Merged revisions 109451 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109451 |
- kpfleming | 2008-03-18 10:50:29 -0500 (Tue, 18 Mar 2008) | 2
- lines ensure that dependencies on AST_C_DEFINE_CHECK symbols work
- properly ........
-
-2008-03-18 15:50 +0000 [r109448-109452] Russell Bryant <russell@digium.com>
-
- * main/dial.c, /: Merged revisions 108962 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108962 | mvanbaak | 2008-03-16 16:50:58 -0500 (Sun, 16 Mar 2008)
- | 15 lines Merged revisions 108961 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008)
- | 7 lines add missing break to case AST_CONTROL_SRCUPDATE (closes
- issue #12228) Reported by: andrew Patches: SRC.patch uploaded by
- andrew (license 240) ........ ................
-
-2008-03-18 15:16 +0000 [r109398] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c, /, main/logger.c: Merged revisions 109396 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r109396 | file | 2008-03-18 12:13:07 -0300 (Tue, 18 Mar
- 2008) | 3 lines Make sure values are interpreted as character
- strings and not format strings. (AST-2008-004) ........
-
-2008-03-18 15:14 +0000 [r109397] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ael-ntest23 (added),
- pbx/ael/ael-test/ael-ntest23/t1/a.ael,
- pbx/ael/ael-test/ael-ntest23/t1/b.ael,
- pbx/ael/ael-test/ael-ntest23/t1/c.ael,
- pbx/ael/ael-test/ael-ntest23/t2/d.ael,
- pbx/ael/ael-test/ael-ntest23/t2/e.ael,
- pbx/ael/ael-test/ael-ntest23/t2/f.ael, res/ael/ael_lex.c,
- pbx/ael/ael-test/ref.ael-ntest23 (added),
- pbx/ael/ael-test/ael-ntest23/t3/g.ael,
- pbx/ael/ael-test/ael-ntest23/t3/h.ael,
- pbx/ael/ael-test/ael-ntest23/t3/i.ael, res/ael/ael.flex,
- pbx/ael/ael-test/ael-ntest23/t3/j.ael,
- pbx/ael/ael-test/ael-ntest23/qq.ael,
- pbx/ael/ael-test/ael-ntest23/t1, pbx/ael/ael-test/ael-ntest23/t2,
- pbx/ael/ael-test/ael-ntest23/t3, /,
- pbx/ael/ael-test/ael-ntest23/extensions.ael: Merged revisions
- 109357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109357 | murf | 2008-03-18 08:09:50 -0600 (Tue, 18 Mar 2008) |
- 25 lines Merged revisions 109309 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) |
- 17 lines (closes issue #11903) Reported by: atis Many thanks to
- atis for spotting this problem and reporting it. The fix was to
- straighten out how items are placed on and removed from the file
- stack. Regressions as well as the provided test case helped to
- straighten out all code paths. valgrind was used to make sure all
- memory allocated was freed. Sorry for not solving this earlier. I
- got distracted. Added the ntest23 regression test, which is
- mainly a copy of ntest22, but with a few juicy errors thrown in,
- to replicate the kind of error that atis spotted. ........
- ................
-
-2008-03-18 15:11 +0000 [r109395] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 109389 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109389 |
- qwell | 2008-03-18 10:07:04 -0500 (Tue, 18 Mar 2008) | 3 lines Do
- not return with a successful authentication if the From header
- ends up empty. (AST-2008-003) ........
-
-2008-03-18 15:09 +0000 [r109392] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /, channels/chan_sip.c: Merged revisions 109390 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r109390 | file | 2008-03-18 12:08:09 -0300 (Tue,
- 18 Mar 2008) | 11 lines Merged revisions 109386 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3
- lines Put a maximum limit on the number of payloads accepted, and
- also make sure a given payload does not exceed our maximum value.
- (AST-2008-002) ........ ................
-
-2008-03-18 00:40 +0000 [r109283] Sean Bright <sean.bright@gmail.com>
-
- * /, configure, configure.ac: Merged revisions 109282 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109282 | seanbright | 2008-03-17 20:28:39 -0400 (Mon, 17 Mar
- 2008) | 1 line Fix a typo ........
-
-2008-03-17 22:24 +0000 [r109254] Terry Wilson <twilson@digium.com>
-
- * build_tools/cflags.xml, /, build_tools/menuselect-deps.in,
- configure, include/asterisk/autoconfig.h.in, main/Makefile,
- configure.ac, main/http.c, main/minimime (removed),
- build_tools/make_buildopts_h, makeopts.in: Merged revisions
- 109229 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109229 |
- twilson | 2008-03-17 17:10:06 -0500 (Mon, 17 Mar 2008) | 5 lines
- Replace minimime with superior GMime library so that the entire
- contents of an http post are not read into memory. This does
- introduce a dependency on the GMime library for handling HTTP
- POSTs, but it is available in most distros. If the library is
- present, then the compile flag for ENABLE_UPLOADS is enabled by
- default in menuselect. ........
-
-2008-03-17 22:07 +0000 [r109228] Mark Michelson <mmichelson@digium.com>
-
- * /, main/utils.c: Merged revisions 109227 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109227 | mmichelson | 2008-03-17 17:06:44 -0500 (Mon, 17 Mar
- 2008) | 20 lines Merged revisions 109226 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar
- 2008) | 12 lines Fix a logic flaw in the code that stores lock
- info which is displayed via the "core show locks" command. The
- idea behind this section of code was to remove the previous lock
- from the list if it was a trylock that had failed. Unfortunately,
- instead of checking the status of the previous lock, we were
- referencing the index immediately following the previous lock in
- the lock_info->locks array. The result of this problem, under the
- right circumstances, was that the lock which we currently in the
- process of attempting to acquire could "overwrite" the previous
- lock which was acquired. While this does not in any way affect
- typical operation, it *could* lead to misleading "core show
- locks" output. ........ ................
-
-2008-03-17 18:11 +0000 [r109175] Michiel van Baak <michiel@vanbaak.info>
-
- * /, channels/chan_skinny.c: Merged revisions 109168 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r109168 | mvanbaak | 2008-03-17 18:43:46 +0100 (Mon, 17 Mar 2008)
- | 11 lines Update the directory of placed calls on skinny phones
- when dialing a channel that does not provide progress (analog ZAP
- lines) The phone does handle the double update on calls to
- channels that do provide progress and wont insert duplicate items
- (closes issue #12239) Reported by: DEA Patches:
- chan_skinny-call-log.txt uploaded by DEA (license 3) ........
-
-2008-03-17 17:42 +0000 [r109167] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /, configure, configure.ac, acinclude.m4: Merged
- revisions 109166 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r109166 |
- kpfleming | 2008-03-17 12:31:46 -0500 (Mon, 17 Mar 2008) | 3
- lines don't define Zaptel features as libraries, they aren't, and
- we don't want '--with-zaptel-<foo>' configure options for them
- also some minor cleanups ........
-
-2008-03-17 16:47 +0000 [r109109-109114] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 109108 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109108 | file | 2008-03-17 13:26:36 -0300 (Mon, 17 Mar 2008) |
- 12 lines Merged revisions 109107 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4
- lines 200 OKs in response to a reinvite need to be sent reliably.
- If the remote side does not receive one the dialog will be torn
- down. (closes issue #12208) Reported by: atrash ........
- ................
-
-2008-03-17 14:21 +0000 [r109027] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 109024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r109024 | mmichelson | 2008-03-17 09:21:14 -0500 (Mon, 17 Mar
- 2008) | 14 lines Merged revisions 109012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r109012 | mmichelson | 2008-03-17 09:18:26 -0500 (Mon, 17 Mar
- 2008) | 6 lines Make sure that we release the lock on the spyee
- channel if the spyee or spy has hung up (closes issue #12232)
- Reported by: atis ........ ................
-
-2008-03-16 17:56 +0000 [r108928-108930] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 108927 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r108927 | russell | 2008-03-16 12:53:46 -0500 (Sun, 16 Mar 2008)
- | 7 lines Fix polling for mailbox changes in mailboxes that are
- not in the default vm context. (closes issue #12223) Reported by:
- DEA Patches: vm-polled-imap.txt uploaded by DEA (license 3)
- ........
-
-2008-03-15 16:21 +0000 [r108741-108895] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 108799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r108799 |
- russell | 2008-03-14 15:14:06 -0500 (Fri, 14 Mar 2008) | 8 lines
- Make sure configure is run before menuselect on a clean checkout
- (closes issue #12197) Reported by: juggie Patches: 12197.diff
- uploaded by juggie (license 24) ........
-
- * channels/chan_oss.c, /: Merged revisions 108797 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108797 | russell | 2008-03-14 15:09:37 -0500 (Fri, 14 Mar 2008)
- | 13 lines Merged revisions 108796 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008)
- | 5 lines Fix a channel name issue. chan_oss registers the
- "Console" channel type, but it created channels with an "OSS"
- prefix. (closes issue #12194, reported by davidw, patched by me)
- ........ ................
-
- * contrib/init.d/rc.suse.asterisk, /: Merged revisions 108793 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r108793 | russell | 2008-03-14 15:04:56 -0500
- (Fri, 14 Mar 2008) | 12 lines Merged revisions 108792 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108792 | russell | 2008-03-14 15:04:35 -0500 (Fri, 14 Mar 2008)
- | 4 lines Update the SuSE init script to start networking before
- asterisk, as well. (closes issue #12200, reported by and change
- suggested by reinerotto) ........ ................
-
- * /, configure, acinclude.m4: Merged revisions 108740 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r108740 | russell | 2008-03-14 12:05:11 -0500 (Fri, 14 Mar 2008)
- | 5 lines Do a link test in AST_EXT_TOOL_CHECK() to ensure we
- have all the required libs reported by the tool. (closes issue
- #12067, reported by Juggie, patched by me) ........
-
-2008-03-14 16:54 +0000 [r108739] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 108738 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108738 | mmichelson | 2008-03-14 11:52:51 -0500 (Fri, 14 Mar
- 2008) | 41 lines Merged revisions 108737 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar
- 2008) | 33 lines Fix a race condition in the SIP packet scheduler
- which could cause a crash. chan_sip uses the scheduler API in
- order to schedule retransmission of reliable packets (such as
- INVITES). If a retransmission of a packet is occurring, then the
- packet is removed from the scheduler and retrans_pkt is called.
- Meanwhile, if a response is received from the packet as
- previously transmitted, then when we ACK the response, we will
- remove the packet from the scheduler and free the packet. The
- problem is that both the ACK function and retrans_pkt attempt to
- acquire the same lock at the beginning of the function call. This
- means that if the ACK function acquires the lock first, then it
- will free the packet which retrans_pkt is about to read from and
- write to. The result is a crash. The solution: 1. If the ACK
- function fails to remove the packet from the scheduler and the
- retransmit id of the packet is not -1 (meaning that we have not
- reached the maximum number of retransmissions) then release the
- lock and yield so that retrans_pkt may acquire the lock and
- operate. 2. Make absolutely certain that the ACK function does
- not recursively lock the lock in question. If it does, then
- releasing the lock will do no good, since retrans_pkt will still
- be unable to acquire the lock. (closes issue #12098) Reported by:
- wegbert (closes issue #12089) Reported by: PTorres Patches:
- 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested
- by: jvandal ........ ................
-
-2008-03-14 14:33 +0000 [r108684] Jason Parker <jparker@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 108683 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r108683 | qwell | 2008-03-14 09:32:55 -0500
- (Fri, 14 Mar 2008) | 12 lines Merged revisions 108682 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108682 | qwell | 2008-03-14 09:29:05 -0500 (Fri, 14 Mar 2008) |
- 4 lines Fix a potential segfault if chan (or chan->music_state)
- is NULL. Closes issue #12210, credit to edantie for pointing this
- out. ........ ................
-
-2008-03-13 21:48 +0000 [r108587] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 108586 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r108586 |
- mmichelson | 2008-03-13 16:47:55 -0500 (Thu, 13 Mar 2008) | 3
- lines Make this compile ........
-
-2008-03-13 21:41 +0000 [r108585] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, main/channel.c, /,
- include/asterisk/channel.h: Merged revisions 108584 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r108584 | russell | 2008-03-13 16:40:43 -0500
- (Thu, 13 Mar 2008) | 19 lines Merged revisions 108583 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008)
- | 11 lines Fix another issue that was causing crashes in chanspy.
- This introduces a new datastore callback, called chan_fixup().
- The concept is exactly like the fixup callback that is used in
- the channel technology interface. This callback gets called when
- the owning channel changes due to a masquerade. Before this was
- introduced, if a masquerade happened on a channel being spyed on,
- the channel pointer in the datastore became invalid. (closes
- issue #12187) (reported by, and lots of testing from atis) (props
- to file for the help with ideas) ........ ................
-
-2008-03-13 21:31 +0000 [r108582] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 108529 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r108529 |
- mmichelson | 2008-03-13 15:59:00 -0500 (Thu, 13 Mar 2008) | 11
- lines Fixing a potential buffer overflow in the manager command
- ModuleCheck. Though this overflow is exploitable remotely, we are
- NOT issuing a security advisory for this since in order to
- exploit the overflow, the attacker would have to establish an
- authenticated manager session AND have the system privilege. By
- gaining this privilege, the attacker already has more powerful
- weapons at his disposal than overflowing a buffer with a
- malformed manager header, so the vulnerability in this case
- really lies with the authentication method that allowed the
- attacker to gain the system privilege in the first place.
- ........
-
-2008-03-13 21:07 +0000 [r108347-108532] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 108531 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008)
- | 18 lines Merged revisions 108530 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008)
- | 10 lines Make a tweak that gets the LEDs on polycom phones to
- blink when an extension that has been subscribed to goes on hold.
- Otherwise, they just stay on like it does when an extension is in
- use. (closes issue #11263) Reported by: russell Patches:
- notify_hold.rev1.txt uploaded by russell (license 2) Tested by:
- russell ........ ................
-
- * apps/app_voicemail.c, /: Merged revisions 108508 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r108508 | russell | 2008-03-13 15:35:28 -0500 (Thu, 13 Mar 2008)
- | 2 lines Fix a place where configuration values could cause an
- overflow of a buffer. ........
-
- * /, apps/app_followme.c: Merged revisions 108472 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108472 | russell | 2008-03-13 15:26:59 -0500 (Thu, 13 Mar 2008)
- | 12 lines Merged revisions 108469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008)
- | 4 lines Fix a couple uses of sprintf. The second one could
- actually cause an overflow of a stack buffer. It's not a security
- issue though, it only depends on your configuration. ........
- ................
-
- * /, main/features.c: Merged revisions 107465 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107465 |
- file | 2008-03-11 10:05:17 -0500 (Tue, 11 Mar 2008) | 4 lines
- Clarify comment about masquerading and playback of the parking
- slot. (closes issue #12180) Reported by: davidw ........
-
- * /, channels/chan_sip.c: Merged revisions 107157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107157 |
- file | 2008-03-10 15:00:21 -0500 (Mon, 10 Mar 2008) | 4 lines If
- we receive a 488 on a T38 request reinvite back to audio. As well
- reinvite across a bridge back to audio if one side doesn't
- negotiate to T38. (closes issue #8677) Reported by: alex-911
- ........
-
- * /: Merged revisions 106892 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106892 |
- mattf | 2008-03-07 16:36:49 -0600 (Fri, 07 Mar 2008) | 1 line
- Make sure we don't start a call when we have already done so in
- response to a COT message ........
-
- * /, main/editline/Makefile.in: Merged revisions 106843 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r106843 | qwell | 2008-03-07 16:15:20 -0600
- (Fri, 07 Mar 2008) | 13 lines Merged revisions 106842 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) |
- 5 lines Fix hardcoded grep in editline, were GNU grep is
- required. (closes issue #12124) Reported by: dmartin ........
- ................
-
- * include/asterisk/http.h, main/tcptls.c, main/manager.c, /,
- channels/chan_sip.c, res/res_phoneprov.c, main/http.c,
- include/asterisk/tcptls.h: Merged revisions 108295 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r108295 | russell | 2008-03-12 17:13:18 -0500 (Wed, 12 Mar 2008)
- | 3 lines Rename ast_tcptls_server_instance to session_instance,
- since this pertains to server and client usage. ........
-
- * /, main/http.c: Merged revisions 108346 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r108346 |
- russell | 2008-03-12 17:49:26 -0500 (Wed, 12 Mar 2008) | 4 lines
- Make the default prefix empty, like it was in Asterisk 1.4.
- (closes issue #12198, reported by bkruse, patched by me) ........
-
-2008-03-12 22:10 +0000 [r108246-108294] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 108293 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r108293 |
- mmichelson | 2008-03-12 17:09:52 -0500 (Wed, 12 Mar 2008) | 3
- lines Let's get this to compile ........
-
- * /, channels/chan_sip.c: Merged revisions 108289 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar
- 2008) | 22 lines Merged revisions 108288 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar
- 2008) | 14 lines Change AST_SCHED_DEL use to ast_sched_del for
- autocongestion in chan_sip. The scheduler callback will always
- return 0. This means that this id is never rescheduled, so it
- makes no sense to loop trying to delete the id from the scheduler
- queue. If we fail to remove the item from the queue once, it will
- fail every single time. (Yes I realize that in this case, the
- macro would exit early because the id is set to -1 in the
- callback, but it still makes no sense to use that macro in favor
- of calling ast_sched_del once and being done with it) This is the
- first of potentially several such fixes. ........
- ................
-
- * /, include/asterisk/sched.h: Merged revisions 108238 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r108238 | mmichelson | 2008-03-12 16:19:30 -0500
- (Wed, 12 Mar 2008) | 20 lines Merged revisions 108227 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar
- 2008) | 12 lines Added a large comment before the AST_SCHED_DEL
- macro to explain its purpose as well as when it is appropriate
- and when it is not appropriate to use it. I also removed the part
- of the debug message that mentions that this is probably a bug
- because there are some perfectly legitimate places where
- ast_sched_del may fail to delete an entry (e.g. when the
- scheduler callback manually reschedules with a new id instead of
- returning non-zero to tell the scheduler to reschedule with the
- same idea). I also raised the debug level of the debug message in
- AST_SCHED_DEL since it seems like it could come up quite
- frequently since the macro is probably being used in several
- places where it shouldn't be. Also removed the redundant line,
- file, and function information since that is provided by ast_log.
- ........ ................
-
-2008-03-12 20:29 +0000 [r108205] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 108191 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar
- 2008) | 14 lines Merged revisions 108086 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar
- 2008) | 6 lines if we receive an INVITE with a Content-Length
- that is not a valid number, or is zero, then don't process the
- rest of the message body looking for an SDP closes issue #11475
- Reported by: andrebarbosa ........ ................
-
-2008-03-12 19:59 +0000 [r108138] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, main/channel.c, /: Merged revisions 108137
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r108137 | russell | 2008-03-12 14:59:05 -0500
- (Wed, 12 Mar 2008) | 48 lines Merged revisions 108135 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008)
- | 40 lines (closes issue #12187, reported by atis, fixed by me
- after some brainstorming on the issue with mmichelson) - Update
- copyright info on app_chanspy. - Fix a race condition that caused
- app_chanspy to crash. The issue was that the chanspy datastore
- magic that was used to ensure that spyee channels did not
- disappear out from under the code did not completely solve the
- problem. It was actually possible for chanspy to acquire a
- channel reference out of its datastore to a channel that was in
- the middle of being destroyed. That was because datastore
- destruction in ast_channel_free() was done near the end. So, this
- left the code in app_chanspy accessing a channel that was
- partially, or completely invalid because it was in the process of
- being free'd by another thread. The following sort of shows the
- code path where the race occurred:
- =============================================================================
- Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
- --------------------------------------||-------------------------------------
- ast_channel_free() || - remove channel from channel list || -
- lock/unlock the channel to ensure || that no references retrieved
- from || the channel list exist. ||
- --------------------------------------||-------------------------------------
- || channel_spy() - destroy some channel data || - Lock chanspy
- datastore || - Retrieve reference to channel || - lock channel ||
- - Unlock chanspy datastore
- --------------------------------------||-------------------------------------
- - destroy channel datastores || - call chanspy datastore d'tor ||
- which NULL's out the ds' || - Operate on the channel ...
- reference to the channel || || - free the channel || || || -
- unlock the channel
- --------------------------------------||-------------------------------------
- =============================================================================
- ........ ................
-
-2008-03-12 18:31 +0000 [r108085] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, /, include/asterisk/audiohook.h,
- main/audiohook.c: Merged revisions 108084 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) |
- 12 lines Merged revisions 108083 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4
- lines Add a trigger mode that triggers on both read and write.
- The actual function that returns the combined audio frame though
- will wait until both sides have fed in audio, or until one side
- stops (such as the case when you call Wait). (closes issue
- #11945) Reported by: xheliox ........ ................
-
-2008-03-12 17:03 +0000 [r108033] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 108032 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r108032 | russell | 2008-03-12 12:02:57 -0500 (Wed, 12 Mar 2008)
- | 12 lines Merged revisions 108031 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r108031 | russell | 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008)
- | 4 lines Destroy the channel lock after the channel datastores.
- (inspired by issue #12187) ........ ................
-
-2008-03-12 07:44 +0000 [r107879-107999] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 107998 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107998 |
- tilghman | 2008-03-12 02:43:03 -0500 (Wed, 12 Mar 2008) | 7 lines
- Deadlock fixes (closes issue #12143) Reported by: kactus Patches:
- 20080312__bug12143__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: kactus ........
-
- * main/loader.c, /, apps/app_dumpchan.c, apps/app_zapras.c: Merged
- revisions 107960 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107960 |
- tilghman | 2008-03-12 00:46:39 -0500 (Wed, 12 Mar 2008) | 4 lines
- Revert several changes from revision 102525, as the changes were
- not compatible, and, in fact, introduced regressions. (Closes
- issue #12190) ........
-
- * contrib/scripts/iax-friends.sql, /,
- contrib/scripts/sip-friends.sql: Merged revisions 107878 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r107878 | tilghman | 2008-03-11 20:54:00 -0500
- (Tue, 11 Mar 2008) | 10 lines Merged revisions 107877 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107877 | tilghman | 2008-03-11 20:52:40 -0500 (Tue, 11 Mar 2008)
- | 2 lines Document all of the possible realtime fields ........
- ................
-
-2008-03-11 23:38 +0000 [r107828] Jason Parker <jparker@digium.com>
-
- * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 107827 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r107827 | qwell | 2008-03-11 18:38:00 -0500
- (Tue, 11 Mar 2008) | 15 lines Merged revisions 107826 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) |
- 7 lines Update documentation for pgsql ODBC voicemail. (closes
- issue #12186) Reported by: jsmith Patches:
- vm_pgsql_doc_update.patch uploaded by jsmith (license 15)
- ........ ................
-
-2008-03-11 22:59 +0000 [r107723-107793] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_sqlite.c, main/config.c, res/res_config_curl.c,
- res/res_config_pgsql.c, res/res_config_odbc.c, /,
- include/asterisk/config.h, res/res_config_ldap.c: Merged
- revisions 107791 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107791 |
- tilghman | 2008-03-11 17:55:16 -0500 (Tue, 11 Mar 2008) | 5 lines
- An offhand comment from Russell made me realize that the
- configuration file caching would not work properly for users.conf
- and any other file read from more than one place. I needed to add
- the filename which requested the config file to get it to work
- properly. ........
-
-2008-03-11 20:54 +0000 [r107720] Jason Parker <jparker@digium.com>
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
- revisions 107718 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107718 | qwell | 2008-03-11 15:53:48 -0500 (Tue, 11 Mar 2008) |
- 13 lines Merged revisions 107714 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) |
- 5 lines Copy voicemail dependency logic for res_adsi to
- chan_gtalk and chan_jingle (for jabber). (closes issue #12014)
- Reported by: junky ........ ................
-
-2008-03-11 20:51 +0000 [r107716] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, Makefile.rules, channels/Makefile: Merged revisions 107715 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r107715 | kpfleming | 2008-03-11 15:50:57 -0500
- (Tue, 11 Mar 2008) | 10 lines Merged revisions 107713 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar
- 2008) | 2 lines get chan_vpb to build properly in dev mode
- ........ ................
-
-2008-03-11 20:37 +0000 [r107584-107711] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_page.c: Merged revisions 107710 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107710 |
- file | 2008-03-11 17:36:14 -0300 (Tue, 11 Mar 2008) | 6 lines
- Dial a device even if it's state is unknown. (closes issue
- #12184) Reported by: bluecrow76 Patches:
- asterisk-svn-app_page.c.devicestate_unknown.diff uploaded by
- bluecrow76 (license 270) ........
-
- * /, main/features.c: Merged revisions 107659 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107659 | file | 2008-03-11 16:23:28 -0300 (Tue, 11 Mar 2008) |
- 12 lines Merged revisions 107646 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107646 | file | 2008-03-11 16:20:01 -0300 (Tue, 11 Mar 2008) | 4
- lines Make sure the visible indication is on the right channel so
- when the masquerade happens the proper indication is enacted.
- (closes issue #11707) Reported by: iam ........ ................
-
- * /, apps/app_meetme.c: Merged revisions 107638 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107638 | file | 2008-03-11 15:48:59 -0300 (Tue, 11 Mar 2008) |
- 12 lines Merged revisions 107637 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4
- lines Add an additional check for setting conference parameter
- when using the marked user options. It was possible for it to
- return to a no listen/no talk state if a masquerade happened.
- (closes issue #12136) Reported by: aragon ........
- ................
-
-2008-03-11 15:39 +0000 [r107374-107526] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_vpb.cc, /: Merged revisions 107525 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r107525 | kpfleming | 2008-03-11 10:39:37 -0500 (Tue, 11 Mar
- 2008) | 2 lines fix another potential bug found by gcc 4.3
- ........
-
- * apps/app_rpt.c, channels/misdn/isdn_lib.c, codecs/Makefile, /,
- apps/app_sms.c: Merged revisions 107466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107466 | kpfleming | 2008-03-11 10:13:38 -0500 (Tue, 11 Mar
- 2008) | 10 lines Merged revisions 107464 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar
- 2008) | 2 lines fix various other problems found by gcc 4.3
- ........ ................
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- apps/app_sms.c: Merged revisions 107462 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107462 | kpfleming | 2008-03-11 09:37:03 -0500 (Tue, 11 Mar
- 2008) | 10 lines Merged revisions 107461 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107461 | kpfleming | 2008-03-11 09:33:45 -0500 (Tue, 11 Mar
- 2008) | 2 lines stop checking for mktime() in the configure
- script... we don't use it, and the test is buggy under gcc 4.3
- ........ ................
-
- * /, configure, main/Makefile, configure.ac, makeopts.in: Merged
- revisions 107409 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107409 | kpfleming | 2008-03-11 09:09:49 -0500 (Tue, 11 Mar
- 2008) | 13 lines Merged revisions 107408 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar
- 2008) | 5 lines check for compiler support for
- -fno-strict-overflow before using it (tested with Debian's gcc
- 4.3, 4.1 and 3.4) (closes issue #12179) Reported by: Netview
- ........ ................
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 107406 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107406 | kpfleming | 2008-03-11 08:58:37 -0500 (Tue, 11 Mar
- 2008) | 10 lines Merged revisions 107405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107405 | kpfleming | 2008-03-11 08:57:08 -0500 (Tue, 11 Mar
- 2008) | 2 lines fix small bug in IMAP toolkit testing ........
- ................
-
- * main/udptl.c, utils/Makefile, /, main/Makefile,
- main/editline/readline.c, res/Makefile: Merged revisions 107373
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r107373 | kpfleming | 2008-03-11 06:36:51 -0500
- (Tue, 11 Mar 2008) | 19 lines Merged revisions 107352 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar
- 2008) | 11 lines fix up various compiler warnings found with
- gcc-4.3: - the output of flex includes a static function called
- 'input' that is not used, so for the moment we'll stop having the
- compiler tell us about unused variables in the flex source files
- (a better fix would be to improve our flex post-processing to
- remove the unused function) - main/stdtime/localtime.c makes
- assumptions about signed integer overflow, and gcc-4.3's improved
- optimizer tries to take advantage of handling potential overflow
- conditions at compile time; for now, suppress these optimizations
- until we can fiure out if the code needs improvement -
- main/udptl.c has some references to uninitialized variables; in
- one case there was no bug, but in the other it was certainly
- possibly for unexpected behavior to occur -
- main/editline/readline.c had an unused variable ........
- ................
-
-2008-03-11 01:27 +0000 [r107336] Terry Wilson <twilson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 107292 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107292 | twilson | 2008-03-10 20:09:46 -0500 (Mon, 10 Mar 2008)
- | 10 lines Merged revisions 107290 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008)
- | 2 lines If we fail to alloc a channel, we should re-lock the
- pvt structure before returning. ........ ................
-
-2008-03-10 23:46 +0000 [r107289] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 107019 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107019 |
- murf | 2008-03-10 08:55:21 -0600 (Mon, 10 Mar 2008) | 1 line way
- back in July, in r.75706, a fix was made ot the strftime usages,
- which was good, but in this case, the check for a nil time was
- accidentally removed, and now it is restored, to keep timevals
- like '1969-12-31 17:00:00' from showing up in the cdrs. No idea
- what databases will do with this. No bugs filed as yet, but it
- felt like a bug. ........
-
-2008-03-10 20:29 +0000 [r107180] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 107177 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107177 | qwell | 2008-03-10 15:28:33 -0500 (Mon, 10 Mar 2008) |
- 13 lines Merged revisions 107173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) |
- 5 lines Make sure to reenable echo can after a "failed"
- (canceled, etc) three-way call. (closes issue #11335) Reported
- by: rebuild ........ ................
-
-2008-03-10 20:18 +0000 [r107101-107163] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /: Merged revisions 107162 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107162 | russell | 2008-03-10 15:17:37 -0500 (Mon, 10 Mar 2008)
- | 16 lines Merged revisions 107161 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008)
- | 8 lines Fix another bug specifically related to asynchronous
- call origination. Once the PBX is started on the channel using
- ast_pbx_start(), then the ownership of the channel has been
- passed on to another thread. We can no longer access it in this
- code. If the channel gets hung up very quickly, it is possible
- that we could access a channel that has been free'd. (inspired by
- BE-386) ........ ................
-
- * main/pbx.c, /: Merged revisions 107159 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107159 | russell | 2008-03-10 15:05:12 -0500 (Mon, 10 Mar 2008)
- | 17 lines Merged revisions 107158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008)
- | 9 lines Fix some bugs related to originating calls. If the code
- failed to start a PBX on the channel (such as if you set a call
- limit based on the system's load average), then there were cases
- where a channel that has already been free'd using ast_hangup()
- got accessed. This caused weird memory corruption and crashes to
- occur. (fixes issue BE-386) (much debugging credit goes to
- twilson, final patch written by me) ........ ................
-
- * main/channel.c, /: Merged revisions 107103 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107103 | russell | 2008-03-10 12:13:34 -0500 (Mon, 10 Mar 2008)
- | 10 lines Merged revisions 107102 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107102 | russell | 2008-03-10 12:13:17 -0500 (Mon, 10 Mar 2008)
- | 2 lines Resolve a compiler warning. ........ ................
-
- * main/channel.c, /: Merged revisions 107100 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107100 | russell | 2008-03-10 11:59:13 -0500 (Mon, 10 Mar 2008)
- | 11 lines Merged revisions 107099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107099 | russell | 2008-03-10 11:58:57 -0500 (Mon, 10 Mar 2008)
- | 3 lines Fix a race condition where the generator can go away
- (closes issue #12175, reported by edantie, patched by me)
- ........ ................
-
-2008-03-10 15:46 +0000 [r107069] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 107068 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r107068 |
- mmichelson | 2008-03-10 10:45:13 -0500 (Mon, 10 Mar 2008) | 10
- lines app_queue has now been doxygenified thanks to snuffy! The
- ony thing I changed was the way that locks are referenced, since
- the old 1.2 names were still used in the comments. (closes issue
- #11997) Reported by: snuffy Patches: bug_11997_queue_doxy.diff
- uploaded by snuffy (license 35) ........
-
-2008-03-10 14:38 +0000 [r107018] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, main/cdr.c, /, include/asterisk/cdr.h: Merged
- revisions 107017 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r107017 | file | 2008-03-10 11:36:16 -0300 (Mon, 10 Mar 2008) |
- 15 lines Merged revisions 107016 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7
- lines Move where unanswered CDRs are dropped to the CDR core, not
- everything uses app_dial. (closes issue #11516) Reported by: ys
- Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested
- by: anest, jcapp, dartvader ........ ................
-
-2008-03-08 17:54 +0000 [r106997] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we don't start a call on a channel
- that has already started a call
-
-2008-03-08 16:14 +0000 [r106947] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 106946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106946 | kpfleming | 2008-03-08 10:03:48 -0600 (Sat, 08 Mar
- 2008) | 10 lines Merged revisions 106945 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar
- 2008) | 2 lines don't generate D-Channel "up" and "down" messages
- unless the channel state is actually changing; also, generate the
- "up" message when an implicit "up" occurs due to reception of a
- normal event when we thought the channel was "down" ........
- ................
-
-2008-03-07 22:53 +0000 [r106897] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 106896 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106896 | russell | 2008-03-07 16:52:46 -0600 (Fri, 07 Mar 2008)
- | 10 lines Merged revisions 106895 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008)
- | 2 lines Only start the SLA thread if SLA has actually been
- configured. ........ ................
-
-2008-03-07 19:34 +0000 [r106790] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 106789 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106789 | file | 2008-03-07 15:33:09 -0400 (Fri, 07 Mar 2008) |
- 12 lines Merged revisions 106788 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106788 | file | 2008-03-07 15:32:00 -0400 (Fri, 07 Mar 2008) | 4
- lines Ignore source update control frame. (closes issue #12168)
- Reported by: plack ........ ................
-
-2008-03-07 17:18 +0000 [r106686-106713] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/sched.h: Merged revisions 106707 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r106707 | russell | 2008-03-07 11:17:30 -0600
- (Fri, 07 Mar 2008) | 16 lines Merged revisions 106704 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008)
- | 8 lines Change a warning message to a debug message. This is
- happening quite frequently, and it is not worth spamming users
- with these messages unless we are pretty confident that it should
- never happen. As it stands today, it _will_ and _does_ happen and
- until that gets cleaned up a reasonable amount on the development
- side, let's not spam the logs of everyone else. (closes issue
- #12154) ........ ................
-
- * doc/smdi.txt, /: Merged revisions 106684 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106684 |
- russell | 2008-03-07 10:31:48 -0600 (Fri, 07 Mar 2008) | 2 lines
- fix example usage ........
-
-2008-03-07 16:27 +0000 [r106554-106662] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 106654 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r106654 | tilghman | 2008-03-07 10:26:07 -0600
- (Fri, 07 Mar 2008) | 11 lines Merged revisions 106635 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106635 | tilghman | 2008-03-07 10:22:11 -0600 (Fri, 07 Mar 2008)
- | 3 lines Warn the user when a temporary greeting exists (Closes
- issue #11409) ........ ................
-
- * main/rtp.c, /: Merged revisions 106607 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106607 | tilghman | 2008-03-07 09:22:34 -0600 (Fri, 07 Mar 2008)
- | 11 lines Merged revisions 106606 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008)
- | 3 lines Properly initialize rtp->schedid (Closes issue #12154)
- ........ ................
-
- * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c,
- apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c,
- funcs/func_enum.c, channels/chan_misdn.c, main/frame.c, /,
- channels/chan_sip.c, funcs/func_odbc.c, funcs/func_strings.c,
- utils/extconf.c: Merged revisions 106553 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106553 | tilghman | 2008-03-07 00:54:47 -0600 (Fri, 07 Mar 2008)
- | 14 lines Merged revisions 106552 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008)
- | 6 lines Safely use the strncat() function. (closes issue
- #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt
- uploaded by Corydon76 (license 14) ........ ................
-
-2008-03-07 01:19 +0000 [r106502-106520] Russell Bryant <russell@digium.com>
-
- * doc/smdi.txt, /: Merged revisions 106518 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106518 |
- russell | 2008-03-06 19:19:02 -0600 (Thu, 06 Mar 2008) | 1 line
- minor text changes ........
-
- * doc/smdi.txt, /: Merged revisions 106507 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106507 |
- russell | 2008-03-06 19:15:36 -0600 (Thu, 06 Mar 2008) | 2 lines
- Add updated SMDI documentation that I had only sitting in my
- email ... oops ........
-
- * main/rtp.c, codecs/codec_g722.c, /, formats/format_pcm.c,
- main/file.c: Merged revisions 106501 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106501 |
- russell | 2008-03-06 18:24:58 -0600 (Thu, 06 Mar 2008) | 28 lines
- Merge changes from team/russell/g722-sillyness ... Fix a number
- of other places where the number of samples in a G722 frame was
- not properly handled because of various reasons. main/rtp.c: -
- When a G722 frame is read from the smoother, the number of
- samples in the frame must be divided by 2 before being sent out
- over the network. Even though G722 is 16 kHz, an error in some
- previous spec has made it so that we have to list the number of
- samples such as if it was 8 kHz. main/file.c: - When scheduling
- the next time to expect a frame, take into account that the
- format of the file we're reading from may not be 8 kHz.
- codecs/codec_g722.c: - When converting from G722 to slinear,
- g722_decode() expects its samples parameter to be in the silly
- (real samples / 2) format. Make it so. - When converting from
- slinear to G722, properly set the number of samples in the frame
- to be the number of bytes of output * 2. formats/format_pcm.c: -
- This format module handles G722, among a number of other formats.
- However, the read() and seek() functions did not account for the
- fact that G722 has 2 samples per byte. (closes issue #12130,
- reported by rickross, patched by me) ........
-
-2008-03-06 22:16 +0000 [r106442] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, /: Merged revisions 106438 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106438 | mmichelson | 2008-03-06 16:11:26 -0600 (Thu, 06 Mar
- 2008) | 16 lines Merged revisions 106437 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar
- 2008) | 8 lines Quell an annoying message that is likely to print
- every single time that ast_pbx_outgoing_app is called. The reason
- is that __ast_request_and_dial allocates the cdr for the channel,
- so it should be expected that the channel will have a cdr on it.
- Thanks to joetester on IRC for pointing this out ........
- ................
-
-2008-03-06 22:15 +0000 [r106440] Jason Parker <jparker@digium.com>
-
- * /, main/file.c: Merged revisions 106439 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106439 |
- qwell | 2008-03-06 16:11:30 -0600 (Thu, 06 Mar 2008) | 8 lines
- Fix file playback in many cases. (closes issue #12115) Reported
- by: pj Patches: v2-fileexists.patch uploaded by dimas (license
- 88) (with modifications by me) Tested by: dimas, qwell, russell
- ........
-
-2008-03-06 20:39 +0000 [r106433] Donny Kavanagh <donnyk@gmail.com>
-
- * /, res/res_agi.c: Merged revisions 106399 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r106399 |
- juggie | 2008-03-06 14:31:50 -0500 (Thu, 06 Mar 2008) | 9 lines
- trivial fix for an agi error when attempting to use EAGI on a
- dead/hungup channel, we now print an error that makes sense given
- our removal of deadagi as an actual application. (closes issue
- #12161) Reported by: explidous Patches: res_agi_12161.patch
- uploaded by juggie (license 24) Tested by: juggie ........
-
-2008-03-06 05:25 +0000 [r106330-106359] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_config_ldap.c: Merged revisions 106346 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r106346 | tilghman | 2008-03-05 23:21:39 -0600 (Wed, 05 Mar 2008)
- | 7 lines Missing braces, fix parsing (closes issue #12112)
- Reported by: cyrenity Patches: res_config_ldap.patch-03-03-2008
- uploaded by cyrenity (license 416) Tested by: cyrenity, Corydon76
- ........
-
- * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 106329
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r106329 | tilghman | 2008-03-05 22:45:16 -0600
- (Wed, 05 Mar 2008) | 10 lines Merged revisions 106328 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106328 | tilghman | 2008-03-05 22:40:06 -0600 (Wed, 05 Mar 2008)
- | 2 lines Upgrade to the next release of sounds ........
- ................
-
-2008-03-06 00:23 +0000 [r106299-106320] Russell Bryant <russell@digium.com>
-
- * channels/chan_oss.c, main/rtp.c, main/channel.c,
- channels/chan_phone.c, main/dial.c, channels/chan_skinny.c,
- main/file.c, channels/chan_h323.c, channels/chan_alsa.c,
- include/asterisk/frame.h, channels/chan_mgcp.c,
- channels/chan_unistim.c, apps/app_dial.c, channels/chan_zap.c, /,
- channels/chan_sip.c, channels/chan_console.c,
- apps/app_followme.c: Merged revisions 106239 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) |
- 12 lines Merged revisions 106235 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4
- lines Add a control frame to indicate the source of media has
- changed. Depending on the underlying technology it may need to
- change some things. (closes issue #12148) Reported by: jcomellas
- ........ ................
-
- * /, channels/chan_iax2.c: Merged revisions 106238 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r106238 | russell | 2008-03-05 16:40:58 -0600
- (Wed, 05 Mar 2008) | 11 lines Merged revisions 106237 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106237 | russell | 2008-03-05 16:37:09 -0600 (Wed, 05 Mar 2008)
- | 3 lines Fix a potential deadlock and a few different potential
- crashes. (closes issue #12145, reported by thiagarcia, patched by
- me) ........ ................
-
- * /, doc/tex/realtime.tex: Merged revisions 106186 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r106186 | mvanbaak | 2008-03-05 15:19:06 -0600 (Wed, 05 Mar 2008)
- | 7 lines document var_metric usage to prevent bugreports that
- are actually configuration issues (closes issue #12151) Reported
- by: caio1982 Patches: DB_metric3.diff uploaded by caio1982
- (license 22) ........
-
- * main/rtp.c, /, main/translate.c, include/asterisk/frame.h: Merged
- revisions 105933 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008)
- | 13 lines Merged revisions 105932 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008)
- | 5 lines Fix a bug that I just noticed in the RTP code. The
- calculation for setting the len field in an ast_frame of audio
- was wrong when G.722 is in use. The len field represents the
- number of ms of audio that the frame contains. It would have set
- the value to be twice what it should be. ........
- ................
-
- * funcs/func_global.c, /: Merged revisions 105899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105899 |
- russell | 2008-03-04 18:45:39 -0600 (Tue, 04 Mar 2008) | 3 lines
- Fix the SHARED() read callback to properly unlock the channel.
- This function could not have worked, as it left the channel
- locked in all cases. ........
-
- * main/manager.c, /: Merged revisions 105864 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105864 |
- mmichelson | 2008-03-04 17:24:56 -0600 (Tue, 04 Mar 2008) | 5
- lines There are several places in manager.c where BUFSIZ is used
- for a buffer which will contain nowhere near that amount of data.
- This makes these buffers more reasonably sized. ........
-
- * main/asterisk.c, channels/chan_zap.c, /, channels/console_gui.c,
- apps/app_queue.c: Merged revisions 105841 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105841 |
- tilghman | 2008-03-04 17:10:45 -0600 (Tue, 04 Mar 2008) | 2 lines
- Fix minor misuses of snprintf ........
-
- * main/rtp.c, main/netsock.c, main/cryptostub.c, main/file.c,
- main/callerid.c, main/alaw.c, main/dsp.c, main/dlfcn.c,
- main/frame.c, /, main/say.c, main/utils.c, main/enum.c,
- main/astobj2.c, main/config.c, main/fskmodem.c, main/poll.c,
- main/loader.c, main/term.c, main/cli.c, main/channel.c,
- main/dial.c, main/manager.c, main/tdd.c, main/strcompat.c,
- main/features.c, main/logger.c, main/app.c, main/image.c,
- main/dns.c, main/pbx.c, main/translate.c, main/jitterbuf.c:
- Merged revisions 105840 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105840 |
- tilghman | 2008-03-04 17:04:29 -0600 (Tue, 04 Mar 2008) | 2 lines
- Whitespace changes only ........
-
- * main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
- main/http.c, include/asterisk/tcptls.h: Merged revisions 105804
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r105804 | russell | 2008-03-04 16:28:03 -0600 (Tue, 04
- Mar 2008) | 2 lines add a destroy API call for a server instance
- ........
-
- * main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
- main/http.c, include/asterisk/tcptls.h: Merged revisions 105785
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r105785 | russell | 2008-03-04 16:23:21 -0600 (Tue, 04
- Mar 2008) | 2 lines More public API name changes to use an
- appropriate ast_ prefix ........
-
- * include/asterisk/http.h, main/tcptls.c, main/manager.c, /,
- channels/chan_sip.c, res/res_phoneprov.c, main/http.c,
- include/asterisk/tcptls.h: Merged revisions 105773 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r105773 | russell | 2008-03-04 16:15:18 -0600 (Tue, 04 Mar 2008)
- | 2 lines Rename public object server_instance to
- ast_tcptls_server_instance ........
-
- * /, channels/chan_sip.c: Merged revisions 105734 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105734 |
- russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines
- Fix some bugs in the SIP tcp helper thread. - fix a spot where a
- lock wouldn't get unlocked in an error condition - call
- ast_mutex_destroy() on the lock before freeing its memory
- (related to issue #11972) ........
-
- * /, res/res_phoneprov.c: Merged revisions 105733 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r105733 |
- twilson | 2008-03-04 14:32:55 -0600 (Tue, 04 Mar 2008) | 2 lines
- Set username to default to the category name if it isn't
- overridden by a usernmae= setting in users.conf ........
-
- * main/rtp.c, /: Merged revisions 105677 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r105677 | file | 2008-03-04 12:11:38 -0600 (Tue, 04 Mar 2008) |
- 10 lines Merged revisions 105676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2
- lines In addition to setting the marker bit let's change our ssrc
- so they know for sure it is a different source. ........
- ................
-
- * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
- Merged revisions 105675 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) |
- 16 lines Merged revisions 105674 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8
- lines When a new source of audio comes in (such as music on hold)
- make sure the marker bit gets set. (closes issue #10355) Reported
- by: wdecarne Patches: 10355.diff uploaded by file (license 11)
- (closes issue #11491) Reported by: kanderson ........
- ................
-
-2008-03-05 17:42 +0000 [r106140] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_talkdetect.c: Merged revisions 106139 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r106139 | tilghman | 2008-03-05 11:40:42 -0600 (Wed, 05 Mar 2008)
- | 3 lines Should check these values for non-NULL before scanning.
- (Closes issue #12147) ........
-
-2008-03-05 15:43 +0000 [r106041] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 106040 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106040 | kpfleming | 2008-03-05 09:40:40 -0600 (Wed, 05 Mar
- 2008) | 15 lines Merged revisions 106038 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar
- 2008) | 7 lines when a PRI call must be moved to a different B
- channel at the request of the other endpoint, ensure that any DSP
- active on the original channel is moved to the new one (closes
- issue #11917) Reported by: mavetju Tested by: mavetju ........
- ................
-
-2008-03-05 15:31 +0000 [r106037] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, include/asterisk/sched.h: Merged
- revisions 106036 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r106036 | tilghman | 2008-03-05 09:23:32 -0600 (Wed, 05 Mar 2008)
- | 15 lines Merged revisions 106015 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008)
- | 7 lines Correctly initialize retransid in SIP, and ensure that
- the warning when failing to delete a schedule entry can actually
- hit the log. (closes issue #12140) Reported by: slavon Patches:
- sch2.patch uploaded by slavon (license 288) (Patch slightly
- modified by me) ........ ................
-
-2008-03-04 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta5 released.
-
-2008-03-04 16:55 +0000 [r105574-105597] Russell Bryant <russell@digium.com>
-
- * CHANGES: Update CHANGES heading
-
- * funcs/func_version.c: Simplify a trivial snprintf() with
- ast_copy_string()
-
- * main/hashtab.c: Make it so you don't have to cast away const in a
- couple places
-
- * main/hashtab.c: remove unnecessary casts
-
- * main/pbx.c: - Add curly braces around the while loop - Properly
- break out of the loop on error when an included context is not
- found
-
- * main/pbx.c: Use ast_copy_string() instead of strncpy(), and use
- sizeof() instead of a magic number
-
- * channels/chan_zap.c: Fix some code that was improperly changed in
- revision 104866 from issue #12079. (closes issue #12129, reported
- by elguero, patched by me)
-
-2008-03-03 18:08 +0000 [r105573] Jason Parker <jparker@digium.com>
-
- * /, res/snmp/agent.c: Merged revisions 105572 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105572 | qwell | 2008-03-03 12:06:52 -0600 (Mon, 03 Mar 2008) |
- 7 lines Fix types for astNumChannels and astConfigCallsProcessed.
- (closes issue #12114) Reported by: jeffg Patches: 12114.patch
- uploaded by jeffg (license 192) ........
-
-2008-03-03 17:17 +0000 [r105564-105571] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 105570 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r105570 | russell | 2008-03-03 11:16:53 -0600 (Mon, 03
- Mar 2008) | 3 lines In the case of an ast_channel allocation
- failure, take the local_pvt out of the pvt list before destroying
- it. ........
-
- * channels/chan_local.c, /: Merged revisions 105568 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r105568 | russell | 2008-03-03 11:05:16 -0600 (Mon, 03
- Mar 2008) | 3 lines Fix a potential memory leak of the local_pvt
- struct when ast_channel allocation fails. Also, in passing,
- centralize the code necessary to destroy a local_pvt. ........
-
- * main/autoservice.c, /: Merged revisions 105565 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105565 | russell | 2008-03-03 10:01:50 -0600 (Mon, 03 Mar 2008)
- | 3 lines Update the copyright information for autoservice. Most
- of the code in this file now is stuff that I have written
- recently ... ........
-
- * main/channel.c, main/autoservice.c, /,
- include/asterisk/_private.h, main/asterisk.c: 3) In addition to
- merging the changes below, change trunk back to a regular LIST
- instead of an RWLIST. The way this list works makes it such that
- a RWLIST provides no additional benefit. Also, a mutex is needed
- for use with the thread condition. Merged revisions 105563 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008)
- | 24 lines Merge in some changes from
- team/russell/autoservice-nochans-1.4 These changes fix up some
- dubious code that I came across while auditing what happens in
- the autoservice thread when there are no channels currently in
- autoservice. 1) Change it so that autoservice thread doesn't keep
- looping around calling ast_waitfor_n() on 0 channels twice a
- second. Instead, use a thread condition so that the thread
- properly goes to sleep and does not wake up until a channel is
- put into autoservice. This actually fixes an interesting bug, as
- well. If the autoservice thread is already running (almost always
- is the case), then when the thread goes from having 0 channels to
- have 1 channel to autoservice, that channel would have to wait
- for up to 1/2 of a second to have the first frame read from it.
- 2) Fix up the code in ast_waitfor_nandfds() for when it gets
- called with no channels and no fds to poll() on, such as was the
- case with the previous code for the autoservice thread. In this
- case, the code would call alloca(0), and pass the result as the
- first argument to poll(). In this case, the 2nd argument to
- poll() specified that there were no fds, so this invalid pointer
- shouldn't actually get dereferenced, but, this code makes it
- explicit and ensures the pointers are NULL unless we have valid
- data to put there. (related to issue #12116) ........
-
-2008-03-03 15:30 +0000 [r105558-105561] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 105560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105560 | file | 2008-03-03 11:28:59 -0400 (Mon, 03 Mar 2008) | 7
- lines It is possible for no audio to pass between the current
- digit and next digit so expand logic that clears emulation to
- AST_FRAME_NULL. (closes issue #11911) Reported by: edgreenberg
- Patches: v1-11911.patch uploaded by dimas (license 88) Tested by:
- tbsky ........
-
- * /, channels/chan_sip.c: Merged revisions 105557 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105557 | file | 2008-03-03 11:15:39 -0400 (Mon, 03 Mar 2008) | 6
- lines Add a comment to describe some logic. (closes issue #12120)
- Reported by: flefoll Patches:
- chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
- 244) ........
-
-2008-03-01 03:59 +0000 [r105509] Joshua Colp <jcolp@digium.com>
-
- * main/slinfactory.c: Add support for 16KHz signed linear.
-
-2008-03-01 02:03 +0000 [r105479] Tilghman Lesher <tlesher@digium.com>
-
- * /: Drop bad property
-
-2008-03-01 01:30 +0000 [r105477] Terry Wilson <twilson@digium.com>
-
- * apps/app_dial.c, include/asterisk/app.h,
- main/global_datastores.c, /, main/features.c, main/app.c,
- include/asterisk/global_datastores.h: Asterisk, when parking can
- drop rights a caller when a parking timeout occurs. Also, when
- doing built-in attended transfers, sometimes incorrectly passes
- rights from the transferrer to the transferee. This patch tries
- to fixes the parking issue and lays some groundwork for later
- fixing the transfer issue. (closes issue #11520) Reported by:
- pliew Tested by: otherwiseguy
-
-2008-03-01 00:53 +0000 [r105461] Russell Bryant <russell@digium.com>
-
- * CHANGES, funcs/func_devstate.c: Add a "devstate change" CLI
- command to control custom device states. Also, do some additional
- code cleanup and improvement in passing. (closes issue #12106)
- Reported by: nizon Patches: devstate-patch.txt uploaded by nizon
- (license 415) -- Updated to trunk, and tab completion added by me
-
-2008-02-29 23:53 +0000 [r105411] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Convert to use ast_str
-
-2008-02-29 23:36 +0000 [r105410] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 105409 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105409 | russell | 2008-02-29 17:34:32 -0600 (Fri, 29 Feb 2008)
- | 23 lines Fix a major bug in autoservice. There was a race
- condition in the handling of the list of channels in autoservice.
- The problem was that it was possible for a channel to get removed
- from autoservice and destroyed, while the autoservice thread was
- still messing with the channel. This led to memory corruption,
- and caused crashes. This explains multiple backtraces I have seen
- that have references to autoservice, but do to the nature of the
- issue (memory corruption), could cause crashes in a number of
- areas. (fixes the crash in BE-386) (closes issue #11694) (closes
- issue #11940) The following issues could be related. If you are
- the reporter of one of these, please update to include this fix
- and try again. (potentially fixes issue #11189) (potentially
- fixes issue #12107) (potentially fixes issue #11573) (potentially
- fixes issue #12008) (potentially fixes issue #11189) (potentially
- fixes issue #11993) (potentially fixes issue #11791) ........
-
-2008-02-29 18:34 +0000 [r105378] Joshua Colp <jcolp@digium.com>
-
- * configs/sip.conf.sample: Add documentation for setting
- username/password in SIP dial string. (closes issue #11587)
- Reported by: sobomax Patches: dialstring_doc.diff uploaded by
- sobomax (license 359)
-
-2008-02-29 14:50 +0000 [r105263-105327] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, res/res_jabber.c: Merged revisions 105326 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105326 | phsultan | 2008-02-29 15:47:10 +0100 (Fri, 29 Feb 2008)
- | 1 line Fix a potential memory leak ........
-
- * channels/chan_jingle.c, channels/chan_gtalk.c, res/res_jabber.c:
- Remove unnecessary if statements before calling iks_delete
- (redundant check is done inside iks_delete), thus making the code
- conform with coding guidelines.
-
-2008-02-29 13:55 +0000 [r105262] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 105261 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r105261 | file | 2008-02-29 09:48:13 -0400 (Fri, 29 Feb
- 2008) | 4 lines Bump up the size of the uniqueid variable.
- (closes issue #12107) Reported by: asgaroth ........
-
-2008-02-29 13:12 +0000 [r105210] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Automatically create new buddy upon reception
- of a presence stanza of type subscribed. (closes issue #12066)
- Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
- phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
- (license 73) Tested by: ffadaie, phsultan
-
-2008-02-29 01:15 +0000 [r105176] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.debian.asterisk, /: Merged revisions 105113 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105113 | tilghman | 2008-02-28 15:56:54 -0600 (Thu, 28 Feb 2008)
- | 7 lines Update init script for LSB compat (closes issue #9843)
- Reported by: ibc Patches: rc.debian.asterisk.patch uploaded by
- ibc (license 211) Tested by: paravoid ........
-
-2008-02-28 22:39 +0000 [r105144] Russell Bryant <russell@digium.com>
-
- * /, main/utils.c, include/asterisk/lock.h, utils/check_expr.c:
- Merged revisions 105116 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008)
- | 8 lines Fix a bug in the lock tracking code that was discovered
- by mmichelson. The issue is that if the lock history array was
- full, then the functions to mark a lock as acquired or not would
- adjust the stats for whatever lock is at the end of the array,
- which may not be itself. So, do a sanity check to make sure that
- we're updating lock info for the proper lock. (This explains the
- bizarre stats on lock #63 in BE-396, thanks Mark!) ........
-
-2008-02-28 20:14 +0000 [r105060-105061] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 105059 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r105059 | mmichelson | 2008-02-28 14:11:57 -0600 (Thu, 28 Feb
- 2008) | 6 lines When using autofill, members who are in use
- should be counted towards the number of available members to call
- if ringinuse is set to yes. Thanks to jmls who brought this issue
- up on IRC ........
-
- * main/dial.c, /: Merged revisions 104841 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb
- 2008) | 17 lines Two fixes: 1. Make the list of ast_dial_channels
- a lockable list. This is because in some cases, the ast_dial may
- exist in multiple threads due to asynchronous execution of its
- application, and I found some cases where race conditions could
- exist. 2. Check in ast_dial_join to be sure that the channel
- still exists before attempting to lock it, since it could have
- gotten hung up but the is_running_app flag on the
- ast_dial_channel may not have been cleared yet. (closes issue
- #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
- putnopvut (license 60) Tested by: jvandal ........
-
-2008-02-28 19:21 +0000 [r105006] Jason Parker <jparker@digium.com>
-
- * main/cdr.c, main/pbx.c, /: Merged revisions 105005 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb
- 2008) | 9 lines Make pbx_exec pass an empty string into
- applications, if we get NULL. This protects against possible
- segfaults in applications that may try to use data before
- checking length (ast_strdupa'ing it, for example) (closes issue
- #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
- uploaded by qwell (license 4) ........
-
-2008-02-28 14:42 +0000 [r104974] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_vpb.cc: Fix crash when configuration does not match
- hardware detection. (closes issue #12096) Reported by: mmickan
- Patches: chan_vpb.cc.diff uploaded by mmickan (license 400)
-
-2008-02-28 04:37 +0000 [r104921] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 104920 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r104920 | qwell | 2008-02-27 22:31:21 -0600 (Wed, 27 Feb
- 2008) | 2 lines According to a video at www.cisco.com, the 7921G
- supports 6 line appearances. ........
-
-2008-02-28 00:11 +0000 [r104869] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/Makefile, build_tools/strip_nonapi: Merged revisions
- 104868 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008)
- | 7 lines Compatibility fix for PPC64 (closes issue #12081)
- Reported by: jcollie Patches: asterisk-1.4.18-funcdesc.patch
- uploaded by jcollie (license 412) Tested by: jcollie, Corydon76
- ........
-
-2008-02-27 23:58 +0000 [r104866] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: reduce indentation in alloc_sub (issue
- #12079) Reported by: tzafrir Patches: alloc_sub uploaded by
- tzafrir (license 46)
-
-2008-02-27 21:02 +0000 [r104788] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 104787 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104787 | file | 2008-02-27 16:56:23 -0400 (Wed, 27 Feb 2008) | 2
- lines Don't loop around infinitely trying to spy on our own
- channel, and don't forget to free/detach the datastore upon
- hangup of the spy. ........
-
-2008-02-27 20:37 +0000 [r104784] Mark Michelson <mmichelson@digium.com>
-
- * /, main/file.c: Merged revisions 104783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104783 | mmichelson | 2008-02-27 14:36:26 -0600 (Wed, 27 Feb
- 2008) | 4 lines Bump a couple of more buffers up by 2 so that
- annoying warnings aren't generated like crazy on every
- fileexists_core call. ........
-
-2008-02-27 19:36 +0000 [r104756] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Remove useless 's' and 'key' variables, in
- favor of 'val', which serves the exact same purpose.
-
-2008-02-27 18:20 +0000 [r104705] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, /: Merged revisions 104704 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104704 | tilghman | 2008-02-27 12:15:10 -0600 (Wed, 27 Feb 2008)
- | 2 lines Ensure the session ID can't be 0. ........
-
-2008-02-27 17:45 +0000 [r104687] Joshua Colp <jcolp@digium.com>
-
- * /, main/file.c: Merged revisions 104665 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104665 | file | 2008-02-27 13:41:40 -0400 (Wed, 27 Feb 2008) | 2
- lines Bump up the buffer by 2. ........
-
-2008-02-27 17:36 +0000 [r104643] Russell Bryant <russell@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 104625 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104625 | russell | 2008-02-27 11:33:04 -0600 (Wed, 27 Feb 2008)
- | 4 lines Fix a problem in ChanSpy where it could get stuck in an
- infinite loop without being able to detect that the calling
- channel hung up. (closes issue #12076, reported by junky, patched
- by me) ........
-
-2008-02-27 17:31 +0000 [r104617] Jason Parker <jparker@digium.com>
-
- * /, main/features.c: Merged revisions 104598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104598 | qwell | 2008-02-27 11:26:55 -0600 (Wed, 27 Feb 2008) |
- 8 lines Inherit language from the transfering channel on a blind
- transfer. (closes issue #11682) Reported by: caio1982 Patches:
- local_atxfer_lang3-1.4.diff uploaded by caio1982 (license 22)
- Tested by: caio1982, victoryure ........
-
-2008-02-27 17:12 +0000 [r104595-104597] Joshua Colp <jcolp@digium.com>
-
- * /, main/loader.c: Merged revisions 104596 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104596 | file | 2008-02-27 13:07:33 -0400 (Wed, 27 Feb 2008) | 4
- lines Use the lock (which already existed, it just wasn't used)
- on the updaters list to protect the contents instead of the
- overall module list lock. (closes issue #12080) Reported by:
- ChaseVenters ........
-
- * channels/chan_sip.c: After further discussion revert my previous
- commit for this. Currently in order to ensure devicestate is the
- expected value in another module (such as app_queue) then
- chan_sip must be loaded before hand.
-
-2008-02-27 16:54 +0000 [r104594] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/file.c: Merged revisions 104593 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104593 | kpfleming | 2008-02-27 10:53:06 -0600 (Wed, 27 Feb
- 2008) | 8 lines fallback to standard English prompts properly
- when using new prompt directory layout (closes issue #11831)
- Reported by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG
- (license 20) (modified by me to improve code and conform rest of
- function to coding guidelines) ........
-
-2008-02-27 16:26 +0000 [r104537-104539] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: When queueing up a device state change when
- the peer is loaded from the configuration give it a state of not
- in use. We have to do this because the channel technology may not
- yet be registered so the state could not be queried and would be
- considered invalid. (closes issue #12087) Reported by: liorm
-
- * res/res_smdi.c, /: Merged revisions 104536 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104536 | file | 2008-02-27 11:52:02 -0400 (Wed, 27 Feb 2008) | 4
- lines Only stop the MWI monitor thread if it was actually
- started. (closes issue #12086) Reported by: francesco_r ........
-
-2008-02-27 15:34 +0000 [r104534] Tilghman Lesher <tlesher@digium.com>
-
- * utils/astcanary.c: open(2) needs a mode argument when O_CREAT is
- specified. (Closes issue #12083)
-
-2008-02-27 15:31 +0000 [r104533] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c, main/rtp.c: Fix T38 passthrough regression
- introduced by state changes. (closes issue #12078) Reported by:
- dimas Patches: v1-12078.patch uploaded by dimas (license 88)
- (closes issue #12074) Reported by: Ivan
-
-2008-02-27 08:20 +0000 [r104502] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_vpb.cc, configs/vpb.conf.sample,
- include/asterisk/module.h: Bring Voicetronix driver up to date
- with current drivers (closes issue #12084) Reported by: mmickan
- Patches: chan_vpb.cc.diff uploaded by mmickan (license 400)
- module.h.diff uploaded by mmickan (license 400) vpb.conf.sample
- uploaded by mmickan (license 400)
-
-2008-02-27 04:42 +0000 [r104419-104473] Russell Bryant <russell@digium.com>
-
- * doc/janitor-projects.txt: note that the chan_sip conversion is
- already in progress
-
- * doc/janitor-projects.txt: add another janitor project
-
- * doc/janitor-projects.txt: Add the stuff from the janitor projects
- page that is still relevant. I figure that if we keep this in the
- tree, it will be much easier to keep up to date. The page on
- asterisk.org just links to this on svn.digium.com/view
-
-2008-02-27 03:52 +0000 [r104418] Jason Parker <jparker@digium.com>
-
- * doc/janitor-projects.txt (added): Create placeholder file...for
- now.
-
-2008-02-27 02:05 +0000 [r104388] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Whitespace changes only
-
-2008-02-27 01:16 +0000 [r104333-104335] Russell Bryant <russell@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 104334 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104334 | russell | 2008-02-26 19:15:02 -0600 (Tue, 26 Feb 2008)
- | 3 lines Avoid some recursion in the cleanup code for the
- chanspy datastore (closes issue #12076, reported by junky,
- patched by me) ........
-
-2008-02-26 22:14 +0000 [r104301] Steve Murphy <murf@digium.com>
-
- * res/snmp/agent.c: small change to allow this file to compile. No
- problem if you don't install the libsnmp package.
-
-2008-02-26 20:33 +0000 [r104244-104270] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: I swear I compiled this ... *cough*
-
- * res/res_phoneprov.c: fix this module, too
-
- * funcs/func_version.c: fix this module
-
- * Makefile, include/asterisk, build_tools/make_version_h (added):
- Re-add the automatically generated version.h, so that modules can
- include for making build time decisions for cross asterisk
- version compatibility
-
- * main/manager.c, channels/chan_sip.c, include/asterisk/version.h
- (removed), build_tools/make_version_c, res/res_agi.c,
- main/http.c, include/asterisk/ast_version.h (added): Rename
- version.h to ast_version.h. Next, I will be re-adding version.h
- as an automatically generated file like it used to be. This still
- needs to be there for modules that have to check it to compile
- against multiple asterisk versions.
-
-2008-02-26 19:14 +0000 [r104215] Joshua Colp <jcolp@digium.com>
-
- * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Add an
- 'e' option to ResetCDR which re-enables a CDR that has been
- disabled. (closes issue #11170) Reported by: kratzers Patches:
- ResetCDR.1.diff uploaded by kratzers (license 307)
-
-2008-02-26 18:40 +0000 [r104176] Tilghman Lesher <tlesher@digium.com>
-
- * doc/CODING-GUIDELINES: 1) Make braces mandatory for if/for/while,
- even around single statements. 2) Revise the argument parsing
- section, showing use of the standard macros. 3) Fix a typo.
-
-2008-02-26 18:27 +0000 [r104140-104142] Jason Parker <jparker@digium.com>
-
- * Makefile, /: Merged revisions 104141 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104141 | qwell | 2008-02-26 12:26:12 -0600 (Tue, 26 Feb 2008) |
- 1 line Add badshell to .PHONY target (thanks Kevin) ........
-
- * Makefile, /: Merged revisions 104139 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104139 | qwell | 2008-02-26 12:09:13 -0600 (Tue, 26 Feb 2008) |
- 2 lines Since all shells aren't as awesome as bash, we have to
- fail if somebody tries to use a literal "~" in DESTDIR. ........
-
-2008-02-26 16:51 +0000 [r104137] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Formatting and doxygen while waiting on an
- airport...
-
-2008-02-26 16:36 +0000 [r104133-104136] Jason Parker <jparker@digium.com>
-
- * /, sounds/Makefile: Merged revisions 104135 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104135 | qwell | 2008-02-26 10:35:06 -0600 (Tue, 26 Feb 2008) |
- 5 lines Revert previous abspath change. ...abspath is new in GNU
- make 3.81. I feel so...defeated. Must find new fix! ........
-
- * /, sounds/Makefile: Merged revisions 104132 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104132 | qwell | 2008-02-26 10:08:44 -0600 (Tue, 26 Feb 2008) |
- 9 lines Fix a very bizarre issue we were seeing with our buildbot
- when using a DESTDIR that wasn't an absolute path (such as
- DESTDIR=~/asterisk-1.4). Apparently what was happening, was that
- some of the targets were being expanded to the full path, so $@
- ended up being /root/asterisk-1.4/[...]/ rather than
- ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
- in GNU make. (*cough*
- http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
- ........
-
-2008-02-26 14:51 +0000 [r104127] Mark Michelson <mmichelson@digium.com>
-
- * main/features.c: Remove more hardcoded pipe symbols and replace
- with commas. (closes issue #12072) Reported by: SimonSharman
- Patches: features.patch uploaded by SimonSharman (license 410)
- Tested by: SimonSharman
-
-2008-02-26 06:43 +0000 [r104125] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_odbc.c: Use the readhandle for reads (closes issue
- #12069)
-
-2008-02-26 00:38 +0000 [r104120-104124] Russell Bryant <russell@digium.com>
-
- * res/res_smdi.c: Add a \todo to convert this module to the event
- system
-
- * CHANGES: Update CHANGES for SMDI stuff
-
- * channels/chan_zap.c, res/res_smdi.c, /, configs/smdi.conf.sample,
- include/asterisk/smdi.h, apps/app_voicemail.c: Merged revisions
- 104119 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008)
- | 33 lines Merge changes from team/russell/smdi-1.4 This commit
- brings in a significant set of changes to the SMDI support in
- Asterisk. There were a number of bugs in the current
- implementation, most notably being that it was very likely on
- busy systems to pop off the wrong message from the SMDI message
- queue. So, this set of changes fixes the issues discovered as
- well as introducing some new ways to use the SMDI support which
- are required to avoid the bugs with grabbing the wrong message
- off of the queue. This code introduces a new interface to SMDI,
- with two dialplan functions. First, you get an SMDI message in
- the dialplan using SMDI_MSG_RETRIEVE() and then you access
- details in the message using the SMDI_MSG() function. A side
- benefit of this is that it now supports more than just chan_zap.
- For example, with this implementation, you can have some FXO
- lines being terminated on a SIP gateway, but the SMDI link in
- Asterisk. Another issue with the current implementation is that
- it is quite common that the station ID that comes in on the SMDI
- link is not necessarily the same as the Asterisk voicemail box.
- There are now additional directives in the smdi.conf
- configuration file which let you map SMDI station IDs to Asterisk
- voicemail boxes. Yet another issue with the current SMDI support
- was related to MWI reporting over the SMDI link. The current code
- could only report a MWI change when the change was made by
- someone calling into voicemail. If the change was made by some
- other entity (such as with IMAP storage, or with a web interface
- of some kind), then the MWI change would never be sent. The SMDI
- module can now poll for MWI changes if configured to do so. This
- work was inspired by and primarily done for the University of
- Pennsylvania. (also related to issue #9260) ........
-
-2008-02-25 23:56 +0000 [r104103-104110] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, UPGRADE.txt: Deprecate the "stripmsd" option
- in favor of dialplan substring variable syntax. (closes issue
- #12060)
-
- * /, apps/app_chanspy.c: Merged revisions 104106 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104106 | russell | 2008-02-25 17:42:42 -0600 (Mon, 25 Feb 2008)
- | 10 lines This patch fixes some pretty significant problems with
- how app_chanspy handles pointers to channels that are being spied
- upon. It was very likely that a crash would occur if the channel
- being spied upon hung up. This was because the current
- ast_channel handling _requires_ that the object is locked or else
- it could disappear at any time (except in the owning channel
- thread). So, this patch uses some channel datastore magic on the
- spied upon channel to be able to detect if and when the channel
- goes away. (closes issue #11877) (patch written by me, but thanks
- to kpfleming for the idea, and to file for review) ........
-
- * /, main/utils.c: Merged revisions 104102 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104102 | russell | 2008-02-25 17:19:05 -0600 (Mon, 25 Feb 2008)
- | 7 lines Improve the lock tracking code a bit so that a bunch of
- old locks that threads failed to lock don't sit around in the
- history. When a lock is first locked, this checks to see if the
- last lock in the list was one that was failed to be locked. If it
- is, then that was a lock that we're no longer sitting in a
- trylock loop trying to lock, so just remove it. (inspired by
- issue #11712) ........
-
-2008-02-25 23:04 +0000 [r104097-104101] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_pgsql.c, CHANGES: Permit additional CDR columns to be
- saved in Postgres. Note that these changes are
- backward-compatible, so no changes to UPGRADE.txt are necessary.
- (closes issue #9279) Reported by: rottenroddy Patches:
- 20080125__bug9279.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76
-
- * funcs/func_global.c: Shared space for variables (instead of
- letting other channels muck with your own) (closes issue #11943)
- Reported by: ramonpeek Patches: 20080208__bug11943__2.diff.txt
- uploaded by Corydon76 (license 14) Tested by: jmls
-
- * /, apps/app_voicemail.c: Merged revisions 104094 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r104094 | tilghman | 2008-02-25 15:31:47 -0600 (Mon, 25
- Feb 2008) | 5 lines If the destination folder is full, don't
- delete a message when exiting. (closes issue #12065) Reported by:
- selsky Patch by: (myself) ........
-
-2008-02-25 21:40 +0000 [r104096] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 104095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6
- lines Make it so a users.conf user creates both a SIP peer and a
- SIP user. The user will be used for inbound authentication for
- the device, and peer will be used for placing calls to the
- device. (closes issue #9044) Reported by: queuetue Patches:
- sip-gui-friend.diff uploaded by qwell (license 4) ........
-
-2008-02-25 20:50 +0000 [r104093] Jason Parker <jparker@digium.com>
-
- * /, main/config.c: Merged revisions 104092 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104092 | qwell | 2008-02-25 14:49:42 -0600 (Mon, 25 Feb 2008) |
- 11 lines Allow the use of #include and #exec in situations where
- the max include depth was only 1. Specifically, this fixes using
- #include and #exec in extconfig.conf. This was basically caused
- because the config file itself raises the include level to 1. I
- opted not to raise the include limit, because recursion here
- could cause very bizarre behavior. Pointed out, and tested by
- jmls (closes issue #12064) ........
-
-2008-02-25 19:02 +0000 [r104089] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Instead of outputting a verbose message
- every so often let's make it a debug message.
-
-2008-02-25 19:00 +0000 [r104088] Brett Bryant <bbryant@digium.com>
-
- * doc/siptls.txt, configs/sip.conf.sample: Adding more tls
- configuration details to sip.conf sample, with a list of valid
- ciphers provided in both files. .. First commit since July, woot
-
-2008-02-25 18:38 +0000 [r104087] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 104086 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r104086 | russell | 2008-02-25 12:38:10 -0600 (Mon, 25
- Feb 2008) | 4 lines Ensure that the channel doesn't disappear in
- agent_logoff(). If it does, it could cause a crash. (fixes the
- crash reported in BE-396) ........
-
-2008-02-25 16:18 +0000 [r104081-104085] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 104084 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6
- lines If a resubscription comes in for a dialog we no longer know
- about tell the remote side that the dialog does not exist so they
- subscribe again using a new dialog. (closes issue #10727)
- Reported by: s0l4rb03 Patches: 10727-2.diff uploaded by file
- (license 11) ........
-
- * /, channels/chan_sip.c: Merged revisions 104082 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6
- lines Due to recent changes tag will no longer be NULL if not
- present so we have to use ast_strlen_zero to see if it's actually
- blank. (closes issue #12061) Reported by: flefoll Patches:
- chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll
- (license 244) ........
-
- * res/res_config_pgsql.c: Fix building of trunk. dbpass is always
- going to exist.
-
-2008-02-24 02:37 +0000 [r104073-104074] Steve Murphy <murf@digium.com>
-
- * channels/chan_sip.c: Enforce a space between function args as per
- code review.
-
- * res/res_config_pgsql.c: On a 64-bit machine, with dev-mode turned
- on, and pgsql installed, I get warnings that stops the compile.
- They are fixed now.
-
-2008-02-22 23:56 +0000 [r104045] Doug Bailey <dbailey@digium.com>
-
- * channels/chan_zap.c, configure, configure.ac: Add protection to
- chan_zap build when NEONMWI events are not defined
-
-2008-02-22 22:55 +0000 [r104036-104039] Tilghman Lesher <tlesher@digium.com>
-
- * doc/manager_1_1.txt, main/manager.c, UPGRADE.txt, CHANGES,
- include/asterisk/manager.h: Move Originate to a separate
- privilege and require the additional System privilege to call out
- to a subshell.
-
- * /, channels/chan_sip.c: Merged revisions 104037 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104037 | tilghman | 2008-02-22 16:45:14 -0600 (Fri, 22 Feb 2008)
- | 6 lines Backwards debug message. (closes issue #12052) Reported
- by: flefoll Patches: chan_sip.c.br14.patch_found-notfound
- uploaded by flefoll (license 244) ........
-
- * res/res_config_pgsql.c: Allow database password to be NULL and
- several other cleanups. (closes issue #12048) Reported by: bukaj
- Patches: 20080222__bug12048.diff.txt uploaded by Corydon76
- (license 14) Tested by: bukaj
-
-2008-02-21 21:27 +0000 [r104031] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: fix a typo
-
-2008-02-21 21:09 +0000 [r104025-104029] Mark Michelson <mmichelson@digium.com>
-
- * res/res_agi.c: Instead of a notice, make the message about a
- hung-up channel a debug message, and revert the original logic on
- the if statement. Thanks to Juggie for bringing this to my
- attention.
-
-2008-02-21 17:38 +0000 [r104024] Doug Bailey <dbailey@digium.com>
-
- * channels/chan_zap.c: Added configuration distinction between neon
- and fsk mwi detection Add the detection for neon MWI events got
- rid of extraneous handle_init_event call in monitor loop
-
-2008-02-21 16:46 +0000 [r104020] Mark Michelson <mmichelson@digium.com>
-
- * res/res_agi.c: Don't print the fact that we are using dead mode
- in AGI if called from the 'h' extension since it is well-known
- that it will be running in dead mode. (closes issue #12046)
- Reported by: explidous
-
-2008-02-21 16:44 +0000 [r104019] Joshua Colp <jcolp@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- Disable epoll as it has caused more obscure issues then any of my
- previous code. I will continue to work on it in a separate branch
- to make it stable for a release and test it against the following
- issues. (closes issue #11253) Reported by: falves11 (closes issue
- #11657) Reported by: davevg (closes issue #11033) Reported by:
- falves11
-
-2008-02-21 14:44 +0000 [r104016] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c, /: Merged revisions 104015 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r104015 | kpfleming | 2008-02-21 08:33:51 -0600 (Thu, 21 Feb
- 2008) | 2 lines reduce the likelihood that HTTP Manager session
- ids will consist of primarily '1' bits ........
-
-2008-02-21 05:21 +0000 [r104014] Tilghman Lesher <tlesher@digium.com>
-
- * utils/astman.c: Ignore some more unused generated events. (closes
- issue #12042) Reported by: junky Patches: astman_events.diff
- uploaded by junky (license 177)
-
-2008-02-20 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta4 released.
-
-2008-02-20 22:34 +0000 [r103957] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 103956 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103956 | mmichelson | 2008-02-20 16:32:22 -0600 (Wed, 20 Feb
- 2008) | 8 lines Clear up confusion when viewing the
- QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
- user's perspective, the queue does exist, we shouldn't tell them
- we couldn't find the queue. Instead since it is a dead queue,
- report a 0 waiting count This issue was brought up on IRC by jmls
- ........
-
-2008-02-20 22:29 +0000 [r103954-103955] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_h323.c: Try to do Packet2Packet bridging with
- chan_h323 if reinviting isn't enabled. (closes issue #11901)
- Reported by: pj
-
- * channels/chan_zap.c, /: Merged revisions 103953 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103953 | file | 2008-02-20 18:06:59 -0400 (Wed, 20 Feb 2008) | 6
- lines Don't wait for additional digits when overlap dialing is
- enabled if the setup message contains the sending_complete
- information element. (closes issue #11785) Reported by: klaus3000
- Patches: sending_complete_overlap_asterisk-1.4.17.patch.txt
- uploaded by klaus3000 (license 65) ........
-
-2008-02-20 21:41 +0000 [r103908] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 103904 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103904 | mmichelson | 2008-02-20 15:40:08 -0600 (Wed,
- 20 Feb 2008) | 6 lines Fix a crash if the channel becomes NULL
- while attempting to lock it. (closes issue #12039) Reported by:
- danpwi ........
-
-2008-02-20 21:36 +0000 [r103903] Jason Parker <jparker@digium.com>
-
- * include/asterisk/dsp.h, main/dsp.c: Largely refactor DSP tone
- detection routines. Separate fax detection from digit detected.
- Added CED (called) tone detection for fax (previously, only CNG
- (calling) was supported). Separate DTMF/MF code paths where
- appropriate. Allow detection of arbitary tones. (closes issue
- #11796) Reported by: dimas Patches: v6-dsp-faxtones.patch
- uploaded by dimas (license 88) Tested by: dimas, IgorG, Cache
-
-2008-02-20 21:08 +0000 [r103902] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix a crash due to the wrong variable being
- used when building a directory string. (closes issue #12027)
- Reported by: jaroth Patches: forward.patch uploaded by jaroth
- (license 50) Tested by: jaroth
-
-2008-02-20 18:29 +0000 [r103846-103847] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/sched.h: Add some documentation fixups
-
- * /, main/stdtime/localtime.c: Merged revisions 103845 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103845 | tilghman | 2008-02-20 11:53:00 -0600 (Wed, 20
- Feb 2008) | 7 lines Compat fix for Solaris (closes issue #12022)
- Reported by: asgaroth Patches: 20080219__bug12022.diff.txt
- uploaded by Corydon76 (license 14) Tested by: asgaroth ........
-
-2008-02-20 15:21 +0000 [r103844] Mark Michelson <mmichelson@digium.com>
-
- * res/res_monitor.c: Fix another spot where a hard-coded '|' hadn't
- been converted to ',' (closes issue #12034) Reported by: kowalma
-
-2008-02-20 03:52 +0000 [r103838-103842] Joshua Colp <jcolp@digium.com>
-
- * main/audiohook.c: *mumble*
-
- * main/audiohook.c: file not found.
-
- * main/audiohook.c: Minor test...
-
-2008-02-20 00:49 +0000 [r103833] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: When using IMAP storage, if the folder you
- attempt to save to does not exist, create it first. (closes issue
- #12032) Reported by: jaroth Patches: createfolder.patch uploaded
- by jaroth (license 50) Tested by: jaroth
-
-2008-02-19 22:35 +0000 [r103831-103832] Jason Parker <jparker@digium.com>
-
- * main/channel.c: Make sure to mask out non-audio first as well
-
- * main/channel.c: Maybe we should set the value before we test it?
- Fixes an issue people have been seeing (unreported?) with file
- playback not working.
-
-2008-02-19 21:54 +0000 [r103824-103828] Joshua Colp <jcolp@digium.com>
-
- * main/loader.c: Add a log message that appears when you try to
- unload a module that isn't loaded. (closes issue #12033) Reported
- by: jamesgolovich Patches: asterisk-loader.diff.txt uploaded by
- jamesgolovich (license 176)
-
- * main/file.c: Only output a log message saying the format does not
- exist if it actually does not exist, not if the file itself could
- not be opened. (closes issue #11828) Reported by: IgorG Patches:
- readfile.v1.diff uploaded by IgorG (license 20)
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 103823 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103823 | file | 2008-02-19 16:28:08 -0400 (Tue, 19 Feb 2008) | 6
- lines Send CallerID Name in setup message. (closes issue #11241)
- Reported by: tusar Patches: h323id_as_callerid_name.patch
- uploaded by tusar (license 344) ........
-
-2008-02-19 20:06 +0000 [r103822] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 103821 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19
- Feb 2008) | 8 lines Account for the fact that the "other" channel
- can disappear while the local pvt is not locked. (fixes a problem
- introduced in rev 100581) (closes issue #12012) Reported by:
- stevedavies Patch by me ........
-
-2008-02-19 19:27 +0000 [r103819-103820] Joshua Colp <jcolp@digium.com>
-
- * apps/app_authenticate.c: len already contains the position we
- want to examine, if we move one left again we'll actually
- probably be looking at a digit. (issue #12030) Reported by:
- alligosh
-
- * apps/app_channelredirect.c, UPGRADE.txt, CHANGES: Add
- CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan
- application. This will either be set to NOCHANNEL if the given
- channel was not found or SUCCESS if it worked. (closes issue
- #11553) Reported by: johan Patches:
- UPGRADE.txt.channelredirect.patch uploaded by johan (license 334)
- CHANGES.channelredirect.patch uploaded by johan (license 334)
- app_channelredirect-20080219.patch uploaded by johan (license
- 334)
-
-2008-02-19 18:14 +0000 [r103818] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: (closes issue #11864) Reported by: julianjm
- Patches: chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm
- (license 99) Patch fixes problem of device state incorrectly
- reporting idle before PBX answers incoming call on FXO channel.
- Device status is updated now during new channel creation.
-
-2008-02-19 17:33 +0000 [r103808-103813] Joshua Colp <jcolp@digium.com>
-
- * /, configure, configure.ac: Merged revisions 103812 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103812 | file | 2008-02-19 13:31:32 -0400 (Tue, 19 Feb
- 2008) | 4 lines Don't look for launchd when cross compiling.
- (closes issue #12029) Reported by: ovi ........
-
-2008-02-19 00:59 +0000 [r103805] Tilghman Lesher <tlesher@digium.com>
-
- * main/say.c: Change verbosity into debug for Hebrew (and various
- whitespace fixes) (Closes issue #12011)
-
-2008-02-18 23:58 +0000 [r103798-103802] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 103801 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103801 | file | 2008-02-18 19:56:48 -0400 (Mon, 18 Feb 2008) |
- 10 lines Ensure that emulated DTMFs do not get interrupted by
- another begin frame. (closes issue #11740) Reported by: gserra
- Patches: v1-11740.patch uploaded by dimas (license 88) (closes
- issue #11955) Reported by: tsearle (closes issue #10530) Reported
- by: xmarksthespot ........
-
- * main/channel.c, main/frame.c, channels/chan_sip.c,
- include/asterisk/channel.h, include/asterisk/frame.h: Add a
- non-invasive API for application level manipulation of T38 on a
- channel. This uses control frames (so they can even pass across
- IAX2) to negotiate T38 and provided a way of getting the current
- status of T38 using queryoption. This should by no means cause
- any issues and if it does I will take responsibility for it.
- (closes issue #11873) Reported by: dimas Patches:
- v4-t38-api.patch uploaded by dimas (license 88)
-
- * main/frame.c: Add some missing control frames.
-
-2008-02-18 22:33 +0000 [r103796] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 103795 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103795 | qwell | 2008-02-18 16:28:56 -0600 (Mon, 18 Feb 2008) |
- 1 line Fix previous commit so that we actually disable
- echocanbridged if echocancel is off. ........
-
-2008-02-18 21:57 +0000 [r103794] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Commit chan_zap portion of #11964: add the
- ability to get ORIG_CALLED_NUM
-
-2008-02-18 21:30 +0000 [r103791] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 103790 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103790 | qwell | 2008-02-18 15:23:32 -0600 (Mon, 18 Feb 2008) |
- 4 lines Correct a message when echocancelwhenbridged is on, but
- echocancel is not. Closes issue #12019 ........
-
-2008-02-18 20:58 +0000 [r103788] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure EC is enabled when SS7 call comes
- in. Also add support for multiple DPCs per linkset. #11779
-
-2008-02-18 20:53 +0000 [r103787] Mark Michelson <mmichelson@digium.com>
-
- * /, main/app.c: Merged revisions 103786 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb
- 2008) | 10 lines There was an invalid assumption when calculating
- the duration of a file that the filestream in question was
- created properly. Unfortunately this led to a segfault in the
- situation where an unknown format was specified in voicemail.conf
- and a voicemail was recorded. Now, we first check to be sure that
- the stream was written correctly or else assume a zero duration.
- (closes issue #12021) Reported by: jakep Tested by: putnopvut
- ........
-
-2008-02-18 19:47 +0000 [r103783] Michiel van Baak <michiel@vanbaak.info>
-
- * main/asterisk.c: make the output of 'core show settings' a bit
- nicer. (closes issue #12020) Reported by: seanbright Patches:
- asterisk.c.patch uploaded by seanbright (license 71)
-
-2008-02-18 17:45 +0000 [r103781] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, main/rtp.c: Merged revisions 103780 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008)
- | 9 lines When a SIP channel is being auto-destroyed, it's
- possible for it to still be in bridge code. When that happens, we
- crash. Delay the RTP destruction until the bridge is ended.
- (closes issue #11960) Reported by: norman Patches:
- 20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: norman ........
-
-2008-02-18 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta3 released.
-
-2008-02-18 17:12 +0000 [r103772] Olle Johansson <oej@edvina.net>
-
- * main/channel.c, channels/chan_sip.c: Make sure we can set up
- calls without audio (text+video). And ... it works!
-
-2008-02-18 16:40 +0000 [r103771] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 103770 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103770 | mmichelson | 2008-02-18 10:37:31 -0600 (Mon, 18 Feb
- 2008) | 10 lines Fix a linked list corruption that under the
- right circumstances could lead to a looped list, meaning it will
- traverse forever. (closes issue #11818) Reported by: michael-fig
- Patches: 11818.patch uploaded by putnopvut (license 60) Tested
- by: michael-fig ........
-
-2008-02-18 16:13 +0000 [r103764-103769] Joshua Colp <jcolp@digium.com>
-
- * apps/app_channelredirect.c, main/pbx.c, include/asterisk/pbx.h:
- Add an API call (ast_async_parseable_goto) which parses a goto
- string and does an async goto instead of an explicit goto.
- (closes issue #11753) Reported by: johan
-
- * /, channels/chan_sip.c: Merged revisions 103763 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103763 | file | 2008-02-18 11:33:14 -0400 (Mon, 18 Feb 2008) | 2
- lines Don't care if the extension given doesn't exist for
- subscription based MWI. ........
-
-2008-02-18 10:10 +0000 [r103755] Olle Johansson <oej@edvina.net>
-
- * CHANGES, channels/chan_iax2.c: - No space in manager event names,
- please - Add new event to CHANGES
-
-2008-02-18 04:43 +0000 [r103754] Tilghman Lesher <tlesher@digium.com>
-
- * build_tools/cflags.xml, main/channel.c, main/pbx.c,
- funcs/func_channel.c, include/asterisk/channel.h, CHANGES,
- main/cli.c: Context tracing for channels (closes issue #11268)
- Reported by: moy Patches:
- chantrace-datastored-encapsulated-rev94934.patch uploaded by moy
- (license 222)
-
-2008-02-16 21:22 +0000 [r103750] Michiel van Baak <michiel@vanbaak.info>
-
- * channels/chan_skinny.c: move two ast_log calls to ast_debug. Now
- monitoring chan_skinny port with nagios or zabbix wont generate
- noise on the console. @ok tilghman
-
-2008-02-15 23:32 +0000 [r103742] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 103741 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103741 | russell | 2008-02-15 17:31:39 -0600 (Fri, 15
- Feb 2008) | 8 lines Fix a crash in chan_iax2 due to a race
- condition (closes issue #11780) Reported by: guillecabeza
- Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
- 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
- 380) ........
-
-2008-02-15 23:20 +0000 [r103740] Mark Michelson <mmichelson@digium.com>
-
- * CHANGES: Document GotoIfTime change from svn revision 103738
-
-2008-02-15 23:14 +0000 [r103739] Russell Bryant <russell@digium.com>
-
- * include/asterisk/aes.h: Fix a regression in Asterisk 1.6 related
- to the use of AES encryption. 1024 was used instead of 128 when
- using AES from OpenSSL. Many thanks to d1mas for figuring this
- one out! (closes issue #11946) Reported by: bbhoss Patches:
- v1-11946.patch uploaded by dimas (license 88)
-
-2008-02-15 23:07 +0000 [r103737-103738] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Add proper "false" case behavior to GotoIfTime
- (closes issue #11719) Reported by: kshumard Patches:
- gotoiftime.twobranches.patch uploaded by kshumard (license 92)
- Tested by: kshumard
-
- * apps/app_voicemail.c: Fix redeclaration of variables when using
- IMAP storage (closes issue #11988) Reported by: jaroth Patches:
- variable_cleanup.patch uploaded by jaroth (license 50)
-
-2008-02-15 19:50 +0000 [r103727-103729] Russell Bryant <russell@digium.com>
-
- * /, main/loader.c: Merged revisions 103728 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103728 | russell | 2008-02-15 13:50:11 -0600 (Fri, 15 Feb 2008)
- | 4 lines In the case that you try to directly reload a module
- has returned AST_MODULE_LOAD_DECLINE, log a message indicating
- that the module is not fully initialized and must be initialized
- using "module load". ........
-
- * /, main/loader.c: Merged revisions 103726 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103726 | russell | 2008-02-15 12:33:29 -0600 (Fri, 15 Feb 2008)
- | 6 lines Don't attempt to execute the reload callback for a
- module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
- that was reported against chan_console in trunk. (closes issue
- #11953, reported by junky, fixed by me) ........
-
-2008-02-15 17:32 +0000 [r103725] Mark Michelson <mmichelson@digium.com>
-
- * doc/tex/imapstorage.tex, /, configure, configure.ac: Merged
- revisions 103722 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb
- 2008) | 8 lines Final round of changes for configure script logic
- for IMAP Now if a directory is specified, then we will search
- that directory for a source installation of the IMAP toolkit. If
- none is found, then we will use that directory as the basis for
- detecting a package installation of the IMAP c-client. If that
- check fails, then configure will fail. ........
-
-2008-02-15 17:29 +0000 [r103723] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, channels/chan_sip.c, res/res_phoneprov.c,
- include/asterisk/extconf.h, channels/misdn/isdn_msg_parser.c,
- apps/app_queue.c, channels/misdn/isdn_lib.c, main/config.c,
- main/channel.c, res/res_config_curl.c, channels/misdn/isdn_lib.h,
- main/ast_expr2f.c, channels/misdn/ie.c,
- channels/misdn/chan_misdn_config.h, channels/misdn/portinfo.c,
- include/asterisk/strings.h, res/res_config_ldap.c,
- include/asterisk/time.h: Fix up some doxygen issues. (closes
- issue #11996) Patches: bug_11996_doxygen.diff uploaded by snuffy
- (license 35)
-
-2008-02-15 15:45 +0000 [r103716] Tilghman Lesher <tlesher@digium.com>
-
- * utils/conf2ael.c: Remove extraneous copy (closes issue #12002)
- Reported by: junky Patches: conf2ael.diff uploaded by junky
- (license 177)
-
-2008-02-15 15:11 +0000 [r103699-103715] Mark Michelson <mmichelson@digium.com>
-
- * configure, configure.ac: Merging of changes from 1.4 revision
- 103713.
-
- * doc/tex/imapstorage.tex, configure, configure.ac: Same changes as
- made to 1.4 in revision 103710
-
- * doc/tex/imapstorage.tex: Trunk version of 1.4's imap
- documentation updates
-
- * configure, configure.ac: See commit message for svn revision
- 103698. This behavior is the same as what is described there. The
- difference is that trunk already had the --with-imap=system
- option, but it only checked the include path for headers in the
- imap directory and not also the c-client directory.
-
-2008-02-14 21:21 +0000 [r103694] Jason Parker <jparker@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: Modify
- ldap autoconf function, so that an incorrect ldap library is not
- found on Solaris (it is incompatible). Also removes second check
- for awk, which causes Solaris to find an incompatible version of
- awk. (closes issue #11829) Reported by: snuffy Patches:
- bug-11829.diff uploaded by snuffy (license 35)
-
-2008-02-14 21:04 +0000 [r103687-103691] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 103690 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103690 | mmichelson | 2008-02-14 15:03:02 -0600 (Thu,
- 14 Feb 2008) | 3 lines Fix build for non-IMAP builds ........
-
- * /, apps/app_voicemail.c: Merged revisions 103688 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103688 | mmichelson | 2008-02-14 14:55:48 -0600 (Thu,
- 14 Feb 2008) | 9 lines Fix the new message count if delete=yes
- when using IMAP storage. (closes issue #11406) Reported by:
- jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
- 50) Tested by: jaroth ........
-
- * configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Change
- the queue holdtime announcement to happen at any interval (not
- just greater than two minutes). Remove the saying of less-than
- for holdtime announcements since it can lead to awkward holdtime
- announcements. Using '1' as a queue-round-seconds value is no
- longer valid. (closes issue #9736) Reported by: caio1982 Patches:
- queue_announce5.diff uploaded by caio1982 (license 22) Tested by:
- caio1982, putnopvut
-
-2008-02-14 19:52 +0000 [r103685] Jason Parker <jparker@digium.com>
-
- * /, funcs/func_cdr.c: Merged revisions 103683 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103683 | qwell | 2008-02-14 13:51:10 -0600 (Thu, 14 Feb 2008) |
- 5 lines Document the 'l' option to the CDR() function. (Thanks
- voipgate for pointing out the option, and Leif for providing text
- for it.) Closes issue #11695. ........
-
-2008-02-14 19:47 +0000 [r103682] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_externalivr.c: a few syntax changes and safer code
-
-2008-02-14 18:39 +0000 [r103677] Jason Parker <jparker@digium.com>
-
- * channels/chan_iax2.c: Add periodic jitter stats to CLI and
- manager. (closes issue #8188) Reported by: stevedavies Patches:
- jblogging-trunk.patch uploaded by stevedavies
- jblogging-trunk_wmgrevent.patch uploaded by johann8384
- updated_jbloggin-trunk_mgrevent.patch uploaded by johann8384
- (license 190) (with additional changes by me) Tested by:
- stevedavies, johann8384
-
-2008-02-14 10:19 +0000 [r103668] Olle Johansson <oej@edvina.net>
-
- * res/res_agi.c, apps/app_externalivr.c: Formatting fixes
-
-2008-02-13 21:04 +0000 [r103662] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_externalivr.c: (closes issue #11825) Reported by:
- ctooley Patches: additional_eivr_commands.patch uploaded by
- ctooley (license 136) Tested by: ctooley
-
-2008-02-13 15:47 +0000 [r103658] Mark Michelson <mmichelson@digium.com>
-
- * UPGRADE.txt, res/res_musiconhold.c: 1. Deprecate SetMusicOnHold
- and WaitMusicOnHold. 2. Add a duration parameter to MusicOnHold
- (closes issue #11904) Reported by: dimas Patches: v2-moh.patch
- uploaded by dimas (license 88) Tested by: dimas
-
-2008-02-13 00:55 +0000 [r103559] Mark Michelson <mmichelson@digium.com>
-
- * main/event.c: Fix a small logic error in ast_event_iterator_next.
- The previous logic allowed for the iterator to indicate there was
- more data than there really was, causing the iterator read beyond
- the end of the event structure. This led to invalid memory reads
- and potential crashes.
-
-2008-02-12 22:26 +0000 [r103447-103506] Jason Parker <jparker@digium.com>
-
- * main/manager.c: Even more sane permissions. This should be
- handled via a umask, like in many other places.
-
- * main/manager.c: Use slight more sane permissions
-
-2008-02-12 15:39 +0000 [r103387-103388] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Remove development version notice.
-
- * main/manager.c: Fix build on *BSD. These permissions constants
- are not available there.
-
-2008-02-12 15:13 +0000 [r103386] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 103385 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4
- lines Even if no CallerID name or number has been provided by the
- remote party still use the configured sip.conf ones. (closes
- issue #11977) Reported by: pj ........
-
-2008-02-12 14:08 +0000 [r103341] Philippe Sultan <philippe.sultan@gmail.com>
-
- * include/asterisk/jabber.h, res/res_jabber.c: Use an ast_flags
- structure in aji_client and aji_buddy rather than an integer.
- Modify calls to various ast_*_flag macros accordingly.
-
-2008-02-12 00:24 +0000 [r103331] Jeff Peeler <jpeeler@digium.com>
-
- * main/manager.c, include/asterisk/config.h, CHANGES,
- main/config.c: Requested changes from Pari, reviewed by Russell.
- Added ability to retrieve list of categories in a config file.
- Added ability to retrieve the content of a particular category.
- Added ability to empty a context. Created new action to create a
- new file. Updated delete action to allow deletion by line number
- with respect to category. Added new action insert to add new
- variable to category at specified line. Updated action newcat to
- allow new category to be inserted in file above another existing
- category.
-
-2008-02-11 22:10 +0000 [r103317-103325] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 103324 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103324 | file | 2008-02-11 18:09:07 -0400 (Mon, 11 Feb 2008) | 4
- lines If entering a conference with the 'w' option ensure that we
- can't listen or speak until the marked user appears. (closes
- issue #11835) Reported by: alanmcmillan ........
-
- * res/res_agi.c: Remove ast_module_user usage from res_agi. This is
- taken care of in the core.
-
- * main/pbx.c, main/manager.c, main/translate.c, main/logger.c,
- main/app.c, main/utils.c, main/indications.c, main/asterisk.c,
- main/rtp.c: Just some minor coding style cleanup...
-
- * main/pbx.c: Fix Manager Redirect while in an AGI. (closes issue
- #10661) Reported by: junky
-
-2008-02-11 17:09 +0000 [r103316] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configs/zapata.conf.sample: Merged revisions 103315 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103315 | kpfleming | 2008-02-11 11:05:22 -0600 (Mon, 11 Feb
- 2008) | 2 lines improve 2BCT documentation a bit (thanks Jared)
- ........
-
-2008-02-11 16:17 +0000 [r103313-103314] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, channels/chan_iax2.c: Add support for allowing a
- native bridge to happen when the L option is enabled. The RTP
- bridging could already handle this, it just needed to be enabled
- in the main bridging code. (issue #10647) Reported by: samdell3
-
- * channels/chan_skinny.c: Change chan_skinny to use debug messages
- as appropriate. (closes issue #11967) Reported by: mvanbaak
- Patches: 2008021000-skinnydebug.diff.txt uploaded by mvanbaak
- (license 7)
-
-2008-02-11 06:05 +0000 [r103306] James Golovich <james@gnuinter.net>
-
- * channels/chan_sip.c: Don't wipe out transport and fd in chan_sip
- on reload (issue #11930)
-
-2008-02-11 03:03 +0000 [r103282-103284] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fix improper indentation. Thanks again to
- snuffy for pointing it out.
-
- * apps/app_queue.c: Add a couple of comments to clarify the
- unreffing of queues. Thanks to snuffy for the idea.
-
- * main/event.c: Fix a problem regarding network vs. host byte order
- in the event API. ast_event_iterator_get_ie_type should return
- the ie type in host byte order. Furthermore, ast_event_get_ie_raw
- should already have its ie type argument in host byte order since
- it could be called externally (and it in fact is called in this
- way by ast_event_get_cached).
-
-2008-02-09 11:27 +0000 [r103249] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_dial.c, apps/app_dictate.c, apps/app_echo.c,
- apps/app_authenticate.c, apps/app_disa.c, apps/app_chanisavail.c,
- apps/app_exec.c, apps/app_db.c, apps/app_controlplayback.c,
- apps/app_channelredirect.c, apps/app_directed_pickup.c,
- apps/app_dumpchan.c, apps/app_amd.c, apps/app_externalivr.c,
- apps/app_directory.c, apps/app_chanspy.c, apps/app_cdr.c:
- whitespace fixes only.
-
-2008-02-09 06:33 +0000 [r103198] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 103197 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103197 | tilghman | 2008-02-09 00:23:49 -0600 (Sat, 09
- Feb 2008) | 4 lines Commit fix for being unable to send voicemail
- from VoiceMailMain Reported by: William F Acker (via the -users
- mailing list) Patch by: Corydon76 (license 14) ........
-
-2008-02-08 21:26 +0000 [r103171] Russell Bryant <russell@digium.com>
-
- * main/udptl.c, main/pbx.c, channels/chan_sip.c,
- channels/chan_iax2.c, res/res_jabber.c, apps/app_playback.c,
- main/rtp.c, channels/chan_usbradio.c, main/cdr.c,
- channels/chan_skinny.c, apps/app_minivm.c, res/res_agi.c,
- pbx/pbx_ael.c, pbx/pbx_dundi.c, funcs/func_devstate.c,
- apps/app_rpt.c, main/asterisk.c, channels/chan_mgcp.c,
- apps/app_voicemail.c: Merge changes from
- team/mvanbaak/cli-command-audit (closes issue #8925) About a year
- ago, as Leif Madsen and Jim van Meggelen were going over the CLI
- commands in Asterisk 1.4 for the next version of their book, they
- documented a lot of inconsistencies. This set of changes
- addresses all of these issues and has been reviewed by Leif.
- While this does introduce even more changes to the CLI command
- structure, it makes everything consistent, which is the most
- important thing. Thanks to all that helped with this one!
-
-2008-02-08 18:58 +0000 [r103071-103122] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Forgot that AST_LIST_REMOVE_CURRENT takes
- different arguments in trunk than 1.4.
-
- * /, apps/app_queue.c: Merged revisions 103120 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r103120 | mmichelson | 2008-02-08 12:48:17 -0600 (Fri, 08 Feb
- 2008) | 10 lines Prevent a potential three-thread deadlock. Also
- added a comment block to explicitly state the locking order
- necessary inside app_queue. (closes issue #11862) Reported by:
- flujan Patches: 11862.patch uploaded by putnopvut (license 60)
- Tested by: flujan ........
-
- * /, channels/chan_iax2.c: Merged revisions 103070 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r103070 | mmichelson | 2008-02-08 12:00:38 -0600 (Fri,
- 08 Feb 2008) | 6 lines Yield the thread and return -1 if the
- ioctl fails for Zaptel timing device. (closes issue #11891)
- Reported by: tzafrir ........
-
-2008-02-08 16:49 +0000 [r103044] Russell Bryant <russell@digium.com>
-
- * UPGRADE-1.2.txt (added), UPGRADE-1.4.txt (added), UPGRADE.txt: At
- the request of ManxPower, include the UPGRADE.txt from 1.2 and
- 1.4, as well. This way, if people need to go back and review what
- was deprecated in previous major releases, it is readily
- available to them. Thanks for the suggestion!
-
-2008-02-08 15:31 +0000 [r102969-103018] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix a network byte order issue and ensure
- when creating an outgoing dialog that the socket always contains
- information such as type and port. (closes issue #11916) Reported
- by: mnnojd
-
- * /, channels/chan_iax2.c: Merged revisions 102968 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r102968 | file | 2008-02-08 11:08:20 -0400 (Fri, 08 Feb
- 2008) | 4 lines Make sure the presence of dbsecret is factored
- into user scoring. (closes issue #11952) Reported by: bbhoss
- ........
-
-2008-02-07 21:37 +0000 [r102933] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: This is a combination new feature/bug fix for
- app_chanspy. New feature: Add the 'e' option, which takes as an
- argument a list of interfaces separated by colons. This way, you
- will only be able to spy on this limited list of interfaces. Bug
- fix: change some pointer checks to ast_strlen_zero so that spying
- would work properly even if no channel was specified as the first
- argument to chanspy. (closes issue #10072) Reported by:
- xmarksthespot Patches:
- bugfix+newfeature10072patchtotrunkrev102726.diff uploaded by
- xmarksthespot (license 16) Tested by: xmarksthespot, mvanbaak
-
-2008-02-07 21:08 +0000 [r102906-102908] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_adsiprog.c: whitespace fixes only
-
- * apps/app_alarmreceiver.c: There she goes! First commit from me to
- trunk \o/ Make app_alarmreceiver honor code guidelines and fix
- whitespace errors. No functional changes.
-
-2008-02-07 20:02 +0000 [r102859] Jason Parker <jparker@digium.com>
-
- * /, main/features.c: Merged revisions 102858 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102858 | qwell | 2008-02-07 13:53:55 -0600 (Thu, 07 Feb 2008) |
- 7 lines Specify which digit string was matched in debug message.
- (closes issue #11949) Reported by: dimas Patches:
- v1-feature-debug.patch uploaded by dimas (license 88) ........
-
-2008-02-07 16:47 +0000 [r102808] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configs/zapata.conf.sample: Merged revisions 102807 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102807 | kpfleming | 2008-02-07 10:41:55 -0600 (Thu, 07 Feb
- 2008) | 2 lines document usage of 'transfer' configuration option
- for ISDN PRI switch-side transfers ........
-
-2008-02-06 20:12 +0000 [r102777] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Add the channel's unique id to the AgentCalled
- manager event to make it more consistent with other manager
- events.
-
-2008-02-06 18:01 +0000 [r102726] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 102725 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102725 | file | 2008-02-06 13:59:23 -0400 (Wed, 06 Feb 2008) | 2
- lines Only consider a T.38-only INVITE compatible if we have both
- a joint capability between us and them and if they provided T.38.
- ........
-
-2008-02-06 16:23 +0000 [r102700] Terry Wilson <twilson@digium.com>
-
- * funcs/func_realtime.c: Add REALTIME_STORE and REALTIME_DESTROY
- dialplan functions provided by sergee. I just added the ability
- to set multiple fields at once after discussions with Tilghman
- and Russell. Currently limited to 30 fields. (closes issue
- #11887) Reported by: sergee Patches:
- rt-func-store-destroy-multivalue.diff uploaded by otherwiseguy
- (license 396) Tested by: sergee, otherwiseguy
-
-2008-02-06 15:20 +0000 [r102652] Russell Bryant <russell@digium.com>
-
- * /, configs/features.conf.sample: Merged revisions 102651 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102651 | russell | 2008-02-06 09:19:41 -0600 (Wed, 06 Feb 2008)
- | 3 lines Clarify setting DYNAMIC_FEATURES so that it gets
- inherited by outbound channels. (due to a discussion between me
- and a user via email) ........
-
-2008-02-06 03:05 +0000 [r102602] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 102576 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r102576 | tilghman | 2008-02-05 18:26:02 -0600 (Tue, 05
- Feb 2008) | 4 lines Move around some defines to unbreak ODBC
- storage. (closes issue #11932) Reported by: snuffy ........
-
-2008-02-06 00:08 +0000 [r102501-102550] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Remove an extra debug message I left in
-
- * channels/chan_unistim.c, apps/app_dial.c, main/pbx.c,
- apps/app_privacy.c, apps/app_alarmreceiver.c, res/res_jabber.c,
- apps/app_followme.c, main/loader.c, channels/chan_usbradio.c,
- main/tcptls.c, res/res_agi.c, apps/app_minivm.c,
- apps/app_dumpchan.c, main/logger.c, apps/app_zapras.c,
- main/astmm.c: Get rid of any remaining ast_verbose calls in the
- code in favor of ast_verb (closes issue #11934) Reported by:
- mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by
- mvanbaak (license 7)
-
- * apps/app_voicemail.c: Change verbose messages to use the ast_verb
- macro. (closes issue #11931) Reported by: snuffy Patches:
- bug-11931.diff uploaded by snuffy (license 35)
-
-2008-02-05 20:51 +0000 [r102500] Jason Parker <jparker@digium.com>
-
- * main/pbx.c: Change where priority of a goto is adjusted.
- Partially reverts 102272. Closes issue #11929 (credit to file for
- fix suggestion - we still <3 you)
-
-2008-02-05 20:03 +0000 [r102454] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_mgcp.c: Merged revisions 102453 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r102453 | mmichelson | 2008-02-05 14:02:44 -0600 (Tue,
- 05 Feb 2008) | 8 lines Clear the DTMF buffer on hangup. (closes
- issue #11919) Reported by: eferro Patches:
- mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
- Tested by: eferro ........
-
-2008-02-05 19:58 +0000 [r102379-102452] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Yeah yeah, I broke building on trunk. Shoot
- me.
-
- * /, channels/chan_sip.c: Merged revisions 102450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102450 | file | 2008-02-05 15:52:30 -0400 (Tue, 05 Feb 2008) | 3
- lines If a REGISTER attempt comes in that is a retransmission of
- a previous REGISTER do not create a new nonce value. (issue
- #BE-381) ........
-
- * /, res/res_clioriginate.c: Merged revisions 102378 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r102378 | file | 2008-02-05 11:09:29 -0400 (Tue, 05 Feb
- 2008) | 4 lines Perform dialing asynchronously when using the
- originate CLI command so the CLI does not appear to block.
- (closes issue #11927) Reported by: bbhoss ........
-
-2008-02-04 21:15 +0000 [r102329] Tilghman Lesher <tlesher@digium.com>
-
- * utils/muted.c, /, configure, include/asterisk/autoconfig.h.in,
- configure.ac, main/asterisk.c: Merged revisions 102323 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102323 | tilghman | 2008-02-04 15:06:09 -0600 (Mon, 04 Feb 2008)
- | 7 lines Cross-platform fix: OS X now deprecates the use of the
- daemon(3) API. (closes issue #11908) Reported by: oej Patches:
- 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76 ........
-
-2008-02-04 18:39 +0000 [r102297] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Add line numbers to warning/error messages
- (and pretty up some existing ones). (closes issue #11894)
- Reported by: jmls Patches: chan_zap.patch uploaded by jmls
- (license 141)
-
-2008-02-04 15:16 +0000 [r102272] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: Update handling of asyncgoto so it properly works on
- channels that are currently executing a PBX. (closes issue
- #11914) Reported by: arnd (closes issue #11753) Reported by:
- johan
-
-2008-02-04 14:37 +0000 [r102262] Jason Parker <jparker@digium.com>
-
- * configs/extensions.ael.sample, configs/extensions.lua.sample:
- Change examples to use G here also. Closes issue #11875
-
-2008-02-04 05:32 +0000 [r102190-102238] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 102214 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r102214 | tilghman | 2008-02-03 23:10:02 -0600 (Sun, 03
- Feb 2008) | 6 lines Missing braces. (closes issue #11912)
- Reported by: dimas Patches: sprintf.patch uploaded by dimas
- (license 88) ........
-
- * main/manager.c: CoreSettings and CoreStatus are missing the
- terminating "\r\n". Also, some miscellaneous spacing and
- initialization issues. (closes issue #11909) Reported by: srt
- Patches: patch-11909-2.diff uploaded by srt (license 378) Tested
- by: srt
-
-2008-02-03 16:46 +0000 [r102091-102143] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 102142 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102142 | oej | 2008-02-03 17:38:12 +0100 (Sön, 03 Feb 2008) | 8
- lines Use the same CSEQ on CANCEL as on INVITE (according to RFC
- 3261) (closes issue #9492) Reported by: kryptolus Patches:
- bug9492.txt uploaded by oej (license 306) Tested by: oej ........
-
- * /, channels/chan_sip.c: Merged revisions 102090 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8
- lines Handle ACK and CANCEL in an invite transaction - even if we
- get INFO transactions during the actual call setup. (closes issue
- #10567) Reported by: jacksch Tested by: oej Patch by: oej
- inspired by suggestions from neutrino88 in the bug tracker
- ........
-
-2008-02-03 06:43 +0000 [r102064] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Change the version number in the
- configure script from 1.4 to 1.6
-
-2008-02-02 06:10 +0000 [r101990-102037] Russell Bryant <russell@digium.com>
-
- * include/asterisk/event.h: The documentation page has to be in its
- own comment block to work, apparently. Fix it up!
-
- * /, channels/chan_sip.c: Merged revisions 101989 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008)
- | 5 lines Change the SDP_SAMPLE_RATE macro. It turns out that
- even though G.722 is 16 kHz, it is supposed to specified as 8 kHz
- in the RTP, and RTP timestamps are supposed to be calculated
- based on 8 kHz. (Apparently this is due to a bug in a spec, but
- people follow it anyway, because it's the spec ...) ........
-
-2008-02-01 22:12 +0000 [r101873-101943] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 101942 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r101942 | tilghman | 2008-02-01 15:54:28 -0600 (Fri, 01
- Feb 2008) | 8 lines Fix the VM_DUR variable for forwarded
- voicemail, and fixed several other bugs while I'm in the area.
- (closes issue #11615) Reported by: jamessan Patches:
- 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76, jamessan ........
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 101894 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101894 | tilghman | 2008-02-01 13:36:12 -0600 (Fri, 01 Feb 2008)
- | 2 lines Change detection of getifaddrs to use
- AST_C_COMPILE_CHECK, backported from trunk (as suggested by
- kpfleming) ........
-
- * res/res_config_curl.c: Fix multi, when using the LIKE query.
- (closes issue #11889) Reported by: jmls Patches:
- res_config_curl.patch uploaded by jmls (license 141) Tested by:
- jmls
-
-2008-02-01 18:24 +0000 [r101869] Jason Parker <jparker@digium.com>
-
- * apps/app_authenticate.c: Comparison, not set :) Thanks mvanbaak.
-
-2008-02-01 18:08 +0000 [r101824] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c, configs/res_odbc.conf.sample: Clarify the pooling
- functionality by changing the config file keyword
-
-2008-02-01 17:44 +0000 [r101823] Jason Parker <jparker@digium.com>
-
- * /, apps/app_authenticate.c: Move an feof() call to before the
- fgets(). This would have exited the loop early if you had an
- authentication file with no newline at the end.
-
-2008-02-01 17:28 +0000 [r101819-101821] Russell Bryant <russell@digium.com>
-
- * /, apps/app_authenticate.c: Merged revisions 101818 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r101818 | russell | 2008-02-01 11:23:47 -0600 (Fri, 01
- Feb 2008) | 4 lines Don't overwrite the last character of a line
- if it's not a newline. This would happen if the last line in the
- file doesn't have a newline. (pointed out by Qwell) ........
-
-2008-02-01 16:01 +0000 [r101773] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/acl.c: Merged revisions 101772 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101772 | tilghman | 2008-02-01 09:55:58 -0600 (Fri, 01 Feb 2008)
- | 2 lines Compatibility fix for OpenWRT (reported by Brian
- Capouch via the mailing list) ........
-
-2008-02-01 06:27 +0000 [r101694-101746] Russell Bryant <russell@digium.com>
-
- * apps/app_authenticate.c: simplify some code, tweak formatting,
- and reduce indentation
-
- * apps/app_authenticate.c: reduce a level of indentation
-
- * apps/app_channelredirect.c: Get rid of a goto where there was no
- extra cleanup happening at the exit point
-
- * /, channels/chan_iax2.c: Merged revisions 101693 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r101693 | russell | 2008-01-31 18:32:49 -0600 (Thu, 31
- Jan 2008) | 8 lines Add some more sanity checking on IAX2 dial
- strings for the case that no peer or hostname was provided, which
- is the one part of the dial string that is absolutely required.
- If it's not there, bail out. (closes issue #11897) Reported by
- sokhapkin Patch by me ........
-
-2008-02-01 00:08 +0000 [r101650] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_amd.c: Merged revisions 101649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101649 | mmichelson | 2008-01-31 18:06:37 -0600 (Thu, 31 Jan
- 2008) | 9 lines From bugtracker: "fix totalAnalysisTime to handle
- periods of no channel activity" (closes issue #9256) Reported by:
- cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
- uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
- rjain ........
-
-2008-01-31 23:14 +0000 [r101611] Russell Bryant <russell@digium.com>
-
- * /, main/translate.c, main/file.c: Merged revisions 101601 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101601 | russell | 2008-01-31 17:10:06 -0600 (Thu, 31 Jan 2008)
- | 12 lines Fix a couple of places where ast_frfree() was not
- called on a frame that came from a translator. This showed itself
- by g729 decoders not getting released. Since the flag inside the
- translator frame never got unset by freeing the frame to indicate
- it was no longer in use, the translators never got destroyed, and
- thus the g729 licenses were not released. (closes issue #11892)
- Reported by: xrg Patches: 11892.diff uploaded by russell (license
- 2) Tested by: xrg, russell ........
-
-2008-01-31 22:12 +0000 [r101578-101580] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Forgot an !
-
- * apps/app_queue.c: A change I made to accommodate the "linear"
- strategy in trunk caused queue strategies to not be loaded from
- realtime queues. This commit fixes that. Thanks to jmls for
- pointing this problem out to me on IRC. This also contains some
- changes to S_OR where it should be used. Thanks to Qwell for
- pointing these out.
-
-2008-01-31 21:33 +0000 [r101577] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Fix a simple deadlock that was introduced
- _right_ before this code got merged into trunk. (closes issue
- #11895, reported by pj, patched by me)
-
-2008-01-31 21:31 +0000 [r101532-101576] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Handle the case of a NULL state_interface when
- checking a realtime member. Thanks to jmls for finding this
- issue.
-
- * /, res/res_monitor.c: Merged revisions 101531 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101531 | mmichelson | 2008-01-31 15:00:24 -0600 (Thu, 31 Jan
- 2008) | 10 lines 1. Prevent the addition of an extra '/' to the
- beginning of an absolute pathname. 2. If ast_monitor_change_fname
- is called and the new filename is the same as the old, then exit
- early and don't set the filename_changed field in the monitor
- structure. Setting it in this case was causing ast_monitor_stop
- to erroneously delete them. (closes issue #11741) Reported by:
- garlew Tested by: putnopvut ........
-
-2008-01-31 19:54 +0000 [r101483] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
- 101482 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101482 | qwell | 2008-01-31 13:52:49 -0600 (Thu, 31 Jan 2008) |
- 4 lines Solaris compat fixes for struct in_addr funkiness. Issue
- #11885, patch by snuffy. ........
-
-2008-01-31 19:43 +0000 [r101481] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 101480 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101480 | murf | 2008-01-31 12:30:37 -0700 (Thu, 31 Jan 2008) | 1
- line closes issue #11845; that's the one where there's a 1004
- byte cdr leak with every AMI Redirect to a zap channel ........
-
-2008-01-31 19:20 +0000 [r101416-101449] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 101433 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r101433 | russell | 2008-01-31 13:17:05 -0600 (Thu, 31
- Jan 2008) | 2 lines Add more missing locking of the agents list
- ... ........
-
- * /, channels/chan_agent.c: Merged revisions 101413-101414 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101413 | russell | 2008-01-31 13:04:52 -0600 (Thu, 31 Jan 2008)
- | 2 lines Add missing locking to the find_agent() function.
- ........ r101414 | russell | 2008-01-31 13:07:46 -0600 (Thu, 31
- Jan 2008) | 3 lines Move the locking from find_agent() into the
- agent dialplan function handler to ensure that the agent doesn't
- disappear while we're looking at it. ........
-
-2008-01-31 15:36 +0000 [r101393] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_realtime.c: Add missing braces. (closes issue #11886)
- Reported by: sergee Patches: func_realtime_fix-r101392.diff
- uploaded by sergee (license 138)
-
-2008-01-31 05:28 +0000 [r101373] Russell Bryant <russell@digium.com>
-
- * CHANGES: remove entry that is no longer in the tree
-
-2008-01-30 23:10 +0000 [r101344] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: The deprecation of "username" in favor of
- "defaultuser" for SIP peers unfortunately broke realtime
- configurations which still used the "username" field. This was
- taken care of properly when reading from realtime but was not
- handled properly when updating a realtime peer. This change also
- adds a deprecation NOTICE CLI message that will print if using
- the deprecated "username" field. (closes issue #11880) Reported
- by: cabal95 Patches: 11880.patch uploaded by putnopvut (license
- 60) Tested by: cabal95
-
-2008-01-30 20:08 +0000 [r101322] Olle Johansson <oej@edvina.net>
-
- * configs/cli.conf.sample: Clarify configuration file that can be
- misunderstood
-
-2008-01-30 19:03 +0000 [r101296] Jason Parker <jparker@digium.com>
-
- * apps/app_controlplayback.c: Allow disabling the default
- ffwd/rewind keys in the ControlPlayback application. This is done
- in a backward compat way. If the "default" key for ffwd/rew is
- used for another option (such as stop), the "default" is removed.
- (closes issue #11754) Reported by: johan Patches:
- app_controlplayback.c.option3.patch uploaded by johan (license
- 334) Tested by: johan, qwell
-
-2008-01-30 17:12 +0000 [r101271] Olle Johansson <oej@edvina.net>
-
- * configs/rtppage.conf.sample (removed), apps/app_rtppage.c
- (removed): Removing applications that wasn't ready for svn trunk,
- as trunk now has pre-release status.
-
-2008-01-30 16:54 +0000 [r101269] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Make the VoicemailUsersList AMI command
- consistent with other manager list functions. (closes issue
- #11874) Reported by: srt Patches: voicemail_ami-11847.patch
- uploaded by srt (license 378)
-
-2008-01-30 16:39 +0000 [r101267-101268] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/rtp.h, main/rtp.c: - doxygen fixes - change
- function to void because it always returned the same value and no
- one read it.
-
- * main/rtp.c: Formatting fixes
-
-2008-01-30 15:42 +0000 [r101224] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_rtppage.c: Get trunk to compile
-
-2008-01-30 15:42 +0000 [r101223] Joshua Colp <jcolp@digium.com>
-
- * /, main/slinfactory.c: Merged revisions 101222 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101222 | file | 2008-01-30 11:41:04 -0400 (Wed, 30 Jan 2008) | 4
- lines Fix an issue where if a frame of higher sample size
- preceeded a frame of lower sample size and ast_slinfactory_read
- was called with a sample size of the combined values or higher a
- crash would happen. (closes issue #11878) Reported by: stuarth
- ........
-
-2008-01-30 15:36 +0000 [r101221] Olle Johansson <oej@edvina.net>
-
- * CHANGES: Update CHANGES with rtppage
-
-2008-01-30 15:35 +0000 [r101220] Jason Parker <jparker@digium.com>
-
- * /, configs/extensions.conf.sample: Merged revisions 101219 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11875) ........ r101219 | qwell | 2008-01-30 09:34:37
- -0600 (Wed, 30 Jan 2008) | 5 lines Change default config to use
- descending channel order of groups, rather than ascending. Fixes
- a potential source of confusion in glare-type situations. Issue
- 11875, reported by JimVanM. ........
-
-2008-01-30 15:30 +0000 [r101218] Olle Johansson <oej@edvina.net>
-
- * configs/rtppage.conf.sample (added), apps/app_rtppage.c (added):
- Add rtppage() application to do multicast or unicast RTP paging
- to SIP phones. (closes issue #11797) Reported by: macbrody
- Patches: app_rtppage-20080130.c uploaded by macbrody (license
- 352)
-
-2008-01-30 15:27 +0000 [r101217] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 101216 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan
- 2008) | 5 lines Fix a logic error with regards to autofill. Prior
- to this change, it was possible for a caller to go out of turn if
- autofill were enabled and callers ahead in the queue were
- attempting to call a member. This change fixes this. ........
-
-2008-01-30 12:48 +0000 [r101196] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: simplify this code and eliminate the return
- value cast that is no longer necessary
-
-2008-01-30 11:27 +0000 [r101153-101154] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, include/asterisk/channel.h: Constifying the
- interface to get pvt_ids in the bridge, based on suggestion from
- (const char *) Kevin. Thanks!
-
- * /, channels/chan_sip.c: Merged revisions 101152 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101152 | oej | 2008-01-30 12:20:31 +0100 (Ons, 30 Jan 2008) | 7
- lines Stop musiconhold on attended transfer. (closes issue
- #11872) Reported by: gareth Patches: svn-101018.patch uploaded by
- gareth (license 208) ........
-
-2008-01-30 00:58 +0000 [r101126] Jason Parker <jparker@digium.com>
-
- * CHANGES: Fix a typo
-
-2008-01-30 00:04 +0000 [r101082] Russell Bryant <russell@digium.com>
-
- * CHANGES, apps/app_speech_utils.c: Add the 'n' option to
- SpeechBackground, which has the application not answer the
- channel if it has not already been answered. (closes SPD-51)
-
-2008-01-29 23:59 +0000 [r101081] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /, build_tools/make_version: Merged revisions 101080 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r101080 | dhubbard | 2008-01-29 17:50:42 -0600 (Tue, 29
- Jan 2008) | 1 line updated build_tools to handle the autotag
- directory structure changes; changes related to BE-353. Patch by
- The Russell and reviewed by The Me. ........
-
-2008-01-29 23:02 +0000 [r101036] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 101035 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r101035 | mmichelson | 2008-01-29 17:02:03 -0600 (Tue, 29 Jan
- 2008) | 3 lines Remove a memory leak from updating realtime
- queues ........
-
-2008-01-29 22:04 +0000 [r101018] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_curl.c: Oops, a sizeof error
-
-2008-01-29 19:41 +0000 [r100974] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 100973 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100973 | mmichelson | 2008-01-29 13:39:00 -0600 (Tue, 29 Jan
- 2008) | 6 lines Fixing an erroneous return value returned when
- attempting to pause or unpause a queue member fails. Fixes
- BE-366, thanks to John Bigelow for writing the patch. ........
-
-2008-01-29 17:44 +0000 [r100933] Russell Bryant <russell@digium.com>
-
- * /, main/Makefile: Merged revisions 100932 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100932 | russell | 2008-01-29 11:43:41 -0600 (Tue, 29 Jan 2008)
- | 4 lines Fix the last couple of issues related to building from
- a path that contains spaces. (closes issue #11834) ........
-
-2008-01-29 17:42 +0000 [r100931] Jason Parker <jparker@digium.com>
-
- * /, channels/misdn_config.c: Merged revisions 100930 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r100930 | qwell | 2008-01-29 11:41:43 -0600 (Tue, 29 Jan
- 2008) | 6 lines Initialize an array to 0s if config option not
- specified. (closes issue #11860) Patches:
- misdn_get_config.v1.diff uploaded by IgorG (license 20) ........
-
-2008-01-29 17:22 +0000 [r100900-100928] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 100922 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100922 | russell | 2008-01-29 11:21:33 -0600 (Tue, 29 Jan 2008)
- | 3 lines Use GNU make magic instead of shell magic to escape
- spaces in the working directory. (related to issue #11834)
- ........
-
- * Makefile, /: Merged revisions 100882 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100882 | russell | 2008-01-29 11:06:43 -0600 (Tue, 29 Jan 2008)
- | 6 lines Fix building Asterisk when the working path has spaces
- in it. (closes issue #11834) Reported by: spendergrass Patched
- by: me ........
-
-2008-01-29 16:14 +0000 [r100843] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 100835 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100835 | qwell | 2008-01-29 10:10:00 -0600 (Tue, 29 Jan 2008) |
- 5 lines Allow zap groups above 30 to work properly. (closes issue
- #11590) Reported by: tbsky ........
-
-2008-01-29 15:30 +0000 [r100833] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make externip work as documented. If no port
- is specified it will use the value of bindport instead of always
- being 5060. (closes issue #11858) Reported by: hmodes
-
-2008-01-29 10:50 +0000 [r100794-100795] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 100793 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r100793 | crichter | 2008-01-29 11:36:19 +0100 (Di, 29
- Jan 2008) | 1 line fixed potential segfault in misdn show
- channels CLI command ........
-
- * channels/chan_misdn.c, /: Merged revisions 96199 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r96199 | crichter | 2008-01-03 13:12:27 +0100 (Do, 03
- Jan 2008) | 1 line make sure frame is completely clean, before we
- send it to asterisk as DTMF. If we don't make it clean, it
- happens that one way audio occurs.. ........
-
-2008-01-29 09:18 +0000 [r100741-100767] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 100740 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100740 | oej | 2008-01-29 09:26:48 +0100 (Tis, 29 Jan 2008) | 8
- lines (closes issue #11736) Reported by: MVF Patches:
- bug11736-2.diff uploaded by oej (license 306) Tested by: oej,
- MVF, revolution (russellb: This was the showstopper for the
- release.) ........
-
- * channels/chan_sip.c: Removing code that wasn't supposed to be
- there at all, only at an experimental stage before I found
- another solution. Thanks Kevin, for reminding me.
-
-2008-01-28 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta2 released.
-
-2008-01-28 21:11 +0000 [r100679] Jason Parker <jparker@digium.com>
-
- * build_tools/menuselect-deps.in, configs/vpb.conf.sample (added),
- doc/tex/channelvariables.tex, makeopts.in: Reintroduce more
- chan_vpb stuff that was removed in r100421 and r100422
-
-2008-01-28 21:07 +0000 [r100678] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_vpb.cc (added), configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- channels/Makefile: Re-inserting chan_vpb into trunk.
-
-2008-01-28 21:05 +0000 [r100677] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /: Merged revisions 100675 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100675 | tilghman | 2008-01-28 15:02:02 -0600 (Mon, 28 Jan 2008)
- | 2 lines WaitExten didn't handle AbsoluteTimeout properly (went
- to 't' instead of 'T') ........
-
-2008-01-28 21:02 +0000 [r100676] Jason Parker <jparker@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 100672 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11795) ........ r100672 | qwell | 2008-01-28 14:42:43
- -0600 (Mon, 28 Jan 2008) | 7 lines When using ODBC_STORAGE, make
- sure we put greeting files into the database like we do with the
- others. Issue #11795 Reported by: dimas Patches: vmgreet.patch
- uploaded by dimas (license 88) ........
-
-2008-01-28 20:40 +0000 [r100632-100671] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix up some T38 state change issues. (closes
- issue #11630) Reported by: dimas Patches: v2-sip-t38state.patch
- uploaded by dimas (license 88)
-
- * channels/chan_sip.c: Fix up two scheduling issues. In one
- instance a scheduled item was not deleted when it should have
- been and in the other it was scheduled again when it shouldn't
- have been.
-
-2008-01-28 18:41 +0000 [r100630-100631] Russell Bryant <russell@digium.com>
-
- * main/features.c: Merge rev 100626 from Asterisk 1.4. The svnmerge
- of this commit was a NoOp, since res_features doesn't exist in
- trunk. Thanks to qwell for pointing it out!
-
- * /, channels/chan_sip.c: Merged revisions 100629 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008)
- | 5 lines For some reason, the use of this strdupa() is leading
- to memory corruption on freebsd sparc64. This trivial workaround
- fixes it. (closes issue #10300, closes issue #11857, reported by
- mattias04 and Home-of-the-Brave) ........
-
-2008-01-28 18:27 +0000 [r100628] Tilghman Lesher <tlesher@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/logger.c: Normalize the detection for execinfo, so that
- Linux (glibc) and other platforms with libexecinfo will generate
- inline stack backtraces correctly.
-
-2008-01-28 18:27 +0000 [r100627] Russell Bryant <russell@digium.com>
-
- * /: Merged revisions 100626 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100626 | russell | 2008-01-28 12:26:31 -0600 (Mon, 28 Jan 2008)
- | 7 lines Fix a crash in ast_masq_park_call() (issue #11342)
- Reported by: DEA Patches: res_features-park.txt uploaded by DEA
- (license 3) ........
-
-2008-01-28 18:24 +0000 [r100625] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 100624 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100624 | qwell | 2008-01-28 12:23:09 -0600 (Mon, 28 Jan 2008) |
- 1 line Correct a comment which made little/no sense. ........
-
-2008-01-28 17:21 +0000 [r100565-100582] Russell Bryant <russell@digium.com>
-
- * main/channel.c, channels/chan_local.c, /,
- include/asterisk/channel.h: Merged revisions 100581 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28
- Jan 2008) | 9 lines Make some deadlock related fixes. These bugs
- were discovered and reported internally at Digium by Steve Pitts.
- - Fix up chan_local to ensure that the channel lock is held
- before the local pvt lock. - Don't hold the channel lock when
- executing the timing function, as it can cause a deadlock when
- using chan_local. This actually changes the code back to be how
- it was before the change for issue #10765. But, I added some
- other locking that I think will prevent the problem reported
- there, as well. ........
-
- * main/pbx.c: Clean up some formatting, and simplify a bit of code
- using ast_str
-
-2008-01-28 13:57 +0000 [r100549] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't do a network byte order conversion
- when setting the socket's port variable to that of bindaddr's. It
- is already in the correct network byte order. (closes issue
- #11800) Reported by: hmodes
-
-2008-01-28 04:43 +0000 [r100514-100533] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Make a couple more uses of ARRAY_LEN, and convert
- some spaces to tabs
-
- * main/channel.c: - Simplify a line with ARRAY_LEN() - Make a few
- little formatting changes
-
- * main/channel.c: These readlocks always fail for me on my mac, and
- I saw it happen again today on another mac. We ignore the return
- value of locking operations almost everywhere in Asterisk. So,
- ignore these, as well, so Asterisk will actually work on systems
- where this is occurring while I look into what the issue is.
-
-2008-01-27 23:14 +0000 [r100488-100497] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c, include/asterisk/sched.h,
- channels/chan_iax2.c: With the switch to the ast_sched_replace*
- API in trunk, we lose the correction that was just merged from
- 1.4, so this is a changeover to those APIs to use the macro
- versions, so that we properly detect errors from ast_sched_del,
- instead of simply ignoring the return values.
-
- * main/cdr.c, channels/chan_misdn.c, main/dnsmgr.c, /,
- channels/chan_sip.c, channels/chan_h323.c,
- include/asterisk/sched.h, main/file.c, pbx/pbx_dundi.c,
- channels/chan_iax2.c, main/rtp.c, channels/chan_mgcp.c: Merged
- revisions 100465 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008)
- | 11 lines When deleting a task from the scheduler, ignoring the
- return value could possibly cause memory to be accessed after it
- is freed, which causes all sorts of random memory corruption.
- Instead, if a deletion fails, wait a bit and try again (noting
- that another thread could change our taskid value). (closes issue
- #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt
- uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan,
- stuarth` ........
-
-2008-01-25 22:54 +0000 [r100421-100422] Jason Parker <jparker@digium.com>
-
- * doc/tex/channelvariables.tex: Get rid of that last little bit.
-
- * build_tools/menuselect-deps.in, configs/vpb.conf.sample
- (removed), makeopts.in: Remove more remnants of chan_vpb
-
-2008-01-25 22:39 +0000 [r100419-100420] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_vpb.cc (removed), configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- channels/Makefile, .cleancount: Removing chan_vpb from the tree
-
-2008-01-25 21:26 +0000 [r100379] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 100378 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100378 | qwell | 2008-01-25 15:24:49 -0600 (Fri, 25 Jan 2008) |
- 2 lines This would have never been true, since we're passing
- (sizeof(req.data) - 1) as the len to recvfrom(). ........
-
-2008-01-25 20:51 +0000 [r100361] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_rpt.c: correct a real problem and silence an annoying
- compiler warning
-
-2008-01-25 14:53 +0000 [r100344] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Insure that we are not going to pass a NULL
- pointer to add_to_interfaces. (closes issue #11840) Reported by:
- junky
-
-2008-01-25 02:52 +0000 [r100325] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c, include/asterisk/dial.h: Add an API call that steals
- the answered channel so that a destruction of the dialing
- structure does not hang it up.
-
-2008-01-24 22:58 +0000 [r100307] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile, build_tools/make_defaults_h: Use the set ASTDBDIR as
- the default, too
-
-2008-01-24 22:36 +0000 [r100305-100306] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/app.h: ummm... might be good if this macro
- argument was actually used :-)
-
- * include/asterisk/app.h: add the ability to define a structure
- type for argument parsing when it would be useful to be able to
- pass it between functions
-
-2008-01-24 22:02 +0000 [r100266] James Golovich <james@gnuinter.net>
-
- * channels/chan_sip.c: Fix simple whitespace issue
-
-2008-01-24 22:01 +0000 [r100265] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/app.h, /: Merged revisions 100264 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r100264 | kpfleming | 2008-01-24 15:57:41 -0600 (Thu, 24
- Jan 2008) | 2 lines make these macros not assume that the only
- other field in the structure is 'argc'... this is true when
- someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
- to define your own structure as long as it has the right fields
- ........
-
-2008-01-24 20:32 +0000 [r100245] Joshua Colp <jcolp@digium.com>
-
- * main/features.c: Minor cosmetic change...
-
-2008-01-24 18:35 +0000 [r100224] James Golovich <james@gnuinter.net>
-
- * main/astmm.c: Increase the size of filenames stored when astmm is
- used. If the path length was long they would be truncated and
- grouped together with whatever matches
-
-2008-01-24 17:47 +0000 [r100206] Joshua Colp <jcolp@digium.com>
-
- * configs/rtp.conf.sample, CHANGES, main/rtp.c: Merge in strictrtp
- branch. This adds a strictrtp option to rtp.conf which drops
- packets that do not come from the remote party. (closes issue
- #8952) Reported by: amorsen
-
-2008-01-24 17:24 +0000 [r100169] Russell Bryant <russell@digium.com>
-
- * /, main/asterisk.c: Merged revisions 100164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100164 | russell | 2008-01-24 11:22:09 -0600 (Thu, 24 Jan 2008)
- | 2 lines Update main Asterisk copyright info to 2008 ........
-
-2008-01-24 16:47 +0000 [r100121-100139] Jason Parker <jparker@digium.com>
-
- * /, res/res_phoneprov.c, main/acl.c: Merged revisions 100138 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r100138 | qwell | 2008-01-24 10:41:29 -0600 (Thu, 24 Jan 2008) |
- 6 lines Fix compilation on Solaris. (closes issue #11832)
- Patches: bug-11832.diff uploaded by snuffy (license 35) ........
-
- * channels/chan_sip.c, main/features.c: Move chan_local dependency
- into places (only one) that previously depended on res_features,
- and used local channels
-
-2008-01-24 15:54 +0000 [r100076-100112] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c,
- channels/chan_mgcp.c: Remove dependency on res_features from some
- channel drivers. It is now part of the core and no longer exists
- as a module.
-
- * main/channel.c: Some more cosmetic changes.
-
- * main/channel.c: Add some spacing.
-
- * main/dial.c: Test hopefully over.
-
- * main/dial.c: Testing something...
-
-2008-01-24 00:04 +0000 [r100057] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: fix flag bit definitions to make code from
- issue #11049 actually work; along the way, clarify comments and
- add some dummy flag definitions for other multi-bit flags to
- hopefully stop this from happening in the future (closes issue
- #11049)
-
-2008-01-23 23:09 +0000 [r100039] Jason Parker <jparker@digium.com>
-
- * res/res_features.c (removed), main/Makefile, main/features.c
- (added), include/asterisk/_private.h, CHANGES, .cleancount,
- main/asterisk.c, main/loader.c, include/asterisk/features.h: Move
- code from res_features into (new file) main/features.c
-
-2008-01-23 22:00 +0000 [r100021] Russell Bryant <russell@digium.com>
-
- * CREDITS: Add Sergey Tamkovich to CREDITS. Thank you for your
- contributions!
-
-2008-01-23 21:11 +0000 [r99979-99980] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 99978 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99978 | oej | 2008-01-23 22:07:16 +0100 (Ons, 23 Jan 2008) | 7
- lines Second attempt. Don't change invitestate when receiving 18x
- messages in CANCEL state. (issue #11736) Reported by: MVF Patch
- by oej. ........
-
- * /, channels/chan_sip.c: Merged revisions 99977 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9
- lines Make sure we don't cancel destruction on calls in CANCEL
- state, even if we get 183 while waiting for answer on our CANCEL.
- (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by
- oej (license 306) Tested by: MVF ........
-
-2008-01-23 20:26 +0000 [r99976] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_externalivr.c: Merged revisions 99975 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r99975 | mmichelson | 2008-01-23 14:25:00 -0600 (Wed, 23
- Jan 2008) | 3 lines Fixing a typo. ........
-
-2008-01-23 17:48 +0000 [r99922-99924] Russell Bryant <russell@digium.com>
-
- * /, apps/app_chanspy.c: Merged revisions 99923 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) |
- 8 lines ChanSpy issues a beep when it starts at the beginning of
- a list of channels to potentially spy on. However, if there were
- no matching channels, it would beep at you over and over, which
- is pretty annoying. Now, it will only beep once in the case that
- there are no channels to spy on, but it will still beep again
- once it reaches the beginning of the channel list again. (closes
- issue #11738, patched by me) ........
-
- * main/tcptls.c: Fix tcptls build when openssl isn't installed
- (closes issue #11813) Reported by: tzafrir Patches:
- asterisk-tcptls.diff.txt uploaded by jamesgolovich (license 176)
-
-2008-01-23 17:27 +0000 [r99920] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: since echo canceler parameters in Zaptel are
- now signed integers, allow them during parsing
-
-2008-01-23 15:23 +0000 [r99860] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_h323.c: Progress messages don't work (closes issue
- #10497) Reported by: pj Patches: h323-announces-r99483.diff
- uploaded by sergee (license 138) Tested by: pj
-
-2008-01-23 10:18 +0000 [r99839] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: - Add a few comments to sip_xmit - Make sure
- that we are aware of a pending INVITE even if we're using TCP
-
-2008-01-23 05:29 +0000 [r99696-99818] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Coding guidelines fixups
-
- * /, apps/app_voicemail.c: Merged revisions 99777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008)
- | 8 lines When we reset the password via an external command, we
- should also reset the password stored in the in-memory list, too
- (otherwise it doesn't really take effect). (closes issue #11809)
- Reported by: davetroy Patches: fix_externpass.diff uploaded by
- davetroy (license 384) ........
-
- * /, res/res_odbc.c: Merged revisions 99775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99775 | tilghman | 2008-01-22 22:20:15 -0600 (Tue, 22 Jan 2008)
- | 2 lines Oops, should have checked for a NULL obj, here, too
- ........
-
- * res/res_config_ldap.c: Coding guidelines cleanup
-
- * /, main/acl.c: Merged revisions 99718 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99718 | tilghman | 2008-01-22 18:56:06 -0600 (Tue, 22 Jan 2008)
- | 2 lines Just confirmed that all current platforms need this
- header file ........
-
- * /: Oops
-
- * /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, doc/ldap.txt (added),
- configure.ac, configs/res_ldap.conf.sample (added),
- res/res_config_ldap.c (added), CHANGES, makeopts.in,
- contrib/scripts/asterisk.ldap-schema (added),
- contrib/scripts/asterisk.ldif (added): Add res_config_ldap for
- realtime LDAP engine. (closes issue #5768) Reported by: mguesdon
- Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon
- (license 121) res_ldap.conf.sample uploaded by suretec (license
- 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70)
- asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested
- by: oej, mguesdon, suretec, cthorner
-
-2008-01-22 21:09 +0000 [r99647-99653] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 99652 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4
- lines Thanks to Russell's education I realize that BUFSIZ has
- changed since I learned the C language over 20 years ago...
- Resetting chan_sip to the size of BUFSIZ that I expected in my
- old head to avoid too heavy memory allocations on some systems.
- ........
-
- * doc/tex/channelvariables.tex, CHANGES: Documentation updates for
- BRIDGEPVTCALLID
-
-2008-01-22 20:42 +0000 [r99646] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/acl.c: Merged revisions 99643 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99643 | tilghman | 2008-01-22 14:34:55 -0600 (Tue, 22 Jan 2008)
- | 2 lines Fix the defines for OS X (and Solaris, too) ........
-
-2008-01-22 20:41 +0000 [r99645] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Make sure the command is not just present but is
- also configured to be executed
-
-2008-01-22 20:35 +0000 [r99644] Olle Johansson <oej@edvina.net>
-
- * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h:
- Add a generic function to set the bridged call PVT unique id
- string as a channel variable BRIDGEPVTCALLID This is important
- for call tracing in log files and CDRs, so that the SIP callID
- can be traced along servers. The CHANNEL dialplan function won't
- work here, since the outbound channel is gone when we need the
- Call-ID. Other channel drivers may now implement the same
- function :-), but this patch only supports chan_sip.so. Inspired
- by (issue #11816) Reported by: ctooley Patch by oej
-
-2008-01-22 20:33 +0000 [r99642] Russell Bryant <russell@digium.com>
-
- * configs/cli.conf.sample (added), CHANGES, main/asterisk.c: Change
- the Asterisk CLI startup commands feature to read commands to run
- from cli.conf after a discussion on the -dev list.
-
-2008-01-22 17:46 +0000 [r99595-99596] Olle Johansson <oej@edvina.net>
-
- * channels/chan_local.c, /, res/res_features.c,
- channels/chan_agent.c, apps/app_followme.c: Merged revisions
- 99594 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3
- lines Add more dependencies on chan_local and add a note to the
- description of chan_local so that people don't disable it in
- menuselect just to clean up. ........
-
- * apps/app_dial.c, /: Merged revisions 99592 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5
- lines Add dependency on chan_local to app_dial. Dial still runs
- without chan_local, but will be missing forwarding functionality.
- ........
-
-2008-01-22 17:15 +0000 [r99559] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/acl.c: Merged revisions 99540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008)
- | 7 lines Ensure that we can get an address even when we don't
- have a default route. (closes issue #9225) Reported by: junky
- Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
- (license 14) Tested by: oej, loloski, sergee ........
-
-2008-01-22 16:55 +0000 [r99542] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Point out a bug in some debug counter
- handling
-
-2008-01-22 15:25 +0000 [r99464-99521] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Add authentication options to the SIP
- dialstring. Documentation follows separately (issue #11587)
- Reported by: sobomax Patches: chan_sip.c-trunk.diff uploaded by
- sobomax (license 359)
-
- * configs/sip.conf.sample: Documentation updates
-
- * doc/siptls.txt: Small fixes
-
- * main/tcptls.c, channels/chan_zap.c, main/abstract_jb.c,
- include/asterisk/tcptls.h: Doxygen updates
-
-2008-01-21 23:56 +0000 [r99427] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 99426 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21
- Jan 2008) | 12 lines Fixing an issue wherein monitoring local
- channels was not possible. During a channel masquerade, the
- monitors on the two channels involved are swapped. In 99% of the
- cases this results in the desired effect. However, if monitoring
- a local channel, this caused the monitor which was on the local
- channel to get moved onto a channel which is immediately hung up
- after the masquerade has completed. By swapping the monitors
- prior to the masquerade, we avoid the problem by tricking the
- masquerade into placing the monitor back onto the channel where
- we want it. During the investigation of the issue, the channel's
- monitor was the only thing that was swapped in such a manner
- which did not make sense to have done. All other variable
- swapping made sense. ........
-
-2008-01-21 23:25 +0000 [r99424] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Fix distinctive ring detection. Reported by:
- milazzo Patches: drings.diff uploaded by milazzo (license 383)
- Closes issue #11799
-
-2008-01-21 22:32 +0000 [r99406] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample, apps/app_queue.c: Adding the
- QUEUENAME variable to the variables set using the setqueuevar
- option in queues.conf. Suggestion comes from Shaun2222 on IRC.
-
-2008-01-21 21:11 +0000 [r99382-99384] Olle Johansson <oej@edvina.net>
-
- * channels/chan_console.c: Remove compiler warning for
- uninitialized variable
-
- * channels/chan_sip.c: Doxygen updates. The TCP/TLS code was
- committed without any doxygen obviously. Tss tss.
-
- * channels/chan_sip.c: Updating doxygen
-
-2008-01-21 18:15 +0000 [r99350] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h, /, res/res_odbc.c,
- configs/res_odbc.conf.sample: Merged revisions 99341 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21
- Jan 2008) | 8 lines Permit the user to specify number of seconds
- that a connection may remain idle, which fixes a crash on
- reconnect with the MyODBC driver. (closes issue #11798) Reported
- by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
- uploaded by Corydon76 (license 14) Tested by: mvanbaak ........
-
-2008-01-21 16:02 +0000 [r99302] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 99301 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4
- lines Bump the buffer size for Via headers up to 512. There are
- some exceptionally large Via headers out there. (closes issue
- #11783) Reported by: ofirroval ........
-
-2008-01-21 07:02 +0000 [r99280] Olle Johansson <oej@edvina.net>
-
- * CREDITS: Update
-
-2008-01-21 03:54 +0000 [r99265] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Change over to using ast_debug so these
- debug messages don't always show up.
-
-2008-01-20 07:28 +0000 [r99166-99248] Russell Bryant <russell@digium.com>
-
- * channels/chan_console.c: Add a "console active" CLI command,
- which lets you find out which console device is currently active
- for the Asterisk CLI, or to set it. Also, knock multiple device
- support off of the to-do list.
-
- * configs/console.conf.sample: correct the name of a CLI command
- for getting available device names
-
- * configs/console.conf.sample, channels/chan_console.c: Merge
- changes from team/russell/console_devices - Add support for
- multiple devices. All devices are configured in console.conf. -
- Add "console list devices" CLI command to show configured
- devices. Also, changed the old "list devices" to be "list
- available", which queries PortAudio for all audio devices that
- are available for use.
-
- * /, main/slinfactory.c: Merged revisions 99187 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) |
- 4 lines Fix a couple of memory leaks with frame handling.
- Specifically, ast_frame_free() needed to be called on the frame
- that came from the translator to signed linear. ........
-
- * README: Add Cygwin as an "other" platform that is supported
-
- * README: Various README updates
-
-2008-01-18 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-beta1 released.
-
-2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell@digium.com>
-
- * CREDITS, include/asterisk/http.h, main/tcptls.c (added),
- main/manager.c, channels/chan_sip.c, doc/siptls.txt (added),
- main/Makefile, main/http.c, include/asterisk/tcptls.h (added),
- configs/sip.conf.sample, CHANGES: Merge changes from
- team/group/sip-tcptls This set of changes introduces TCP and TLS
- support for chan_sip. There are various new options in
- configs/sip.conf.sample that are used to enable these features.
- Also, there is a document, doc/siptls.txt that describes some
- things in more detail. This code was implemented by Brett Bryant
- and James Golovich. It was reviewed by Joshua Colp and myself. A
- number of other people participated in the testing of this code,
- but since it was done outside of the bug tracker, I do not have
- their names. If you were one of them, thanks a lot for the help!
- (closes issue #4903, but with completely different code that what
- exists there.)
-
- * main/frame.c, /, include/asterisk/translate.h: Merged revisions
- 99081 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) |
- 9 lines Revert adding the packed attribute, as it really doesn't
- make sense why that would do any good. Fix the real bug, which is
- to do the check to see if the frame came from a translator at the
- beginning of ast_frame_free(), instead of at the end. This
- ensures that it always gets checked, even if none of the parts of
- the frame are malloc'd, and also ensures that we aren't looking
- at free'd memory in the case that it is a malloc'd frame. (closes
- issue #11792, reported by explidous, patched by me) ........
-
- * /, include/asterisk/translate.h: Merged revisions 99079 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) |
- 4 lines Since we're relying on the offset between the frame and
- the beginning of the translator pvt struct, set the packed
- attribute to make sure we get to the right place. (potential fix
- for issue #11792) ........
-
-2008-01-18 16:58 +0000 [r99026] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: This should at least temporarily fix a
- problem where the 't' Dial option is incorrectly passed to the
- transferee when built-in attended transfers are used. There is
- still a problem with 'T', but better to fix some problems than no
- problems while we work on it. (closes issue #7904) Reported by:
- k-egg Patches: transfer-fix-trunk-r97657.diff uploaded by sergee
- (license 138) Tested by: sergee, otherwiseguy
-
-2008-01-18 06:58 +0000 [r99015-99018] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_odbc.c: Convert func_odbc to use SQLExecDirect for
- speed (closes issue #10723) Reported by: mnicholson Patches:
- func-odbc-direct-execute1.diff uploaded by mnicholson (license
- 96) Tested by: Corydon76, mnicholson, falves11
-
- * res/res_odbc.c: Permit username and password to be NULL (which
- enables pass-through from the layer above). Reported by: lurcher
- Patch by: tilghman (Closes issue #11739)
-
- * funcs/func_cut.c: Reset default CUT delimiter back to '-'
-
-2008-01-17 23:28 +0000 [r99006-99011] Russell Bryant <russell@digium.com>
-
- * channels/chan_console.c: Make the output of "console list
- devices" a bit prettier.
-
- * channels/chan_console.c: List which devices are inputs and
- outputs in "console list devices"
-
- * main/channel.c: Add AST_FORMAT_SLINEAR16 to the list for
- ast_best_codec()
-
- * main/frame.c, /, channels/chan_iax2.c, include/asterisk/frame.h:
- Merged revisions 99004 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) |
- 10 lines Have IAX2 optimize the codec translation path just like
- chan_sip does it. If the caller's codec is in our codec list,
- move it to the top to avoid transcoding. (closes issue #10500)
- Reported by: stevedavies Patches: iax-prefer-current-codec.patch
- uploaded by stevedavies (license 184)
- iax-prefer-current-codec.1.4.patch uploaded by stevedavies
- (license 184) Tested by: stevedavies, pj, sheldonh ........
-
-2008-01-17 22:22 +0000 [r99002] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixing trunk IMAP build (closes issue
- #11788) Reported by: DEA Patches: vm-imap-build-fix.txt uploaded
- by DEA (license 3)
-
-2008-01-17 20:51 +0000 [r98998] Jason Parker <jparker@digium.com>
-
- * Makefile, build_tools/cflags.xml, channels/chan_zap.c,
- main/dsp.c, configs/zapata.conf.sample: Add several busy
- detection related defines to menuselect. Allow better busy detect
- debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and
- BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches:
- busydetect_enhancement.patch uploaded by agx (license 298)
- busydetect-r94975.diff uploaded by sergee (license 138)
- Additional changes/cleanup by me.
-
-2008-01-17 16:33 +0000 [r98993-98994] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: state_interface could be NULL, so use the
- never-NULL cur->state_interface for this check
-
- * apps/app_queue.c: Get the device state of the state interface
- instead of the interface when creating a new queue member. Thanks
- to Atis Lezdins for bringing this up on the Asterisk-Dev mailing
- list.
-
-2008-01-17 16:21 +0000 [r98992] Jason Parker <jparker@digium.com>
-
- * /, configs/zapata.conf.sample: Merged revisions 98991 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #11784) ........ r98991 | qwell | 2008-01-17 10:19:46 -0600
- (Thu, 17 Jan 2008) | 4 lines Add a clarification about the
- immediate= option of zapata.conf Issue 11784, patch by klaus3000.
- ........
-
-2008-01-17 16:17 +0000 [r98989-98990] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, configs/zapata.conf.sample: major
- reliability and performance improvement in VWMI monitoring for
- FXO ports (code by markster, me and dbailey)
-
- * res/res_config_curl.c: resolve (valid) compiler warning about
- variable that could be used before being initialized
-
-2008-01-17 03:09 +0000 [r98988] Terry Wilson <twilson@digium.com>
-
- * res/res_phoneprov.c, doc/tex/phoneprov.tex,
- configs/phoneprov.conf.sample: Update res_phoneprov to default to
- setting the SERVER variable to the IP the HTTP request for the
- config came in on and the SERVER_PORT to the bindport setting in
- sip.conf. I've left in the ability to override these options,
- because I can't always guess how someone might decide to do
- something weird with what is available to them--although needing
- to is pretty unlikely. Documentation was updated to reflect
- preference for not setting serveraddr, serveriface, or
- serverport. Tested on Linux and OS X.
-
-2008-01-17 00:13 +0000 [r98987] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Change the way the new filter feature
- works, by allowing it to be a column NOT logged into the
- database. This will allow more granularity of a decision
- evaluated in the dialplan, then takes effect when posting the
- CDR.
-
-2008-01-17 00:05 +0000 [r98986] Russell Bryant <russell@digium.com>
-
- * CHANGES, main/asterisk.c: Add support for an easy way to
- automatically execute some Asterisk CLI commands immediately at
- startup. Any commands in the startup_commands file in the
- Asterisk config diretory will get executed. (closes issue #11781)
- Reported by: jamesgolovich Patches: asterisk-startupcmds.diff.txt
- uploaded by jamesgolovich (license 176) -- With some changes by
- me.
-
-2008-01-16 23:08 +0000 [r98985] Jason Parker <jparker@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4: Change AST_EXT_TOOL_CHECK to attempt to build
- against <package>_LIB, per recommendations from Russell.
-
-2008-01-16 22:36 +0000 [r98984] Tilghman Lesher <tlesher@digium.com>
-
- * CHANGES: Info about res_config_curl
-
-2008-01-16 22:20 +0000 [r98981] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_curl.c (added), main/utils.c: New module
- res_config_curl (closes issue #11747) Reported by: Corydon76
- Patches: res_config_curl.c uploaded by Corydon76 (license 14)
- 20080116__bug11747.diff.txt uploaded by Corydon76 (license 14)
- Tested by: jmls
-
-2008-01-16 21:53 +0000 [r98978] Russell Bryant <russell@digium.com>
-
- * CREDITS, channels/chan_sip.c, configs/sip.conf.sample: Merge the
- changes from issue #10665 from the team/group/sip_session_timers
- branch. This set of changes introduces SIP session timers support
- (RFC 4028). In short, this prevents stuck SIP sessions that were
- not properly torn down due to network or endpoint failures during
- an established SIP session. To quote some of the documentation
- supplied with the patch: "The SIP Session-Timers is an extension
- of the SIP protocol that allows end-points and proxies to refresh
- a session periodically. The sessions are kept alive by sending a
- RE-INVITE or UPDATE request at a negotiated interval. If a
- session refresh fails then all the entities that support Session-
- Timers clear their internal session state. In addition, UAs
- generate a BYE request in order to clear the state in the proxies
- and the remote UA (this is done for the benefit of SIP entities
- in the path that do not support Session-Timers)." (closes issue
- #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by
- rjain (license 226) chan_sip.c.diff uploaded by rjain (license
- 226) sip.conf.sample.diff uploaded by rjain (license 226)
- proc_422_rsp_comment.diff uploaded by rjain (license 226)
- chan_sip.c.cache.diff uploaded by rjain (license 226)
- chan_sip.memalloc uploaded by rjain (license 226)
- chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches
- tracked in team/group/sip_session_timers, with some additional
- fixes by russell and oej. Tested by: jtodd, rjain, loloski
-
-2008-01-16 19:41 +0000 [r98968-98971] Jason Parker <jparker@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- Partially revert r93898, because it broke the way netsnmp was
- being detected. rizzo, do you want to discuss so we can rethink
- this, or do you have another way?
-
- * CHANGES: Add note about new update.log to CHANGES, by request of
- jmls and further prodding by jsmith.
-
- * Makefile, /: Add logging for 'make update' command (also fixes
- updates in some places). Issue #11766, initial patch by jmls.
-
-2008-01-16 17:51 +0000 [r98967] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 98966 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6
- lines Add missing NULLs at end of two ast_load_realtimes. (closes
- issue #11769) Reported by: tequ Patches: chaniax.patch uploaded
- by dimas (license 88) ........
-
-2008-01-16 17:21 +0000 [r98965] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 98964 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16
- Jan 2008) | 10 lines Fix a deadlock in chan_local in
- local_hangup. There was contention because the local_pvt was held
- and it was attempting to lock a channel, which is the incorrect
- locking order. (closes issue #11730) Reported by: UDI-Doug
- Patches: 11730.patch uploaded by putnopvut (license 60) Tested
- by: UDI-Doug ........
-
-2008-01-16 16:06 +0000 [r98962] Terry Wilson <twilson@digium.com>
-
- * res/res_phoneprov.c: Make users list static
-
-2008-01-16 15:09 +0000 [r98954-98961] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c, /: Merged revisions 98960 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6
- lines Introduce a lock into the dialing API that protects it when
- destroying the structure. (closes issue #11687) Reported by:
- callguy Patches: 11687.diff uploaded by file (license 11)
- ........
-
- * /, main/rtp.c: Merged revisions 98958 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4
- lines Add two more SDP names for ulaw and alaw. (closes issue
- #11777) Reported by: tootai ........
-
- * /, channels/chan_sip.c: Merged revisions 98955 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6
- lines Don't drop the old record route information when dealing
- with packets related to a reinvite. (closes issue #11545)
- Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by
- kebl0155 (license 356) ........
-
- * channels/chan_sip.c: Remove DNS lookup from sip_devicestate. This
- seems to come from way back when and I can't think of a reason
- for it being here, plus it could cause needless DNS lookups.
- (closes issue #10983) Reported by: jtodd
-
-2008-01-16 01:35 +0000 [r98953] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Terry found
- this problem with running the expr2 parser on OSX. Make the
- #defines come out the same between the parser & lexer.
-
-2008-01-16 01:17 +0000 [r98952] Joshua Colp <jcolp@digium.com>
-
- * /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
- configure.ac, makeopts.in: Merged revisions 98951 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan
- 2008) | 4 lines Add autoconf logic for speexdsp. Later versions
- use a separate library for some things so we need to use it if
- present in codec_speex. (closes issue #11693) Reported by: yzg
- ........
-
-2008-01-15 23:53 +0000 [r98948] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 98946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) |
- 11 lines Change a buffer in check_auth() to be a thread local
- dynamically allocated buffer, instead of a massive buffer on the
- stack. This fixes a crash reported by Qwell due to running out of
- stack space when building with LOW_MEMORY defined. On a very
- related note, the usage of BUFSIZ in various places in chan_sip
- is arbitrary and careless. BUFSIZ is a system specific define. On
- my machine, it is 8192, but by definition (according to google)
- could be as small as 256. So, this buffer in check_auth was 16
- kB. We don't even support SIP messages larger than 4 kB! Further
- usage of this define should be avoided, unless it is used in the
- proper context. ........
-
-2008-01-15 23:52 +0000 [r98947] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
- Add the "filter" keyword
-
-2008-01-15 23:35 +0000 [r98944-98945] Russell Bryant <russell@digium.com>
-
- * main/translate.c, include/asterisk/translate.h: Clean up
- something I did for ABI compatability in 1.4
-
- * main/frame.c, /, main/translate.c, main/abstract_jb.c,
- channels/chan_iax2.c, codecs/codec_zap.c,
- include/asterisk/frame.h, main/rtp.c,
- include/asterisk/translate.h: Merged revisions 98943 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15
- Jan 2008) | 25 lines Commit a fix for some memory access errors
- pointed out by the valgrind2.txt output on issue #11698. The
- issue here is that it is possible for an instance of a translator
- to get destroyed while the frame allocated as a part of the
- translator is still being processed. Specifically, this is
- possible anywhere between a call to ast_read() and
- ast_frame_free(), which is _a lot_ of places in the code. The
- reason this happens is that the channel might get masqueraded
- during this time. During a masquerade, existing translation paths
- get destroyed. So, this patch fixes the issue in an API and ABI
- compatible way. (This one is for you, paravoid!) It changes an
- int in ast_frame to be used as flag bits. The 1 bit is still used
- to indicate that the frame contains timing information. Also, a
- second flag has been added to indicate that the frame came from a
- translator. When a frame with this flag gets released and has
- this flag, a function is called in translate.c to let it know
- that this frame is doing being processed. At this point, the flag
- gets cleared. Also, if the translator was requested to be
- destroyed while its internal frame still had this flag set, its
- destruction has been deffered until it finds out that the frame
- is no longer being processed. Admittedly, this feels like a hack.
- But, it does fix the issue, and I was not able to think of a
- better solution ... ........
-
-2008-01-15 20:10 +0000 [r98895-98935] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 98934 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4
- lines Based on the boundary found move over the correct amount.
- (closes issue #11750) Reported by: tasker ........
-
- * /, channels/chan_sip.c: Merged revisions 98894 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4
- lines Accept "; boundary=" not just ";boundary=" in the multipart
- mixed content type. (closes issue #11750) Reported by: tasker
- ........
-
-2008-01-14 22:19 +0000 [r98889] Jason Parker <jparker@digium.com>
-
- * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
- backupdeleted option to app_voicemail (closes issue #10740)
- Reported by: ruffle Patches: app_voicemail.diff uploaded by
- ruffle (license 201) 10740-voicemail.diff uploaded by qwell
- (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak
- (license 7) Tested by: blitzrage, mvanbaak, qwell
-
-2008-01-14 22:11 +0000 [r98850-98888] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_directory.c: Big improvement for app_directory. This
- patch breaks the do_directory function up so that it is more
- easily parsed by the human brain. It also fixes some errors. I'll
- quote dimas from the original bug description: "app_directory
- contained some duplicate code even before addition of 'm' option.
- Addition of that option doubled amount of that code. Worst of
- all, there are minor differences between these code block and
- bugs caused by these differences. 1. There is a memory leak. In
- the 'menu' mode, result of the convert(pos) function is not freed
- while it should be. 2. In the 'menu' mode check for
- OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result,
- application works in the mode opposite to what user expect
- (checking last name when user wants the first nd vice versa). 3.
- select_item function plays message for user using res = func1()
- || func2() || func3()... construct. This construct loses the
- actual value returned by ast_waitstream() for example so at the
- end, res does not contain digit user dialed while listening to
- the message. 4. (also in 1.4) application announces entries from
- voicemail.conf/realtime separately from entries from users.conf.
- I see no reason why doing so instead of building combined list.
- 5. Alot of duplicated code as already mentioned." This was tested
- by dimas and I (I tested under valgrind). A word of caution: any
- bug fixes that happen in app_directory in 1.4 will almost
- certainly not merge cleanly into trunk as a result of this, but
- it is well worth it. Huge thanks to dimas for this wonderful
- submission. (closes issue #11744) Reported by: dimas Patches:
- dir3.patch uploaded by dimas (license 88) Tested by: putnopvut,
- dimas
-
-2008-01-14 20:01 +0000 [r98830] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Make sure the user's manager secret exists, even
- if it is blank. (closes issue #11749) Reported by: srt
-
-2008-01-14 18:42 +0000 [r98811] Terry Wilson <twilson@digium.com>
-
- * CHANGES: Add description of TOUPPER and TOLOWER dialplan
- functions to CHANGES.
-
-2008-01-14 17:40 +0000 [r98776] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Add proper call forwarding (all and busy)
- support for chan_skinny. Note: NoAnswer support is currently not
- implemented, as it would take a significant amount of work to
- figure out how to do correctly. Closes issue #11310, patches,
- testing, and support by DEA, mvanbaak, and myself.
-
-2008-01-14 17:39 +0000 [r98775] Russell Bryant <russell@digium.com>
-
- * /, main/translate.c: Merged revisions 98774 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) |
- 3 lines Revert a change that introduces an unacceptable
- performance hit and is causing memory leaks ... (from rev 97973)
- ........
-
-2008-01-14 17:18 +0000 [r98773] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix for potential crash with vmexten
-
-2008-01-14 16:36 +0000 [r98735-98738] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Merged revisions 98737 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan
- 2008) | 3 lines Fixing another compilation error. I'm a bit off
- today :( ........
-
- * /, apps/app_queue.c: Merged revisions 98733 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan
- 2008) | 8 lines Adding explicit defaults for missing options to
- init_queue. This is necessary because if a user either removes or
- comments one of these options and reloads their queues, the
- option will not reset to its default, instead maintaining the
- value from prior to the reload. Thanks to John Bigelow for
- pointing this error out to me. ........
-
-2008-01-14 15:07 +0000 [r98695-98714] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: Print out a warning when spaces are used in the
- variable name in Set and MSet. It is extremely hard to debug this
- issue so this should make it easier. (closes issue #11759)
- Reported by: caio1982 Patches: setvar_space_warning1.diff
- uploaded by caio1982 (license 22)
-
- * apps/app_meetme.c, doc/tex/qos.tex, doc/tex/realtime.tex: Update
- documentation. (closes issue #11763) Reported by: IgorG Patches:
- docupd.v1.diff uploaded by IgorG (license 20)
-
-2008-01-14 04:53 +0000 [r98558-98676] Russell Bryant <russell@digium.com>
-
- * apps/app_jack.c: Add another small option for the JACK app and
- JACK_HOOK function. The 'n' option tells JACK not to start jackd
- automatically if it is not already running. Otherwise, the
- default is that jackd will get started for you if it isn't
- running already.
-
- * CHANGES: - Break up the Misc. section a bit with a new section
- for Misc. New Modules - Change spacing a bit in some places for
- consistent indentation
-
- * CHANGES, apps/app_jack.c (added): Bring in the code from
- team/russell/jack/. Add a new module, app_jack, which provides
- interfaces to JACK, the Jack Audio Connection Kit
- (http://www.jackaudio.org/). Two interfaces are provided; there
- is a JACK() application, and a JACK_HOOK() function. Both
- interfaces create an input and output JACK port. The application
- makes these ports the endpoint of the call. The audio coming from
- the channel goes out the output port and whatever comes back in
- on the input port is what gets sent to the channel. The
- JACK_HOOK() function turns on a JACK audiohook on the channel.
- This lets you run the audio coming from a channel through JACK,
- and whatever comes back in is what gets forwarded on as the
- channel's audio. This is very useful for building custom vocoders
- or doing recording or analysis of the channel's audio in another
- application. In case anyone is curious, the platform that
- inspired me to write this is PureData (http://puredata.info/). I
- wrote these JACK interfaces so that I could use Pd to do
- interesting things with the audio of phone calls ...
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
- configure script check for JACK.
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
- Remove KDE configure script check that isn't used
-
- * main/audiohook.c: Remove a duplicate lock of the audiohook lock
- when destroying manipulate audiohooks. This causes an error when
- we attempt to destroy the lock later when freeing the audiohook.
-
- * main/pbx.c, CHANGES: Add a new CLI command, "core set chanvar",
- which allows you to set a channel variable (or function) on an
- active channel from the CLI.
-
-2008-01-12 18:12 +0000 [r98536] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c: Conversion to load manager.conf into memory did
- not convert the password functions correctly. (Closes issue
- #11749)
-
-2008-01-12 05:13 +0000 [r98514] Pari Nannapaneni <paripurnachand@digium.com>
-
- * /, main/http.c: merging a comment added in 1.4
-
-2008-01-12 00:20 +0000 [r98488] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, CHANGES: Add 'zap set dnd' CLI command, and
- ensure that the AMI DNDState event always gets generated. (closes
- issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded
- by tzafrir (modified by me) (license 46)
-
-2008-01-12 00:17 +0000 [r98487] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_odbc.c: Merged revisions 98467 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008)
- | 4 lines Add a connection timeout attribute, as that was what
- was intended with the login timeout, but ODBC divides it up into
- 2 different timeouts. (Closes issue #11745) ........
-
-2008-01-11 23:57 +0000 [r98454] Russell Bryant <russell@digium.com>
-
- * configure, doc/tex/Makefile, configure.ac, makeopts.in: Add some
- extra checking to help out with a potential error when trying to
- run "make asterisk.pdf" when not all of the right packages are
- installed. (closes issue #10763) Reported by: Corydon76 Patches:
- 20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76
-
-2008-01-11 23:10 +0000 [r98436] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add
- 'auto' signalling mode for Zaptel channels. (closes issue #11690)
- Reported by: tzafrir Patches: signaling_to_signalling.diff
- uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded
- by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir
- (license 46) zap_no_default_sig.diff uploaded by tzafrir (license
- 46) zap_signal_auto.diff uploaded by tzafrir (license 46)
-
-2008-01-11 23:09 +0000 [r98424-98435] Joshua Colp <jcolp@digium.com>
-
- * main/event.c: Goodbye again drumkilla.
-
- * main/event.c: drumkilla ftw.
-
- * main/audiohook.c: I am no longer Rockin'
-
- * main/audiohook.c: Testing something...
-
-2008-01-11 22:52 +0000 [r98400] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 98390 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) |
- 9 lines Fix up setting the EID on BSD based systems. (closes
- issue #11646) Reported by: caio1982 Patches:
- dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
- dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
- by: caio1982, mvanbaak ........
-
-2008-01-11 19:53 +0000 [r98318-98334] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 98325 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6
- lines If the incoming RTP stream changes codec force the bridge
- to break if the other side does not support it. (closes issue
- #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch
- uploaded by tsearle (license 373) ........
-
- * /, res/res_agi.c: Merged revisions 98317 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6
- lines If the channel is hungup during RECORD FILE send a result
- code of -1 to be uniform with everything else. (closes issue
- #11743) Reported by: davevg Patches: res_agi.diff uploaded by
- davevg (license 209) ........
-
-2008-01-11 19:12 +0000 [r98316] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 98315 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan
- 2008) | 5 lines Properly report the hangup cause as no answer
- when someone does not answer (closes issue #10574, reported by
- boch, patched by moy) ........
-
-2008-01-11 19:05 +0000 [r98270-98308] Russell Bryant <russell@digium.com>
-
- * codecs/codec_resample.c: Kevin noted that the thing that I
- _actually_ changed here was that I converted a value from a
- double, to a float, back to a double. Sure enough, when I changed
- my interim variable back to a double, it still blows up.
- Switching all of these to a float fixes the problem. This seems
- like a compiler bug where a double passed as an argument isn't
- getting properly aligned, so I'll have to see if I can replicate
- it with a small test program. (related to issue #11725)
-
- * codecs/codec_resample.c: Fix a bus error that happened when
- asterisk was built with optimizations on with platforms that
- explode on unaligned access. I'm not exactly sure why this fixes
- it, but it fixed it on the machine I was testing on. If it makes
- sense to you, feel free to enlighten me. :) (closes issue #11725,
- patched by me)
-
-2008-01-11 18:35 +0000 [r98268-98269] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Port Nick Gorham's timestamp patch to
- adaptive_odbc, too
-
- * cdr/cdr_odbc.c: Commit Nick Gorham's suggestion for timestamp fix
-
-2008-01-11 17:27 +0000 [r98220] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_followme.c: Merged revisions 98219 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4
- lines Ensure the return value of ast_bridge_call is passed back
- up as the application return value. This is needed for transfers
- to function so the PBX core knows to continue execution. (closes
- issue #10327) Reported by: kkiely ........
-
-2008-01-11 17:17 +0000 [r98218] Russell Bryant <russell@digium.com>
-
- * codecs/codec_g722.c: At one point during working on this module,
- I had the lin/lin16 versions of the framein callbacks different.
- However, they are now the same again, so remove the duplicate
- code and use the same functions for the lin/lin16 versions.
-
-2008-01-11 16:08 +0000 [r98152-98193] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 98164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008)
- | 2 lines Back out changes from revision 97077, since it wasn't
- perfect ........
-
- * doc/manager_1_1.txt: Documentation updates
-
-2008-01-11 12:51 +0000 [r98124] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: Ascom phones send Flash events as SIP INFO
- using '!' as the 'digit'
-
-2008-01-11 03:40 +0000 [r98081-98083] Russell Bryant <russell@digium.com>
-
- * codecs/codec_g722.c, main/frame.c: - Fix the last set of places
- where incorrect assumptions were made about the sample length
- with g722. It is _2_ samples per byte, not 1. This was all over
- the place, and I believed it, and it is what caused me to take so
- long to figure out what was broken. - Update copyright
- information on codec_g722.
-
-2008-01-11 00:54 +0000 [r98047] Mark Michelson <mmichelson@digium.com>
-
- * main/translate.c: Fix "core show translation" to not output
- information for "unknown" codecs. This fix was made in favor of
- the proposed patch since it doesn't involve changing a core codec
- define. (closes issue #11722, reported and initially patched by
- caio1982, final patch by me)
-
-2008-01-11 00:38 +0000 [r98024-98027] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add a new
- global and per-peer option to chan_sip, qualifyfreq, which allows
- you to set the qualify frequency. (closes issue #11597) Reported
- by: wilder Patches: qualifyfreq5.patch uploaded by wilder
- (license 362) -- with some mods by me
-
- * main/translate.c: Simplify this code with a suggestion from Luigi
- on the asterisk-dev list. Instead of using is16kHz(), implement a
- format_rate() function.
-
-2008-01-10 23:40 +0000 [r97978] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, main/translate.c: Merged revisions 97973
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008)
- | 6 lines 1) When we get a translated frame out, clone it,
- because if the translator pvt is freed before we use the frame,
- bad things happen. 2) Getting a failure from ast_sched_delete
- means that the schedule ID is currently running. Don't just
- ignore it. (Closes issue #11698) ........
-
-2008-01-10 23:33 +0000 [r97974-97977] Russell Bryant <russell@digium.com>
-
- * /, main/translate.c: Merged revisions 97976 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) |
- 3 lines Fix various timing calculations that made assumptions
- that the audio being processed was at a sample rate of 8 kHz.
- ........
-
- * codecs/codec_g722.c: Fix various issues in codec_g722. - The most
- common fix being made here is to fix all of the places where the
- number of output samples and output bytes gets updated in the
- translator state structure. - Fix a number of other places where
- the number of samples provided as an initialization value to a
- struct was incorrect.
-
- * codecs/codec_resample.c: Fix the buffer_samples value. For signed
- linear, the number of samples needed to fill the buffer is half
- the buffer size.
-
-2008-01-10 21:58 +0000 [r97933] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 97925 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan
- 2008) | 6 lines Let us leave a voicemail for ourself if we have
- logged into VoiceMailMain and chosen to leave a message. (closes
- issue #11735, reported and patched by jamessan) ........
-
-2008-01-10 21:46 +0000 [r97850-97890] Steve Murphy <murf@digium.com>
-
- * /, res/ael/ael_lex.c, res/Makefile, res/ael/ael.flex: Merged
- revisions 97889 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1
- line Applied the same fixes for ael.flex as was done in 97849 for
- ast_expr2.fl; overrode the normally generate yyfree func with our
- own version that checks the pointer for non-null before passing
- to free(). Also takes care of a little problem with 2.5.33 and
- the use of the __STDC_VERSION__ macro. ........
-
- * /, main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Merged
- revisions 97849 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1
- line This is a fix for 2 things: a problem Terry was having in
- OSX with null pointers, which was my fault, as I probably forgot
- to run the sed script last time I made mods. So, I moved the fix
- into the flex input itself. Then, I found when I used flex
- 2.5.33, that it was using __STDC_VERSION__, and that's not real
- good; so I added back in a DIFFERENT sed script to fix that
- little mess. Tested everything, a couple different ways. Hope I
- did no harm, at the least. ........
-
-2008-01-10 20:13 +0000 [r97848] Jason Parker <jparker@digium.com>
-
- * /, include/asterisk/frame.h: Merged revisions 97847 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan
- 2008) | 1 line Fix a comment that is no longer true. ........
-
-2008-01-10 20:05 +0000 [r97846] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Use the appropriate line ending for the
- X-Asterisk-VM-Message-Type header. (closes issue #11734, reported
- and patched by jaroth)
-
-2008-01-10 19:07 +0000 [r97825-97826] Terry Wilson <twilson@digium.com>
-
- * main/ast_expr2f.c: heh, remove patch to generated file.
-
- * main/ast_expr2f.c, main/cli.c: Check pointers before freeing (was
- getting WARNINGS under MALLOC_DEBUG)
-
-2008-01-10 17:38 +0000 [r97805] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_odbc.c: Fix problem with timestr going out of scope
- (Closes issue #11726, closes issue #11731)
-
-2008-01-10 17:30 +0000 [r97745-97804] Russell Bryant <russell@digium.com>
-
- * formats/format_sln16.c: minor formatting changes
-
- * main/translate.c: spaces to tabs
-
- * configure, configure.ac: Use AST_EXT_TOOL_CHECK() for the GTK
- check again. I changed this to an inline implementation to fix a
- small bug, but after a discussion with rizzo, I went to change it
- back. Also, it turns out that the implementation of the macro
- already supported what was needed to fix the problem.
-
- * pbx/pbx_kdeconsole.h (removed), /, configs/modules.conf.sample,
- pbx/kdeconsole_main.cc (removed): Merged revisions 97753 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) |
- 2 lines Remove other remnants of pbx_kdeconsole ........
-
- * /, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
- pbx/pbx_kdeconsole.cc (removed): Merged revisions 97734 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) |
- 4 lines Remove pbx_kdeconsole from the tree. It hasn't worked in
- ages, and nobody has complained. (closes issue #11706, reported
- by caio1982) ........
-
-2008-01-10 15:12 +0000 [r97698] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_groupcount.c, /: Merged revisions 97697 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r97697 | file | 2008-01-10 11:07:12 -0400 (Thu, 10 Jan
- 2008) | 6 lines Don't try to copy the category from the group if
- no category exists. (closes issue #11724) Reported by: IgorG
- Patches: group_count.v1.patch uploaded by IgorG (license 20)
- ........
-
-2008-01-10 00:54 +0000 [r97657] Russell Bryant <russell@digium.com>
-
- * include/asterisk.h: These prototypes are not supposed to be in
- asterisk.h. They are already in version.h.
-
-2008-01-10 00:50 +0000 [r97656] Steve Murphy <murf@digium.com>
-
- * include/asterisk.h, channels/console_video.c, utils/astman.c,
- channels/console_board.c, channels/vgrabbers.c: The fixes in this
- commit are mainly to allow compiling of trunk with
- --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly,
- the version.c needs to be in the OBJS line; asterisk.h was chosen
- to have the prototypes for ast_get_version, ast_get_version_num;
- and the ASTERISK_FILE_VERSION macro needs to be used after
- including asterisk.h in a few files. I hope I did the right
- thing. If not, let me know.
-
-2008-01-10 00:39 +0000 [r97655] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c: oops, missed the case of a 0 permission (which
- should mean everybody is allowed, not nobody)
-
-2008-01-10 00:22 +0000 [r97653] Terry Wilson <twilson@digium.com>
-
- * res/res_phoneprov.c: Attempt at making lookup_iface work under
- FreeBSD. Not yet tested, but it compiles under OS X. And still
- works under linux.
-
-2008-01-10 00:17 +0000 [r97652] Russell Bryant <russell@digium.com>
-
- * codecs/Makefile: Fix this so it doesn't force codec_g722 to get
- relinked every time
-
-2008-01-10 00:12 +0000 [r97651] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, main/manager.c, channels/chan_sip.c,
- res/res_features.c, pbx/pbx_realtime.c,
- configs/manager.conf.sample, CHANGES, channels/chan_iax2.c,
- include/asterisk/manager.h, apps/app_stack.c, main/db.c,
- apps/app_voicemail.c: Several manager changes: 1) Add the
- Dialplan class, for NewExten and VarSet events, which should cut
- down on the volume of traffic in the Call class. 2) Permit some
- commands to be run from multiple classes, such as allowing DBGet
- to be run from either the System or the Reporting class. 3)
- Heavily document each class in the sample config, as there were
- several that made no sense to be in the write= line, and two that
- made no sense to be in the read= line (since they controlled no
- permissions there). (Closes issue #10386)
-
-2008-01-10 00:11 +0000 [r97641-97650] Russell Bryant <russell@digium.com>
-
- * codecs/Makefile: Ensure that libg722.a gets rebuilt if one of the
- files changes
-
- * /, pbx/pbx_gtkconsole.c: Merged revisions 97645 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) |
- 2 lines Strip terminal sequences from the verbose messages
- ........
-
- * configure: re-gen configure
-
- * configure.ac: re-add check for gtk1, which is used for
- pbx_gtkconsole (related to issue #11706)
-
- * /, pbx/pbx_gtkconsole.c: Merged revisions 97640 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97640 | russell | 2008-01-09 16:26:33 -0600 (Wed, 09 Jan 2008) |
- 3 lines Make pbx_gtkconsole build ... but doesn't actually load
- on my system still (related to issue #11706) ........
-
-2008-01-09 21:37 +0000 [r97634] Terry Wilson <twilson@digium.com>
-
- * phoneprov/000000000000.cfg, phoneprov/000000000000-directory.xml,
- phoneprov/polycom.xml, res/res_phoneprov.c (added),
- funcs/func_strings.c, phoneprov/000000000000-phone.cfg,
- configs/modules.conf.sample, main/acl.c,
- include/asterisk/localtime.h, CHANGES,
- configs/phoneprov.conf.sample (added), Makefile, phoneprov
- (added), doc/tex/phoneprov.tex (added), main/stdtime/localtime.c,
- doc/tex/asterisk.tex: Added a new module, res_phoneprov, which
- allows auto-provisioning of phones based on configuration
- templates that use Asterisk dialplan function and variable
- substitution. It should be possible to create phone profiles and
- templates that work for the majority of phones provisioned over
- http. It is currently only intended to provision a single user
- account per phone. An example profile and set of templates for
- Polycom phones is provided. NOTE: Polycom firmware is not
- included, but should be placed in AST_DATA_DIR/phoneprov/configs
- to match up with the included templates.
-
-2008-01-09 20:30 +0000 [r97620-97623] Jason Parker <jparker@digium.com>
-
- * /, main/cli.c: Merged revisions 97622 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11718) ........ r97622 | qwell | 2008-01-09 14:28:43 -0600
- (Wed, 09 Jan 2008) | 5 lines Correctly display a message if a
- command could not be found. Also fix a comment which may have led
- to this happening. Issue 11718, reported by kshumard. ........
-
- * /, main/cli.c: Merged revisions 97618 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1
- line Fix some locking and return value funkiness. We really
- shouldn't be unlocking this lock inside of a function, unless we
- locked it there too. ........
-
-2008-01-09 18:53 +0000 [r97577] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 97575 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97575 | mmichelson | 2008-01-09 12:48:15 -0600 (Wed, 09 Jan
- 2008) | 3 lines Part 2 of app_queue doxygen improvements. Some
- smaller functions this time ........
-
-2008-01-09 18:12 +0000 [r97532-97533] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c: remove a wrong 'const'
-
- * images/kpad2.jpg: add annotations for the two message windows we
- use.
-
-2008-01-09 18:04 +0000 [r97531] Russell Bryant <russell@digium.com>
-
- * /, res/res_features.c: Merged revisions 97529 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97529 | russell | 2008-01-09 12:02:08 -0600 (Wed, 09 Jan 2008) |
- 2 lines Fix saying the parking space number to the caller doing
- the parking ... ........
-
-2008-01-09 18:03 +0000 [r97530] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/console_board.c,
- channels/console_video.h: Two changes: - support scrolling of
- message window; - simplify the code for creating a message
- window, and try it using a second one in the top of the keypad
- (where we echo the dialed number). The 'skin' that supports these
- two windows will be committed separately.
-
-2008-01-09 17:30 +0000 [r97495] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 97491 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008)
- | 2 lines report the same message whether Zaptel does not have
- transcoder support loaded or no transcoders were found ........
-
-2008-01-09 16:59 +0000 [r97490] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 97489 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09
- Jan 2008) | 7 lines Set the caller id within the gtalk_alloc
- function. As underlined in issue #10437 by Josh, we need to
- prevent a possible memory leak. We only set the name part of the
- caller id, the number part is not relevant when dealing with
- JIDs. Closes issue #11549. ........
-
-2008-01-09 16:44 +0000 [r97488] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/console_video.c,
- channels/console_board.c, channels/console_video.h: Implement
- keyboard handling, and use it to enter a number to dial in the
- 'message' area under the keypad. Now you can make calls using the
- keypad as a regular phone (or the keyboard for chars not present
- on the keypad)
-
-2008-01-09 16:13 +0000 [r97451] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 97450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97450 | file | 2008-01-09 12:11:17 -0400 (Wed, 09 Jan 2008) | 6
- lines Don't do conferencing totally in Zaptel if Monitor is
- running on the channel. (closes issue #11709) Reported by:
- BigJimmy Patches: patch-meetmerec uploaded by BigJimmy (license
- 371) ........
-
-2008-01-09 15:45 +0000 [r97421-97449] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 97448 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97448 | kpfleming | 2008-01-09 09:43:19 -0600 (Wed, 09 Jan 2008)
- | 2 lines pass the right variable to get an error string... oops
- ........
-
- * channels/chan_zap.c, /: Merged revisions 97410 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97410 | kpfleming | 2008-01-09 09:26:23 -0600 (Wed, 09 Jan 2008)
- | 2 lines add error number output to ioctl failure messages to
- help with debugging ........
-
-2008-01-09 12:23 +0000 [r97389-97390] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_video.c, channels/console_video.h: implement the
- "console startgui" and "console stopgui" commands so you can
- start and stop the gui even outside of a call. This is convenient
- for testing, and also for using the keypad to pick up a call, and
- to dial a number (the latter not yet implemented, but should be
- close).
-
- * channels/chan_oss.c: make get_video_desc() return the active
- console if passed a null argument (channel).
-
-2008-01-09 00:58 +0000 [r97364-97365] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: New option in trunk, needs strdupa to be safe,
- too
-
- * /, main/editline/readline.c, main/cli.c: Merged revisions 97350
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008)
- | 5 lines Allow filename completion on zero-length modules,
- remove a memory leak, remove a file descriptor leak, and make
- filename completion thread-safe. Patched and tested by tilghman.
- (Closes issue #11681) ........
-
-2008-01-09 00:18 +0000 [r97307-97309] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 97308 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97308 | mmichelson | 2008-01-08 18:17:40 -0600 (Tue, 08 Jan
- 2008) | 3 lines use the \retval doxygen command properly ........
-
- * /, apps/app_queue.c: Merged revisions 97304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97304 | mmichelson | 2008-01-08 17:49:11 -0600 (Tue, 08 Jan
- 2008) | 5 lines Part 1 of N of adding doxygen comments to
- app_queue. I picked some of the most common functions used (which
- also happen to be some the biggest/ugliest functions too) to
- document first. I'm pretty new to doxygen so criticism is
- welcome. ........
-
-2008-01-08 23:51 +0000 [r97305] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Add a new flag 'd' (with optional context)
- permitting any extension within that context to be entered as a
- new extension during the playback of a voicemail greeting. Patch
- inspired by bluecrow76, by tilghman. (Closes issue #7063)
-
-2008-01-08 23:35 +0000 [r97280-97303] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_board.c: add copyright (most of this code was
- written by Marta Carbone), remove some unused code, add/clarify
- some comments.
-
- * images/kpad2.jpg: Add the annotation for the textarea used for
- messages, and also change the background from white to something
- different to show that we can make use of fonts with transparent
- background.
-
- * images/font.png (added): add a font suitable for use with the
- console GUI. The background of this particular image is
- transparent so we can preserve the original background when we
- draw strings.
-
- * channels/console_gui.c, channels/console_video.c,
- channels/console_board.c (added), channels/Makefile: add support
- for textareas, used for various dialog windows on the gui. The
- main code to implement the textarea is in console_board.c, and
- uses a simple png image with the font, blitting characters on the
- designated areas of the main screen. Additionally we provide some
- annotations in the image used as a skin to indicate which areas
- are used for text messages. (images will be committed
- separately). At the moment the dialog area is only used to
- display a running counter, just as a proof of concept.
-
-2008-01-08 21:56 +0000 [r97248] Terry Wilson <twilson@digium.com>
-
- * apps/app_queue.c: Initialize new variable to NULL
-
-2008-01-08 21:28 +0000 [r97203-97208] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the
- option of specifying a second interface in a member definition
- for a queue. app_queue will monitor this second device's state
- for the member, even though it actually calls the first
- interface. This ability has been added for statically defined
- queue members, realtime queue members, and dynamic queue members
- added through the CLI, dialplan, or manager. (closes issue
- #11603, reported by acidv)
-
-2008-01-08 21:01 +0000 [r97199-97200] Olle Johansson <oej@edvina.net>
-
- * channels/chan_console.c: Change reference to external library so
- it appears on the extref listing
- http://www.asterisk.org/doxygen/trunk/extref.html
-
- * res/res_jabber.c: Iksemel is alive in a new home. Release 1.3 is
- out with bug fixes.
-
-2008-01-08 20:56 +0000 [r97198] Tilghman Lesher <tlesher@digium.com>
-
- * main/autoservice.c, /, main/utils.c: Merged revisions 97194 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008)
- | 3 lines Increase constants to where we're less likely to hit
- them while debugging. (Closes issue #11694) ........
-
-2008-01-08 20:52 +0000 [r97196-97197] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: One line documentation ftw!
-
- * /, channels/chan_mgcp.c: Merged revisions 97195 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6
- lines Fix various DTMF issues in chan_mgcp. (closes issue #11443)
- Reported by: eferro Patches:
- dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
- 337) ........
-
-2008-01-08 20:45 +0000 [r97193] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 97192 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan
- 2008) | 9 lines Making some changes designed to not allow for a
- corrupted mailstream for a vm_state. 1. Add locking to the
- vm_state retrieval functions so that no linked list corruption
- occurs. 2. Make sure to always grab the persistent vm_state when
- mailstream access is necessary. 3. Correct an incorrect return
- value in the init_mailstream function. (closes issue #11304,
- reported by dwhite) ........
-
-2008-01-08 20:06 +0000 [r97153-97154] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Move common code for setting T38
- capabilities and fix a bug with fax detection in the SIP RTP read
- callback. It's still sort of silly... but more on that later.
- (closes issue #11239) Reported by: dimas Patches:
- sipt38prop.patch uploaded by dimas (license 88)
-
- * funcs/func_groupcount.c, /: Merged revisions 97152 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r97152 | file | 2008-01-08 15:53:52 -0400 (Tue, 08 Jan
- 2008) | 4 lines If no group has been provided to the GROUP_COUNT
- dialplan function then use the first one specific to the channel.
- (closes issue #11077) Reported by: m4him ........
-
-2008-01-08 19:06 +0000 [r97125] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, main/asterisk.c: Merged revisions 97077
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008)
- | 3 lines Apply multiple crash fixes, found in issue #11386, but
- not completely closing that issue. ........
-
-2008-01-08 18:42 +0000 [r97041-97103] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 97093 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r97093 | file | 2008-01-08 14:36:40 -0400 (Tue, 08 Jan 2008) | 4
- lines Make app_queue calls work with directed pickup. (closes
- issue #11700) Reported by: jbauer ........
-
- * utils/extconf.c: Make ast_atomic_fetchadd_int_slow magically
- appear in extconf. (closes issue #11703) Reported by: dmartin
-
-2008-01-07 23:03 +0000 [r96988] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c: add support for cropping the keypad image
- while displaying it. This way it can contain additional elements
- (e.g. fonts, buttons, widgets) without having to use a zillion
- files to store them.
-
-2008-01-07 22:31 +0000 [r96987] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Explicitly make literal constants long
- where they are expected to be.
-
-2008-01-07 21:12 +0000 [r96936] Jason Parker <jparker@digium.com>
-
- * main/config.c: Display a message if no config mappings are found
- with "core show config mappings". Closes issue #11704, patch by
- kshumard.
-
-2008-01-07 21:10 +0000 [r96934-96935] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Document some weird casting magic that's
- necessary to interface with the c-client
-
- * doc/tex/imapstorage.tex, apps/app_voicemail.c: Adding
- user-configurable TCP timeout settings to IMAP voicemail. This
- could go a long way towards preventing unexplainable hangs
- experienced by people. In the case of MWI hangs, this also will
- mean that the SIP port isn't blocked anymore. (closes issue
- #11665, reported by yehavi)
-
-2008-01-07 20:48 +0000 [r96885-96933] Russell Bryant <russell@digium.com>
-
- * /, configs/extensions.conf.sample: Merged revisions 96932 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r96932 | russell | 2008-01-07 14:47:52 -0600
- (Mon, 07 Jan 2008) | 10 lines Merged revisions 96931 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07
- Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com
- ........ ................
-
- * configs/http.conf.sample: Add a note about viewing the default
- set of documentation using the built-in http server
-
- * Makefile: If the HTML documentation exists, install it in the
- static-http/docs directory so that it can be viewed through the
- Asterisk http server if it is turned on.
-
- * build_tools/prep_tarball: Build the HTML version of the doc files
- for tarballs, as well
-
- * res/res_smdi.c, /: Merged revisions 96884 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96884 | russell | 2008-01-07 10:39:23 -0600 (Mon, 07 Jan 2008) |
- 3 lines Don't crash if something happens when setting up an SMDI
- interface and it gets destroyed before the SMDI port handling
- thread gets created. ........
-
-2008-01-07 16:17 +0000 [r96862] Kevin P. Fleming <kpfleming@digium.com>
-
- * formats/format_sln16.c (added): add a file-format driver for
- 16KHz signed linear... which may or may not work
-
-2008-01-07 15:52 +0000 [r96858] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c, main/loader.c: Move ModuleLoad and ModuleCheck
- manager commands from loader.c to manager.c. Previously they
- would get registered twice because of the way manager.c operates.
- (closes issue #11699) Reported by: caio1982 Patches:
- manager_module_commands1.diff uploaded by caio1982 (license 22)
-
-2008-01-07 15:06 +0000 [r96776-96836] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c: update comments to reflect reality (or at
- least planned behaviour). minor code cleanups
-
- * channels/console_gui.c: resolve a load-time problem avoiding a
- call to console_do_answer. On passing, fix dialling from the
- keypad.
-
-2008-01-05 23:05 +0000 [r96645-96743] Russell Bryant <russell@digium.com>
-
- * res/snmp/agent.c: Convert this file over the new method of
- getting the Asterisk version. (I don't have this building on this
- machine, so caio1982 on IRC is going to test it for me. :) )
-
- * Makefile, funcs/func_version.c, main/manager.c,
- channels/chan_sip.c, main/Makefile, build_tools/make_version_c
- (added), include/asterisk/version.h (added), res/res_agi.c, main,
- main/http.c, build_tools/make_version_h (removed),
- include/asterisk, main/asterisk.c: Now that the version.h file
- was getting properly regenerated every time the svn revision
- changed, every module that used the version was getting rebuilt
- after every svn update. This severly annoyed me pretty quickly,
- so I have improved the situation. Now, instead of generating
- version.h, main/version.c is generated. version.c includes the
- version information, as well as a couple of API calls for modules
- to retrieve the version. So now, only version.c will get rebuilt,
- and the main asterisk binary relinked, which is must faster than
- rebuilding http.c, manager.c, asterisk.c, relinking the asterisk
- binary, chan_sip.c, func_version.c, res_agi ... The only minor
- change in behavior here is that the version information reported
- by chan_sip, for example, is the version of the Asterisk core,
- and not necessarily the Asterisk version that the chan_sip module
- came from.
-
- * main/pbx.c: Print out the name of a function being registered in
- color, just like the name of applications when they get
- registered.
-
- * UPGRADE.txt: Add a note about changing modules.conf since another
- console channel driver is now present that can not be used at the
- same time as chan_alsa or chan_oss.
-
- * channels/chan_console.c: Add the URL to the home page for
- portaudio. Also add the location of the svn repository to check
- out portaudio v19.
-
- * /, main/devicestate.c: Merged revisions 96644 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) |
- 2 lines Don't pass an empty string as the device name. ........
-
-2008-01-05 01:05 +0000 [r96621] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_usbradio.c: improve chan_usbradio to use
- indications just like chan_alsa/chan_oss do now
-
-2008-01-04 23:12 +0000 [r96576] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/devicestate.c: Merged revisions 96575 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008)
- | 7 lines Fix the problem of notification of a device state
- change to a device with a '-' in the name. Could probably do with
- a better fix in trunk, but this bug has been open way too long
- without a better solution. Reported by: stevedavies Patch by:
- tilghman (Closes issue #9668) ........
-
-2008-01-04 22:57 +0000 [r96574] Jason Parker <jparker@digium.com>
-
- * /, res/res_features.c: Merged revisions 96573 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #11237) ........ r96573 | qwell | 2008-01-04 16:55:56 -0600
- (Fri, 04 Jan 2008) | 4 lines Properly continue in the dialplan if
- using PARKINGEXTEN and the slot is full. Issue 11237, patch by
- me. ........
-
-2008-01-04 19:35 +0000 [r96547] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 96525 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008)
- | 4 lines If you change the bindaddr in sip.conf to a non-bound
- address and reload, sip goes kablooie. Reported and patched by:
- one47 (Closes issue #11535) ........
-
-2008-01-04 17:21 +0000 [r96500] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
- configure.ac, acinclude.m4: [commit message] (closes issue
- #10393) Reported by: tzafrir Patches: chan_alarm_asterisk.diff
- uploaded by tzafrir (license 46) (modified by me and added
- configure script support)
-
-2008-01-04 17:19 +0000 [r96499] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Use SASL DIGEST-MD5 authentication over
- unsecured network connections only. This authentication mechanism
- is implemented under the iksemel API, which makes use of GnuTLS,
- whereas we use OpenSSL. Note : there's ongoing dicsussion at the
- SASL IETF WG in order to deprecate SASL DIGEST-MD5, see
- http://ietfreport.isoc.org/ids-wg-sasl.html.
-
-2008-01-04 16:21 +0000 [r96450] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 96449 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) |
- 7 lines Make use of the temporary channel pointer while the pvt
- is unlocked. (closes issue #11675) Reported by: flefoll Patches:
- chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll
- (license 244) ........
-
-2008-01-03 23:14 +0000 [r96397-96398] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile: we have to *always* use a completely silent 'make'
- invocation for generating the module embedding rules
-
- * Makefile: there was no reason to add this define for non-Solaris
- platforms
-
-2008-01-03 22:46 +0000 [r96395] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 96394 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) |
- 3 lines Don't crash if the iax2 pvt structure has been destroyed
- before we get to this point (closes issue #11672, reported by
- snuffy, patched by me) ........
-
-2008-01-03 21:58 +0000 [r96301-96368] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/channel.h: Document recent API addition
-
- * res/res_config_pgsql.c, /: Merged revisions 96318 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r96318 | tilghman | 2008-01-03 15:37:02 -0600 (Thu, 03
- Jan 2008) | 4 lines Missed initialization caused crash. Reported
- and fixed by: tiziano (Closes issue #11671) ........
-
- * main/channel.c: Allow the uniqueid to be used for searching for a
- channel in the list. Reported and initially patched by:
- michael-fig (Closes issue #11340)
-
-2008-01-03 20:04 +0000 [r96245-96272] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, tests/Makefile (added), tests/test_skel.c (added),
- tests (added): add some simple infrastructure for modules to be
- used for testing parts of Asterisk
-
- * channels/answer.h (removed), channels/ring10.h (removed),
- channels/busy.h (removed), channels/ringtone.h (removed),
- channels/Makefile, channels/chan_oss.c, channels/gentone.c
- (removed), channels: eliminiate sound_thread() and other stuff
- from chan_oss since Asterisk indications can handle it remove
- gentone and all the headers containing tones that are no longer
- needed
-
- * channels/chan_alsa.c: coding guidelines cleanup remove background
- thread and all sound generation mechanisms, as the built-in
- indications can handle everything that is needed
-
-2008-01-03 14:47 +0000 [r96221] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 96198 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03
- Jan 2008) | 1 line when overlapdial was used and no number was
- dialed, the call was dropped, now we just jump into the s
- extension, which makes a lot more sense. ........
-
-2008-01-03 06:16 +0000 [r96147-96174] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Add coordination between AMI and AGI applications,
- with an asyncagi method Feature proposed and patched by: moy
- (Closes issue #11282)
-
- * apps/app_mp3.c, apps/app_ices.c, main/asterisk.c: Compatibility
- fix for OpenBSD Report and fix by: mvanbaak (Closes issue #11669)
-
-2008-01-02 23:48 +0000 [r96103] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 96102 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan
- 2008) | 4 lines We need to reset the membername to NULL on each
- iteration of this loop, otherwise the result is that multiple
- members can have the same name, since the variable was not reset
- on each iteration of the loop. ........
-
-2008-01-02 23:22 +0000 [r96076-96079] Russell Bryant <russell@digium.com>
-
- * channels/chan_console.c: Add support for generating a ringing
- sound on an incoming call. This is a bit of a hack. It just asks
- the core to generate the same tone that it would when you hear
- ringback when making an outbound call. But hey, it works, and you
- get the localized ring tone for the appropriate language set on
- the channel.
-
- * channels/chan_console.c: Note that this module doesn't actually
- play a ringing sound for an incoming call ... oops
-
- * channels/chan_console.c: Show the correct CLI command to answer
- the call
-
-2008-01-02 22:41 +0000 [r96073] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: actually parse and store echocan parameters
- from zapata.conf... this *should* work <G>
-
-2008-01-02 22:40 +0000 [r96071] Joshua Colp <jcolp@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: Don't
- use AST_C_DEFINE_CHECK for the two pthread things that may not
- actually be definitions, they could be enums for example.
-
-2008-01-02 22:29 +0000 [r96028] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c: Add curly braces around a compound if
- statement so that trunk will build properly
-
-2008-01-02 21:51 +0000 [r96019] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, configs/zapata.conf.sample: another
- checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS
- ioctl if it is present, but doesn't parse any supplied parameters
- yet (this implementation is not very memory efficient as the
- parameters and their values will be duplicated for each channel
- that has the same settings, but we can worry about that later
- once it is working)
-
-2008-01-02 21:49 +0000 [r96018] Russell Bryant <russell@digium.com>
-
- * main/libresample/include/libresample.h: Add doxygen documentation
- to libresample.h while it's still fresh on my mind
-
-2008-01-02 21:08 +0000 [r95994] Mark Michelson <mmichelson@digium.com>
-
- * funcs/func_odbc.c, channels/chan_agent.c, funcs/func_strings.c,
- apps/app_rpt.c: Change instances of AST_NONSTANDARD_APP_ARGS(foo,
- bar, ',') to AST_STANDARD_APP_ARGS(foo, bar) (closes issue
- #11668, reported and patched by mvanbaak)
-
-2008-01-02 20:26 +0000 [r95947] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 95946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4
- lines Allocate a SIP refer structure when performing a transfer
- using BYE with Also so that the transfer information is properly
- stored. (AST-2008-001) (closes issue #11637) Reported by:
- greyvoip ........
-
-2008-01-02 20:23 +0000 [r95944-95945] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Since ',' is the standard argument separator in
- trunk, change app_queue to use AST_STANDARD_APP_ARGS instead of
- AST_NONSTANDARD_APP_ARGS for determining member data.
-
- * include/asterisk/app.h: Fix a typo in a comment.
- AST_STANDARD_APP_ARGS uses ',' as the separator, not '|'.
-
-2008-01-02 19:47 +0000 [r95893-95939] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: clean up hwgain CLI command and improve docs
- for swgain CLI command
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4: improve AC_C_DEFINE_CHECK to not try to evaluate
- the macro being checked for, but just check for its existence
- finish implementation of check for Zaptel HWGAIN support add
- check for Zaptel ECHOCANCEL_PARAMS support
-
- * codecs/Makefile, include/asterisk/libresample.h (added),
- codecs/codec_resample.c: and now just to keep the libresample
- party going... if the functions from libresample are going to be
- in the main Asterisk binary, it makes sense for the header that
- defines them to be available without any special CFLAGS and to
- out-of-tree modules building against /usr/include/asterisk
-
- * channels/chan_zap.c: umm... this did not compile on x86-64, and
- could not possibly have worked on any platform as it was passing
- string pointers to a function expecting ints
-
-2008-01-02 18:05 +0000 [r95891] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 95890 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan
- 2008) | 9 lines A change to improve the accuracy of queue logging
- in the case where a member does not answer during the specified
- timeout period. Prior to this change, there was a small chance
- that the member name recorded in this case would be blank. Also
- prior to this change, if using the ringall strategy, if no one
- answered the call during the specified timeout, the member name
- listed in the queue log would randomly be one of the members that
- was rung. (closes issue #11498, reported and tested by hloubser,
- patched by me) ........
-
-2008-01-02 17:38 +0000 [r95888] Jason Parker <jparker@digium.com>
-
- * apps/app_osplookup.c: Update osplookup documentation to use
- commas instead of pipes. Closes issue #11666, patch by Laureano.
-
-2008-01-02 16:20 +0000 [r95864] Russell Bryant <russell@digium.com>
-
- * main/Makefile, main/translate.c: For some odd reason, the last
- set of libresample build changes from Kevin did not work for
- everyone, but it did for some. This set of changes makes trunk
- start again for those having problems. Instead of building
- libresample as a static library, it just links the object files
- in directly with the asterisk binary.
-
-2008-01-02 14:53 +0000 [r95816-95841] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/Makefile: fix some long-time breakage that kept
- chan_misdn from being embedded
-
- * channels/Makefile: use the proper technique for including
- submodules so that embedding will work
-
- * CHANGES: note that chan_console requires portaudio v19
-
- * configure, configure.ac: actually check for a function present in
- libiconv (don't know how this test could have worked before) and
- don't do the check on Linux/GNU systems because libiconv is not
- present there and attempting to link with '-liconv' always fails
- (it's not necessary as the iconv functionality is always
- available)
-
- * main/libresample/src/filterkit.h,
- main/libresample/src/resample.c,
- main/libresample/win/libresample.dsp, main/libresample/configure,
- main/libresample/Makefile.in, res/Makefile,
- main/libresample/configure.in, main/libresample/src,
- main/libresample/tests/testresample.c,
- main/libresample/win/libresample.vcproj,
- main/libresample/tests/compareresample.c, main/libresample/tests,
- codecs/codec_resample.c, res/res_resample.c (removed),
- main/libresample/README.txt, main/libresample/src/resamplesubs.c,
- main/libresample/tests/resample-sndfile.c,
- main/libresample/src/configtemplate.h,
- main/libresample/install-sh, main/Makefile, main/translate.c,
- main/libresample/include, main/libresample/src/resample_defs.h,
- codecs/Makefile, main/libresample/config.guess,
- main/libresample/config.sub, main/libresample/win,
- main/libresample/LICENSE.txt, main/libresample (added),
- main/libresample/Makefile.asterisk, build_tools/strip_nonapi,
- res/libresample (removed), main/libresample/src/filterkit.c,
- main/libresample/include/libresample.h: go back to including
- libresample in the main Asterisk binary, but this time including
- a small hack to ensure that it does get linked in (and also
- modify the strip_nonapi script to leave the resample_<foo>
- symbols alone)
-
-2008-01-02 11:34 +0000 [r95794] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Set stream flags to zero upon initialization.
- When the XMPP over TLS/SSL connection resets for some reason, it
- is wrongly believed as being secured, which makes the
- re-connection process endlessly fail. This was reported by
- mvanbaak in issue #11644.
-
-2008-01-02 09:16 +0000 [r95771-95772] Luigi Rizzo <rizzo@icir.org>
-
- * main/loader.c: some cleanup of this code while I am trying to
- debug a problem with gdb dying while debugging asterisk. The
- problem seems to be related with a race in the handling of
- module_list, which in turn is triggeded by calling dlopen() on a
- system which uses initializers to create locks.
-
- * include/asterisk/module.h: There are three instances of the
- module definition macros, which make maintaining this file very
- error prone. This commit merges the embedded and !embedded
- versions, and fixes the C++ version. Eventually we should move to
- a single version of the macro. Too bad C++ doesn't like the
- C-style struct initializers .foo = some_value
-
-2008-01-02 04:33 +0000 [r95697-95746] Russell Bryant <russell@digium.com>
-
- * res/libresample/src/resample_defs.h,
- res/libresample/src/resample.c: Don't make libresample print out
- debugging output
-
- * main/translate.c: Make the translation table show slin16
-
- * apps/app_meetme.c: fix a spacing issue introduced in revision
- 95443.
-
- * main/Makefile, res/libresample/README.txt, res/Makefile,
- res/libresample/install-sh, res/libresample/configure,
- res/libresample/Makefile.in, res/libresample/include,
- codecs/Makefile, res/libresample/configure.in,
- res/libresample/src, res/libresample/config.guess,
- main/libresample (removed), res/libresample/config.sub,
- res/libresample/win, codecs/codec_resample.c,
- res/libresample/LICENSE.txt, res/libresample (added),
- res/libresample/Makefile.asterisk, res/libresample/tests,
- res/res_resample.c (added): Instead of linking libresample into
- the main Asterisk binary, build it as res_resample, and mark
- codec_resample as dependent upon res_resample. This prevents the
- linker from optimizing away libresample, and also makes it so the
- libresample code isn't linked in to multiple places. (I have
- another module in a branch that needs it, too.)
-
-2008-01-01 23:55 +0000 [r95671-95673] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c: call directly the cli command to
- implement hangup.
-
- * channels/vcodecs.c: prevent a panic when destroying a channel
- with no incoming video.
-
- * channels/console_video.c: remove a leftover sleep(1) used for
- debugging
-
-2008-01-01 23:09 +0000 [r95648] Joshua Colp <jcolp@digium.com>
-
- * codecs/Makefile: Fix building of codec_resample on platforms
- other then Cygwin. On everything else it actually gets built
- after codec_resample, so you can't exactly link it in since it
- doesn't exist.
-
-2008-01-01 22:21 +0000 [r95624-95625] Luigi Rizzo <rizzo@icir.org>
-
- * codecs/Makefile, codecs/codec_resample.c: make codec_resample
- build on __CYGWIN__, and make it load on FreeBSD (and probably
- other systems as well). Both need libresample.a to be specified
- in the linking phase, and cygwin needs <float.h> as other BSD.
- The checks for OS-specific headers should really be moved to some
- common header though.
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- funcs/func_iconv.c, makeopts.in: implement "configure" checks for
- libiconv, and add the iconv dependency for func_iconv. This fixes
- some build issues on CYGWIN and FreeBSD and probably other
- platforms where libiconv is not there by default
-
-2007-12-31 23:44 +0000 [r95578] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, /: Merged revisions 95577 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec
- 2007) | 9 lines Avoiding a potentially bad locking situation.
- ast_merge_contexts_and_delete writelocks the conlock, then calls
- ast_hint_extension, which attempts to readlock the same lock.
- Recursion with read-write locks is dangerous, so the inner lock
- needs to be removed. I did this by copying the "guts" of
- ast_hint_extension into ast_merge_contexts_and_delete (sans the
- extra lock). (this change is inspired by the locking problems
- seen in issue #11080, but I have no idea if this is the
- problematic area experienced by the reporters of that issue)
- ........
-
-2007-12-31 22:41 +0000 [r95501-95550] Russell Bryant <russell@digium.com>
-
- * codecs/codec_resample.c: Use float.h to fix the build on FreeBSD.
- Also, add some other platforms as they are likely the same.
-
- * channels/chan_console.c: Update chan_console to natively use a 16
- kHz sample rate. If it is talking to an 8 kHz endpoint, then
- codec_resample will automatically be used to properly resample
- the audio before sending it to/from chan_console.
-
- * main/libresample/src/filterkit.h, main/libresample/README.txt,
- main/libresample/tests/resample-sndfile.c,
- main/libresample/src/resamplesubs.c, main/Makefile,
- main/libresample/install-sh,
- main/libresample/src/configtemplate.h,
- main/libresample/src/resample.c,
- main/libresample/win/libresample.dsp, main/libresample/configure,
- main/libresample/Makefile.in, main/libresample/include, CHANGES,
- main/libresample/src/resample_defs.h,
- main/libresample/configure.in, main/libresample/src,
- main/libresample/config.guess, codecs/Makefile,
- main/libresample/tests/testresample.c, codecs/slin_resample_ex.h
- (added), main/libresample/config.sub, main/libresample/win,
- main/libresample/win/libresample.vcproj,
- main/libresample/LICENSE.txt, main/libresample (added),
- main/libresample/Makefile.asterisk, main/libresample/tests,
- main/libresample/tests/compareresample.c, codecs/codec_resample.c
- (added), main/libresample/src/filterkit.c,
- main/libresample/include/libresample.h: Merge changes from
- team/russell/codec_resample This commit imports libresample for
- use in Asterisk. It also adds a new codec module, codec_resample.
- This module uses libresample to re-sample signed linear audio
- between 8 kHz and 16 kHz. It also provides an alternative for
- converting between 16 kHz G.722 and 8 kHz signed linear when
- using G.722, which will likely be useful as some people have
- complained about volume issues when the current codec_g722
- converts to 8 kHz signed linear. But, to test this, you will have
- to disable the g722-to-slin and g722-to-slin16 translators in
- codec_g722.c.
-
-2007-12-31 20:33 +0000 [r95490] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_env.c: Merged revisions 95470 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95470 | tilghman | 2007-12-31 14:27:26 -0600 (Mon, 31 Dec 2007)
- | 3 lines Allow the default "0" to be returned if the STAT fails
- (Closes issue #11659) ........
-
-2007-12-31 18:46 +0000 [r95443] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_meetme.c: Fix a compiler warning (closes issue #11658,
- reported and patched by eliel)
-
-2007-12-31 16:13 +0000 [r95383-95412] Russell Bryant <russell@digium.com>
-
- * configs/console.conf.sample (added), configs/modules.conf.sample,
- channels/chan_console.c (added), CHANGES: Merge the main set of
- changes from team/russell/chan_console. Add a new console channel
- driver, chan_console, which is a console channel driver that uses
- portaudio as a cross platform audio interface. It was written to
- provide a console channel driver that works with Mac CoreAudio,
- but it supports a number of other audio interfaces, as well,
- including OSS and ALSA. It could one day be the single console
- channel driver, but does not yet have as many features as
- chan_oss.
-
- * include/asterisk/channel.h: fix a spelling error in a comment
-
- * include/asterisk/config.h: Add CV_STRINGFIELD() macro. This lets
- you set a config variable to a string field. (from
- team/russell/chan_console)
-
- * configure, include/asterisk/autoconfig.h.in: Regenerate configure
- script to include check for portaudio.
-
- * build_tools/menuselect-deps.in, configure.ac, makeopts.in: Add
- configure script checking for portaudio.
-
-2007-12-29 02:02 +0000 [r95262-95313] Luigi Rizzo <rizzo@icir.org>
-
- * channels/vcodecs.c, channels/console_video.c, channels/Makefile,
- channels/console_video.h, channels/vgrabbers.c (added): Move
- grabbers definitions to a separate file, vgrabbers.c, so it is
- easier to add more entries. This required moving struct grab_desc
- to the common header, and adding an entry in the Makefile. On
- passing, cleanup some comments and file headers (some are still
- missing).
-
- * channels/console_gui.c, channels/console_video.c: virtualize the
- interface for video grabbers, which should make it easier to add
- support for more grabbers (V4L2, firewire, and so on).
-
- * channels/console_video.c: Add a few entries up to 1408x1152 in
- the table of known video resolutions. This makes it very
- convenient to enlarge images using the right-click on the video
- window.
-
- * channels/vcodecs.c, channels/console_video.c: change the
- interface of video encapsulation routines, they only need the
- buffer and mtu as input.
-
- * channels/console_gui.c, channels/vcodecs.c,
- channels/console_video.c, channels/console_video.h: various
- rearrangements and renaming of console_video stuff
-
-2007-12-28 18:39 +0000 [r95233] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: The diff for this change looks really bad, but
- all I did here was decrease the indentation of most of the
- queue_exec function by reversing the logic of an if statement.
- This change makes the function comply better with the coding
- guidelines. Since this change is purely a cosmetic change to the
- code, I am only committing the change to trunk.
-
-2007-12-28 18:26 +0000 [r95192] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 95191 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) |
- 6 lines Remove duplicate increment of the header count in the
- add_header() function. (closes issue #11648) Reported by: makoto
- Patch provided by sergee, committed patch by me, inspired by
- comments from putnopvut ........
-
-2007-12-28 16:12 +0000 [r95167] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_amd.c, CHANGES: Some changes to app_amd. The channel
- name is printed in verbose messages maximumWordLength option
- added. Duration of words that do not meet the minimum word
- duration will be logged The duration of pre-greeting silence will
- be logged Only consider us in the greeting if we actually
- detected a valid word duration. (closes issue #11650, reported
- and patched by davevg)
-
-2007-12-28 08:57 +0000 [r95139] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_video.c: fix a small bug in printing out
- geometries - wrong input.
-
-2007-12-28 00:17 +0000 [r95096] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 95095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec
- 2007) | 8 lines I found a bug while browsing the queue code and
- managed to reproduce it in a small setup. If a queue uses the
- ringall strategy, it was possible through unfortunate coincidence
- for a single member at a given penalty level to make app_queue
- think that all members at that penalty level were unavailable and
- cause the members at the next penalty level to be rung. With this
- patch, we will only move to the next penalty level if ALL the
- members at a given penalty level are unreachable. ........
-
-2007-12-27 23:32 +0000 [r95073] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_dictate.c, apps/app_mp3.c, apps/app_voicemail.c: remove
- more unnecessary casts for NULL. main/say.c is a big offender in
- this respect.
-
-2007-12-27 23:28 +0000 [r95070] Jason Parker <jparker@digium.com>
-
- * doc/asterisk.8, main/asterisk.c: Fix -s socket option, and
- document it as well. Closes issue #11645, patch by Laureano.
-
-2007-12-27 23:13 +0000 [r95068-95069] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_ices.c, apps/app_queue.c, apps/app_voicemail.c: NULL
- does not need to be cast to (char *)
-
- * channels/chan_oss.c: remove useless casts
-
-2007-12-27 21:41 +0000 [r95025] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 95024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) |
- 9 lines Don't report a syntax error when an empty string is
- passed to ast_get_group. Just return 0. (closes issue #11540)
- Reported by: tzafrir Patches: group_empty.diff uploaded by
- tzafrir (license 46) -- slightly changed by me ........
-
-2007-12-27 20:11 +0000 [r94978] Mark Michelson <mmichelson@digium.com>
-
- * /, main/io.c: Merged revisions 94977 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94977 | mmichelson | 2007-12-27 14:09:06 -0600 (Thu, 27 Dec
- 2007) | 3 lines Fixing a typo in a comment. ........
-
-2007-12-27 17:34 +0000 [r94908-94934] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_h323.c: Merged revisions 94924 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6
- lines Include types.h in chan_h323 as without it it can not be
- compiled on some operating systems like FreeBSD to name one.
- (closes issue #11585) Reported by: sobomax Patches:
- chan_h323.c.diff uploaded by sobomax (license 359) ........
-
- * /, channels/chan_sip.c: Merged revisions 94905 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94905 | file | 2007-12-27 13:27:11 -0400 (Thu, 27 Dec 2007) | 4
- lines Use ast_strlen_zero to see if our_contact is set or not on
- the dialog. It is possible for it to be a pointer to NULL.
- (closes issue #11557) Reported by: FuriousGeorge ........
-
-2007-12-27 17:26 +0000 [r94904] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/console_video.c: more
- localization of gui stuff
-
-2007-12-27 17:18 +0000 [r94903] Mark Michelson <mmichelson@digium.com>
-
- * doc/manager_1_1.txt: Adding documentation for new manager actions
- and events in app_queue
-
-2007-12-27 16:51 +0000 [r94902] Luigi Rizzo <rizzo@icir.org>
-
- * CHANGES: clarify the type of video support in chan_oss
-
-2007-12-27 16:11 +0000 [r94830-94877] Russell Bryant <russell@digium.com>
-
- * codecs/codec_g722.c: I went looking for where we downloaded the
- g722 implementation and came across these two links. So, I'm
- adding them so they are available for reference later.
-
- * /, main/translate.c, include/asterisk/translate.h: Merged
- revisions 94828-94829 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) |
- 9 lines Change ast_translator_best_choice() to only pay attention
- to audio formats. This fixes a problem where Asterisk claims that
- a translation path can not be found for channels involving video.
- (closes issue #11638) Reported by: cwhuang Tested by: cwhuang
- Patch suggested by cwhuang, with some additional changes by me.
- ........ r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27
- Dec 2007) | 2 lines Use the constant that I really meant to use
- here ... ........
-
-2007-12-27 09:13 +0000 [r94826-94827] Olle Johansson <oej@edvina.net>
-
- * funcs/func_dialplan.c: This function checks more than just
- contexts...
-
- * apps/app_pickupchan.c: - Add Copyright - Doxygen fixes Note: -
- This application needs better documentation and a RESULT code in
- the dialplan.
-
-2007-12-27 01:03 +0000 [r94825] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c, /: Merged revisions 94824 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94824 | kpfleming | 2007-12-26 18:01:47 -0700 (Wed, 26 Dec 2007)
- | 2 lines make this comment explain the situation in an even more
- explicit fashion ........
-
-2007-12-27 00:48 +0000 [r94819-94823] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c: more steps to decouple the gui from the
- rest of the code.
-
- * channels/console_gui.c, channels/console_video.c,
- channels/console_video.h: Enable building the code even if SDL is
- not present (similarly, SDL is also detected at runtime). Now we
- should be able to stream video even without a rendering device
- (useful for remote monitoring).
-
- * channels/console_gui.c, channels/console_video.c: more
- localizations around sdl_setup
-
- * channels/console_gui.c: use fread instead of mmap to read in the
- comment area from the keypad. fread is simpler and more portable,
- and there is no performance gain in using mmap.
-
- * images/kpad2.jpg: update the region description with an empty
- line at the beginning.
-
-2007-12-26 22:38 +0000 [r94818] Tilghman Lesher <tlesher@digium.com>
-
- * build_tools/cflags.xml, channels/chan_zap.c: Allow more spans
- than 32. Also, rearrange compiler flags so the most often used
- flags appear closer to the top. Reported by: tzafrir Patch by:
- tzafrir,tilghman (Closes issue #11528)
-
-2007-12-26 22:29 +0000 [r94817] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/console_video.c: another bunch
- of gui localizations
-
-2007-12-26 22:14 +0000 [r94814] Jason Parker <jparker@digium.com>
-
- * apps/app_exec.c: Make 'else' argument to ExecIf optional. Clean
- up the description and usage text a bit. Closes issue #11564,
- patch by pnlarsson (with some extra cleanup by me).
-
-2007-12-26 22:10 +0000 [r94810-94813] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/console_video.c: more
- localization of sdl stuff
-
- * channels/console_gui.c, channels/console_video.c,
- channels/console_video.h: move more gui stuff into console_gui.c
-
-2007-12-26 20:49 +0000 [r94809] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, /: Merged revisions 94808 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94808 | tilghman | 2007-12-26 14:43:38 -0600 (Wed, 26 Dec 2007)
- | 6 lines Workaround for what is probably a glibc bug (but we'll
- see this crop up again and again, if we don't add the
- workaround). Reported by: rolek Patch by: tilghman (Closes issue
- #11601, closes issue #11426) ........
-
-2007-12-26 20:02 +0000 [r94806] Jason Parker <jparker@digium.com>
-
- * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c,
- apps/app_authenticate.c, apps/app_zapscan.c, apps/app_zapras.c,
- apps/app_alarmreceiver.c, apps/app_amd.c, pbx/pbx_realtime.c,
- pbx/pbx_dundi.c, apps/app_zapateller.c, pbx/pbx_config.c,
- pbx/pbx_gtkconsole.c, apps/app_adsiprog.c, apps/app_cdr.c: Use
- defined return values in load_module in more places. (closes
- issue #11096) Patches: pbx_config.c.patch uploaded by moy
- (license 222) pbx_dundi.c.patch uploaded by moy (license 222)
- pbx_gtkconsole.c.patch uploaded by moy (license 222)
- pbx_loopback.c.patch uploaded by moy (license 222)
- pbx_realtime.c.patch uploaded by moy (license 222)
- pbx_spool.c.patch uploaded by moy (license 222)
- app_adsiprog.c.patch uploaded by moy (license 222)
- app_alarmreceiver.c.patch uploaded by moy (license 222)
- app_amd.c.patch uploaded by moy (license 222)
- app_authenticate.c.patch uploaded by moy (license 222)
- app_cdr.c.patch uploaded by moy (license 222)
- app_zapateller.c.patch uploaded by moy (license 222)
- app_zapbarge.c.patch uploaded by moy (license 222)
- app_zapras.c.patch uploaded by moy (license 222)
- app_zapscan.c.patch uploaded by moy (license 222)
-
-2007-12-26 20:01 +0000 [r94805] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c, channels/vcodecs.c,
- channels/console_video.c, channels/console_video.h: more
- preparation for untangling of the various console_video stuff
-
-2007-12-26 19:09 +0000 [r94796-94802] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 94801 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) |
- 4 lines Just in case the AST_FLAG_END_DTMF_ONLY flag was already
- set before starting autoservice, remember it and ensure that the
- channel has the same setting when autoservice gets stopped.
- (pointed out by d1mas, patched up by me) ........
-
- * funcs/func_dialplan.c (added), CHANGES: Add a new dialplan
- function, DIALPLAN_EXISTS(), which allows you to check for the
- existence of a dialplan target. (closes issue #11579) Reported
- by: irroot Patches: func_dialplan2.c uploaded by irroot (license
- 52) -- Additional changes by me.
-
- * main/autoservice.c, /: Merged revisions 94797 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) |
- 4 lines When a channel is in autoservice, mark a flag on the
- channel that says that we only care about the END of a digit.
- That way, no magic digit emulation stuff will happen when all
- we're doing is queueing up END frames. ........
-
- * main/channel.c: Leave a note for a minor bug that was pointed out
- by d1mas
-
-2007-12-26 18:05 +0000 [r94795] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: Convert raw bits for callprogress bitfield
- to use constants, for greater code clarity Reported by: dimas
- Patch by: dimas (Closes issue #11280)
-
-2007-12-26 17:26 +0000 [r94787-94794] Russell Bryant <russell@digium.com>
-
- * /, res/res_features.c: Merged revisions 94793 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94793 | russell | 2007-12-26 11:24:17 -0600 (Wed, 26 Dec 2007) |
- 3 lines Don't try to send a parked call back to itself. (closes
- issue #11622, reported by djrodman, patched by me) ........
-
- * Makefile, /: Merged revisions 94789 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94789 | russell | 2007-12-26 11:00:03 -0600 (Wed, 26 Dec 2007) |
- 5 lines List include/asterisk/version.h as a .PHONY target
- because we want the commands listed for this target to be
- executed regardless of whether the file exists or not. This fixes
- having the version not up to date when running from svn. (closes
- issue #11619, reported by plack, fixed by me) ........
-
- * main/autoservice.c, /: Merged revisions 94790 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) |
- 5 lines Don't store DTMF BEGIN frames while a channel is in
- autoservice. It's just going to make ast_read() do a lot of extra
- work when the channel comes back out of autoservice. (closes
- issue #11628, patched by me) ........
-
- * channels/chan_iax2.c: Fix a bug in peer handling that caused
- multiple instances of a peer to end up in the peers container
- after a reload. Somehow, this bug doesn't exist in 1.4 ...
- (closes issue #11626) (reported by pnlarsson, additional info
- from mvanbaak, fixed by me)
-
- * utils: update svn:ignore for astcanary
-
-2007-12-26 15:58 +0000 [r94782] Mark Michelson <mmichelson@digium.com>
-
- * configs/extconfig.conf.sample, main/logger.c, CHANGES: Adding
- support for storing the queue log entries in a realtime backend.
- (closes issue #11625, reported and patched by sergee) Thank you
- very much to sergee for adding this new feature!
-
-2007-12-26 10:14 +0000 [r94774] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_gui.c (added), channels/vcodecs.c (added),
- channels/console_video.c: Split console_video.c so that video
- codecs and gui functions are in separate files (still #include'd
- because of tangling in the data structures, but this is going to
- be cleaned up). The video grabbing functions still need to be
- moved to a separate file.
-
-2007-12-25 04:10 +0000 [r94771-94773] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_pickupchan.c (added): Add pickup by channel (Closes
- issue #11161)
-
- * channels/chan_zap.c, configs/zapata.conf.sample: Change the
- abbreviated TON from 'A' to 'V', since 'A' is a legitimate DTMF
- character. Also, fix the documentation to match the code.
-
- * res/res_agi.c: Add channel thread ID to the information passed to
- AGI. Reported by: dror99 Patch by: tilghman (Closes issue #11162)
-
-2007-12-24 19:43 +0000 [r94764-94768] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 94767 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007)
- | 5 lines Race: we need to wait to queue a NewChannel event until
- after the channel is inserted into the channel list. The reason
- is because some manager users immediately queue requests from the
- channel when they see that event and are confused when Asterisk
- reports no such channel. (Closes issue #11632) ........
-
- * /: Merged revisions 94763 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94763 | tilghman | 2007-12-24 09:39:56 -0600 (Mon, 24 Dec 2007)
- | 5 lines Another bit of bad logic in realtime_peer Reported by:
- dimas Patch by: dimas (Closes issue #11631) ........
-
-2007-12-23 14:51 +0000 [r94713-94741] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_video.c, channels/console_video.h: support
- sdl_videodriver to send output to x11/aalib/console
-
- * channels/console_video.c: move reading info from the keypad to a
- separate function. Remove an unused keypad field and some
- debugging messages. Adjust formatting on config file parsing
-
- * channels/console_video.c: make sure the minimum surface depth is
- 16bpp so we can create YUVoverlays. With this change we can do
- setenv SDL_VIDEODRIVER aalib and output to an ascii window (which
- is still in an X11 window). If you also do unsetenv DISPLAY then
- the output goes into the main asterisk window, unfortunately it
- interferes with the normal output so you don't see much. In any
- case, i don't think we are very far away from having a working
- xterm videophone!
-
- * channels/Makefile: avoid rebuilding dependent files if the
- generated busy.h and ringtone.h do not change. Ths masks (but
- does not solve) a but that i am seeing in doing a 'gmake install'
- without donig a 'gmake all' first.
-
-2007-12-23 01:38 +0000 [r94662] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 94660 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94660 | tilghman | 2007-12-22 19:21:03 -0600 (Sat, 22 Dec 2007)
- | 2 lines Argh... I suppose third time's the charm. ........
-
-2007-12-22 22:44 +0000 [r94615-94638] Luigi Rizzo <rizzo@icir.org>
-
- * configs/oss.conf.sample, channels/console_video.c: Change the
- name of config file entries for keypad regions from
- 'keypad_entry' to 'region'. Fix the example file accordingly.
- Also make some fixes in the code do reset entries on reload of
- the keypad. The recently committed kpad2.jpg has the correct
- names.
-
- * images/kpad2.jpg (added): add a sample keypad (with annotations)
- for console video
-
- * channels/console_video.c, channels/Makefile, channels/chan_oss.c,
- channels/console_video.h (added): Build console_video support by
- linking in, as opposed to including, console_video.c This will
- ease the task of splitting console_video.c into its components
- (V4L and X11 grabbers, various video codecs and packetizers,
- SDL), as well as ease future extensions (e.g. additional video
- sources, codecs and rendering engines). For the time being
- nothing changes for users: video support is off by default, and
- requires -DHAVE_VIDEO_CONSOLE on the command line to be included
- (if SDL and FFMPEG are available).
-
-2007-12-21 21:19 +0000 [r94593] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Something I've been itching to do for a
- while now. A minor optimization in app_voicemail. Since the
- dtable in base_encode always gets populated with the same values
- every time and never changes, make it static and const and only
- initialize it once. Also, there's no reason to define
- BASEMAXINLINE twice, so remove the redundant #define.
-
-2007-12-21 20:50 +0000 [r94549-94551] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: We should only clear this value if we have
- to
-
- * channels/chan_zap.c: Commit non TCP transport part of #11506.
- Includes numerous additional parameters, as well as RLT support
- for DMS type switches
-
-2007-12-21 20:38 +0000 [r94542-94548] Mark Michelson <mmichelson@digium.com>
-
- * res/res_config_sqlite.c: Store dates using local time instead of
- UTC (closes issue #11610, reported and patched by
- rbraun_performatique)
-
- * apps/app_queue.c: Fix a memory leak when reloading queue rules.
-
- * CHANGES: The one documentation source I forgot to update after
- the merge of the queue-penalty branch was the CHANGES file. No
- longer!
-
- * apps/app_voicemail.c: Lots of coding guidelines cleanup.
-
- * /, apps/app_voicemail.c: Merged revisions 94540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94540 | mmichelson | 2007-12-21 14:11:34 -0600 (Fri, 21 Dec
- 2007) | 8 lines Better quota support for using IMAP storage
- voicemail (closes issue #11415, reported by jaroth) (closes issue
- #11152, reported by selsky) Patch provided by jaroth ........
-
-2007-12-21 20:12 +0000 [r94541] Jason Parker <jparker@digium.com>
-
- * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c,
- codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c,
- codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_zap.c:
- codecs.conf really shouldn't be mandatory.. it never had been
- before, so let's go back to being optional. A big "thank you" to
- pnlarsson on IRC for allowing me access to his system to debug
- this. Closes issue #11584.
-
-2007-12-21 20:01 +0000 [r94477-94539] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 94538 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94538 | mmichelson | 2007-12-21 13:59:45 -0600 (Fri, 21 Dec
- 2007) | 5 lines The mail_copy c-client function does not expect a
- full imap mailbox string, just the name of the mailbox. (closes
- issue #11419, reported and patched by jaroth, with additional
- patchwork from me) ........
-
- * main/dial.c: AST_LIST_REMOVE_CURRENT only takes one argument in
- trunk
-
- * main/dial.c, /: Merged revisions 94468 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec
- 2007) | 6 lines Since we are freeing list elements within a list
- traversal, we need to use the safe traversal and remove the item
- from the list before freeing it. (closes issue 11612, reported by
- dtyoo) ........
-
-2007-12-21 16:12 +0000 [r94463-94465] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 94464 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94464 | mmichelson | 2007-12-21 10:11:44 -0600 (Fri, 21 Dec
- 2007) | 3 lines Removing a debug message I accidentally just
- committed ........
-
- * /, main/say.c, apps/app_queue.c: Merged revisions 94420 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94420 | mmichelson | 2007-12-21 09:45:14 -0600 (Fri, 21 Dec
- 2007) | 5 lines Fixing Portuguese syntax for saying dates and
- times. Also some coding guidelines cleanup. (closes issue #11599,
- reported and patched by caio1982, coding guidelines cleanup by
- me) ........
-
-2007-12-21 15:14 +0000 [r94419] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/asterisk.c: Merged revisions 94418 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94418 | tilghman | 2007-12-21 09:07:42 -0600 (Fri, 21 Dec 2007)
- | 2 lines Fix for restart-as-user problem reported via the -dev
- list ........
-
-2007-12-21 01:14 +0000 [r94345-94396] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Moved the update of the queue_ent's rule list
- to just before we try to call queue members. This allows for the
- change in penalty levels to be executed at the most logical time
- frame.
-
- * configs/queues.conf.sample, doc/tex/channelvariables.tex,
- apps/app_queue.c, configs/queuerules.conf.sample (added): Merging
- the queue-penalty branch. In short, this allows one to
- dynamically adjust the QUEUE_MAX_PENALTY and the newly introduced
- QUEUE_MIN_PENALTY during a call depending on the amount of time
- passed. The purpose is to allow the call to open up to more (or
- maybe just different) members without the caller's losing his
- place in the queue. See configs/queuerules.conf.sample for an
- example of how to set up queue rules and
- configs/queues.conf.sample for how to associate a rule with a
- queue. Along with the functional changes, new CLI and manager
- commands exist to show the rules defined and there is an
- additional CLI command to reload the queue rules. Future
- enhancements that may be made: support for realtime queue rules
- and support for dynamically adding a rule through the manager or
- CLI. Also a manager command to reload the queue rules (I'll
- probably write this myself very soon).
-
- * apps/app_voicemail.c: The changes to header inclusion in trunk
- broke compilation of app_voicemail when using IMAP storage. The
- reason is that c-client has its own definitions for LOG_WARNING
- and LOG_DEBUG, so we need to be sure to include asterisk's
- definitions last so that we use the proper values in
- app_voicemail. (closes issue #11437, reported by blitzrage, patch
- suggested by blitzrage)
-
-2007-12-20 22:39 +0000 [r94320] Russell Bryant <russell@digium.com>
-
- * configs/zapata.conf.sample: Add a bit more to the description of
- the "mwimonitor" option.
-
-2007-12-20 22:28 +0000 [r94319] Steve Murphy <murf@digium.com>
-
- * build_tools/make_buildopts_h: closes issue #11287; thanks to
- snuffy for this fix, which will surely make all solaris owners
- shout praises to his name.
-
-2007-12-20 20:25 +0000 [r94252-94257] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 94256 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r94256 | russell | 2007-12-20 14:22:22 -0600
- (Thu, 20 Dec 2007) | 13 lines Merged revisions 94255 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20
- Dec 2007) | 5 lines Fix another potential seg fault ... (closes
- issue #11606) Reported by: dimas ........ ................
-
- * channels/chan_zap.c, /: Merged revisions 94251 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94251 | russell | 2007-12-20 14:08:42 -0600 (Thu, 20 Dec 2007) |
- 10 lines Fix a deadlock in d-channel handling in chan_zap. This
- deadlock was introduced by the fix to ensure that channels are
- properly locked when handling channel variables. There were
- sections of this code where the channel pvt was locked before the
- channel lock, when in fact it _must_ be the other way around.
- (closes issue #11582) Reported by: bugi ........
-
-2007-12-20 12:56 +0000 [r94168-94191] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_usbradio.c, include/asterisk/config.h,
- channels/console_video.c, channels/chan_oss.c: add some macros to
- simplify parsing the config file, see description in config.h .
- They are a variant of the set of macros i used in chan_oss.c,
- structured in a way to be more robust to the presence of spurious
- ';' - basically, they define wrappers for 'do {' and '} while
- (0)', plus some helper functions to deal with simple cases such
- as ast_copy_string, ast_malloc, strtoul, ast_true ... The prefix
- (CV_ as 'Config Variable') tries to be easy to remember and has
- been chosen to not conflict with other existing macros in the
- tree. For the time being, I have only updated the three source
- files in the tree that used the old M_* macros. Hopefully, more
- files will be converted. NOTE: I understand that inventing my own
- dialect of C is generally wrong; however, the lack of adequate
- support in the language encourages lazy programming practices
- (such as ignoring errors, bounds, etc.) and this increases the
- chance of vulnerability in the code, especially because we are
- parsing user input here. Hopefully, these macros and the use of
- ast_parse_arg (in config.h) should encourage the programmer to
- write more robust code.
-
- * include/asterisk/paths.h, res/snmp/agent.c, utils/ael_main.c,
- utils/extconf.c, main/asterisk.c, utils/conf2ael.c: modify
- http://svn.digium.com/view/asterisk?view=rev&rev=93603 so that
- paths and filename are writable by asterisk.c without causing
- segfaults. This involves defining the variables as const char *,
- and having them point to as static, writable buffer defined in
- asterisk.c On passing, fix some errors in using these variables
- in some files in utils/ , and in res/snmp/agent.c which was
- redefining a variable without using paths.h (not applicable to
- 1.4)
-
-2007-12-19 23:17 +0000 [r94123-94124] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: 1. Unify the check for a penalty < 0 into the
- set_member_penalty code. 2. Fix an error when checking the CLI
- command for setting a member's penalty. 3. Fix a logging error if
- the incorrect parameter was the queue name or interface. (closes
- issue #11544, reported and patched by Laureano)
-
- * /, res/res_monitor.c: Merged revisions 94122 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec
- 2007) | 6 lines Sox versions 13.0.0 and newer do not have
- "soxmix" and instead use sox -m. res_monitor needs to use this if
- the user does not have soxmix. (closes issue #11589, reported by
- amessina, patch inspired by amessina but with a flourish from me)
- ........
-
-2007-12-19 22:51 +0000 [r94085] Russell Bryant <russell@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 94077 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r94077 | russell | 2007-12-19 16:48:48 -0600 (Wed, 19 Dec 2007) |
- 4 lines Check for the existence of the soxmix application on the
- target platform and have the result available in autoconfig.h.
- (part of issue #11589) ........
-
-2007-12-19 20:20 +0000 [r94052-94053] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Add 'voicemail reload' command. Reported
- by: eliel Patch by: eliel (Closes issue #11365)
-
- * apps/app_waituntil.c (added): Add contributed WaitUntil app.
- Original code by pprindeville, updated for trunk by tilghman.
- (Closes issue #11487)
-
-2007-12-19 19:29 +0000 [r94029] Russell Bryant <russell@digium.com>
-
- * include/asterisk/time.h: Add a couple of new time API calls -
- ast_tvdiff_sec and ast_tvdiff_usec (closes issue #11270) Reported
- by: dimas Patches: tvdiff_us-4.patch uploaded by dimas (license
- 88)
-
-2007-12-19 17:58 +0000 [r94002] Luigi Rizzo <rizzo@icir.org>
-
- * channels/console_video.c: Add instructions on how to generate
- your own font.
-
-2007-12-19 17:31 +0000 [r93956] Joshua Colp <jcolp@digium.com>
-
- * /: Merged revisions 93955 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93955 | file | 2007-12-19 13:29:20 -0400 (Wed, 19 Dec 2007) | 2
- lines Make the 1.4 builders happy, ensure var is NULL. ........
-
-2007-12-19 17:13 +0000 [r93952] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 93949 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93949 | tilghman | 2007-12-19 11:04:13 -0600 (Wed, 19 Dec 2007)
- | 3 lines Avoid segfault in chan_iax when peer isn't defined
- (Closes issue #11602) ........
-
-2007-12-19 17:09 +0000 [r93925-93950] Luigi Rizzo <rizzo@icir.org>
-
- * main/utils.c, include/asterisk/strings.h: Add a new API function,
- written at least twice in app_voicemail.c and likely in other
- places too. This is quite useful when placing mail/html stuff in
- config files. /*! \brief Convert some C escape sequences
- (\b\f\n\r\t) into the equivalent characters. \brief s The string
- to be converted (will be modified). \return The converted string.
- */ char *ast_unescape_c(char *s);
-
- * include/asterisk/config.h, main/config.c: add support for
- PARSE_DOUBLE, and remove identifiers for types not supported
- (INT16 and UINT16)
-
-2007-12-19 09:20 +0000 [r93899] Olle Johansson <oej@edvina.net>
-
- * CHANGES: Reorganize CHANGES a bit. The "misc" section grew too
- large...
-
-2007-12-19 08:57 +0000 [r93898] Luigi Rizzo <rizzo@icir.org>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4, makeopts.in: Properly document AST_EXT_TOOL_CHECK()
- and use it to check for NETSMP and GTK (GTK is not used thoug).
- AST_EXT_TOOL_CHECK() could be used for checking curl status as
- well, perhaps with a small addition because we currently seem to
- require a curl version greater than X.Y.Z Add a NETSMP_INCLUDE
- entry in makeopts.in We don't have yet any macros for using
- pkg-config to check for a specific package (right now there is
- only gtk2+ in the category).
-
-2007-12-19 08:57 +0000 [r93897] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding the
- ability to specify the To: header in an outbound INVITE by adding
- an exclamation mark to the dial string. This patch also exists
- for 1.4 in the fixtoheader-1.4 branch and has been in production
- for quite some time.
-
-2007-12-19 08:12 +0000 [r93875] Luigi Rizzo <rizzo@icir.org>
-
- * res/snmp/agent.c: make netsmp build under AST_DEVMODE.
- Description, included in the source, is below. I should note that
- the PACKAGE_* macros that asterisk defines in autoconfig.h are
- not used anywhere in the tree so they should just be removed. /*
- * There is some collision collision between netsmp and asterisk
- names, * causing build under AST_DEVMODE to fail. * * The
- following PACKAGE_* macros are one place. * Also netsnmp has an
- improper check for HAVE_DMALLOC_H, using * #if HAVE_DMALLOC_H
- instead of #ifdef HAVE_DMALLOC_H * As a countermeasure we define
- it to 0, however this will fail * when the proper check is
- implemented. */ No
-
-2007-12-19 07:01 +0000 [r93854] Olle Johansson <oej@edvina.net>
-
- * CHANGES, main/asterisk.c, doc/asterisk.sgml: Add option for
- starting remote Asterisk by naming the actual runtime socket
- instead of pointing to configuration file with -C Reported by:
- sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax
- (license 359) doc changes by committer (closes issue #11598)
-
-2007-12-19 00:09 +0000 [r93827] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * apps/app_osplookup.c: add missing header file
-
-2007-12-18 23:38 +0000 [r93804-93805] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: Making the canary error message a little more
- obvious.
-
- * utils/Makefile, utils/astcanary.c (added), main/asterisk.c: Add a
- canary process, for high priority mode (asterisk -p) to ensure
- that if Asterisk goes into a busy loop, the machine will be
- recoverable. We'd still need to do a restart to put Asterisk back
- into high priority mode, but at least a reboot won't be required.
- (Closes issue #11559)
-
-2007-12-18 21:13 +0000 [r93741] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Move some warnings away to debug since some
- devices send a packet with a silly string as a NAT keepalive
- packet.
-
-2007-12-18 18:39 +0000 [r93672] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
- 93668 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r93668 | tilghman | 2007-12-18 12:29:39 -0600
- (Tue, 18 Dec 2007) | 10 lines Merged revisions 93667 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18
- Dec 2007) | 2 lines Fixing AST-2007-027 (Closes issue #11119)
- ........ ................
-
-2007-12-18 18:20 +0000 [r93666] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/paths.h: remove a leftover line with only a '#'
- (wonder why the compiler does not complain!) and variables that
- are only used in asterisk.c
-
-2007-12-18 17:05 +0000 [r93626] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 93625 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec
- 2007) | 6 lines Rework deadlock avoidance used in ast_write,
- since it meant that agent channels which were being monitored had
- one audio file recorded and one empty audio file saved. (closes
- issue #11529, reported by atis patched by me) ........
-
-2007-12-18 10:24 +0000 [r93558-93603] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/paths.h, channels/chan_sip.c, res/res_crypto.c,
- utils/ael_main.c, utils/extconf.c, main/asterisk.c,
- res/res_monitor.c, utils/conf2ael.c: make configuration variable
- const so they are not accidentally modified. This requires
- casting the strings in asterisk.c when writing to them, so we do
- it through a macro to do it consistently.
-
- * channels/chan_unistim.c, res/res_crypto.c, main/astmm.c,
- apps/app_ices.c, utils/extconf.c, channels/chan_iax2.c,
- main/asterisk.c, main/config.c, main/db.c, apps/app_adsiprog.c,
- cdr/cdr_csv.c: remove unnecessary (char *) casts for
- ast_config_AST_* variables. There are some left in the .flex
- files, left to the maintainer...
-
- * build_tools/make_defaults_h, main/asterisk.c: Rename the macros
- in defaults.h - they are not meant to be globally visible.
- Document the fact that DEFAULT_TMP_DIR cannot be overridden from
- the default configuration (this needs to be fixed, as you could
- have a totally different spooldir configured at runtime, and yet
- DEFAULT_TMP_DIR keeps the compile-time default). Remove two
- unused entries for sounds and images.
-
- * Makefile.moddir_rules: make the code match documentation - now
- you can specify multiple words in MODULE_PREFIX.
-
- * CREDITS: Name the people responsible for some recent
- contributions to the tree.
-
- * Makefile: Two small changes: + document the difference between
- "A=foo make ..." and "make A=foo ..." and suggest using
- COPTS/LDOPTS if you need to use the second form to pass compiler
- and loader flags; + define only in one place the environment used
- to build stuff in menuselect/
-
-2007-12-18 07:56 +0000 [r93557] Olle Johansson <oej@edvina.net>
-
- * doc/CODING-GUIDELINES: A minor update, caused by a recent bug
- report ;-)
-
-2007-12-18 07:22 +0000 [r93536] Luigi Rizzo <rizzo@icir.org>
-
- * doc/CODING-GUIDELINES: small documentation update (nothing
- important).
-
-2007-12-18 02:57 +0000 [r93514] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_unistim.c: You... will... build! I say so and
- therefore you will.
-
-2007-12-18 02:42 +0000 [r93493] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_unistim.c, include/asterisk/threadstorage.h: minor
- cleanups
-
-2007-12-17 23:10 +0000 [r93464] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_unistim.c: fix building under cygwin. At this point
- WINARCH should go away.
-
-2007-12-17 22:54 +0000 [r93405] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_unistim.c: remove some unnecessary includes
-
-2007-12-17 22:50 +0000 [r93390] Jason Parker <jparker@digium.com>
-
- * /, main/translate.c: Merged revisions 93381 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4
- lines What was I thinking when I wrote this masterpiece? -1 + 1 =
- 0.. who woulda thunk it?. ........
-
-2007-12-17 22:38 +0000 [r93380] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_oss.c: surprising as it may be, chan_oss compiles
- correctly under cygwin as well, provided you look for soundcard.h
- in the right place...
-
-2007-12-17 22:29 +0000 [r93378] Joshua Colp <jcolp@digium.com>
-
- * /, main/utils.c: Merged revisions 93377 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7
- lines Do not try to access information about a lock when printing
- out a trylock attempt. It is possible for the lock that it
- references to no longer be valid. This would have caused
- segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
- Reported by: callguy (closes issue #11100) Reported by: callguy
- ........
-
-2007-12-17 21:14 +0000 [r93337] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/time.h: Merged revisions 93336 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r93336 | tilghman | 2007-12-17 15:12:42 -0600 (Mon, 17
- Dec 2007) | 6 lines Today is tomorrow's yesterday, and
- yesterday's tomorrow is today, and tomorrow's tomorrow is the day
- after tomorrow, so who cares if you recycle anyway? If this
- confuses you, that's nothing compared to what this fixes. ;-)
- ........
-
-2007-12-17 21:12 +0000 [r93335] Olle Johansson <oej@edvina.net>
-
- * channels/chan_zap.c, /, channels/chan_sip.c, apps/app_queue.c,
- channels/chan_iax2.c, channels/chan_mgcp.c: Merged revisions
- 93182 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8
- lines Issue 11574: Add dependencies on res_monitor and
- res_features. I wonder if Asterisk can run at all without
- res_features. My guess is that there's propably a lot of more
- modules and the core that depends on it. Reported by: caio1982
- (closes issue #11574) ........
-
-2007-12-17 20:42 +0000 [r93293-93297] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Removing some leftover debug messages from a
- while back.
-
- * /, apps/app_voicemail.c: Merged revisions 93291 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec
- 2007) | 6 lines We need to create the directory for a voicemail
- user even if they are using IMAP storage since greetings are
- stored in the filesystem. (closes issue #11388, reported by
- spditner, patch by me inspired by a patch by spditner) ........
-
-2007-12-17 18:07 +0000 [r93252] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 93250 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93250 | file | 2007-12-17 14:05:55 -0400 (Mon, 17 Dec 2007) | 6
- lines If a call is received with a called number IE containing
- nothing go to the 's' extension. (closes issue #9099) Reported
- by: kb1_kanobe2 Patches: 20070906__9099.diff.txt uploaded by
- Corydon76 (license 14) ........
-
-2007-12-17 17:16 +0000 [r93191-93224] Kevin P. Fleming <kpfleming@digium.com>
-
- * utils: all created files need to be listed in the ignore property
-
- * channels/chan_unistim.c, build_tools/menuselect-deps.in,
- configure, configure.ac, channels/Makefile, channels/chan_oss.c:
- make the configure script detect that it is running on a Windows
- platform, and report that information so that menuselect can use
- it (all information that is used to decide whether to build
- modules or not must be fed to menuselect so the user knows what
- will be built and why... don't make module build decisions in the
- makefiles, please)
-
- * Makefile: make using PRINT_DIR a little easier
-
-2007-12-17 15:18 +0000 [r93187-93190] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix usage of rtptimeout. It can be used
- without rtpkeepalive, and the value can not be accessed directly
- in the SIP pvt structure. All RTP related timeouts have to be
- retrieved using the ast_rtp_* function calls. (closes issue
- #11562) Reported by: ibc
-
- * channels/chan_unistim.c: If no timezone is available use the
- default message. (closes issue #11576) Reported by: junky
-
- * channels/chan_unistim.c: Make chan_unistim actually be able to
- unload. When creating a thread that you want to pthread_join you
- have to explicitly create it as joinable, and also if using
- pthread_cancel you have to have a pthread_testcancel to see if it
- has been called.
-
-2007-12-17 07:27 +0000 [r93184-93185] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs, /, build_tools/make_version,
- include/asterisk/autoconfig.h.in, configure.ac, apps,
- Makefile.moddir_rules, res/Makefile, pbx/Makefile,
- build_tools/prep_moduledeps (removed), channels/Makefile, cdr,
- formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
- configure, build_tools/embed_modules.xml, cdr/Makefile,
- build_tools/prep_tarball, makeopts.in, formats/Makefile, res,
- pbx, channels, funcs/Makefile: Merged revisions 93180 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007)
- | 23 lines In
- http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
- rizzo brought up some issues related to the way that the metadata
- required for menuselect and the rest of the build system is
- extracted from the source files. Since I had a few hours to kill
- on an airplane today, I decided to improve this situation... so
- now the system caches the extracted metadata and uses it to build
- the menuselect 'tree' as much as it can. The result of this is
- that when a single source file is changed, only the metadata for
- that file needs to be extracted again, and the rest is used from
- the cache files. I also reduced the number of forked processes
- required to do the metadata extraction; it was actually possible
- to do most of what we needed in the Makefiles themselves without
- using any shell scripts at all! On my laptop, these changes
- resulted in an 80% decrease in the time required for the
- 'menuselect.makeopts' automatic check to occur after editing a
- single source file. While doing this work I also cleaned up a few
- minor things in the Makefiles, adding a check for 'awk' to the
- configure script and changed all remaining places we use 'grep'
- or 'awk' to use the ones found by the configure script, and
- changed the 'prep_tarball' script to build the menuselect
- metadata so that tarballs of Asterisk will include it and won't
- require the user to wait while it is extracted after unpacking.
- ........
-
-2007-12-16 19:06 +0000 [r93173] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: menuselect.makeopts is not a .PHONY target
-
-2007-12-16 13:38 +0000 [r93163-93167] Olle Johansson <oej@edvina.net>
-
- * pbx/pbx_dundi.c: Convert from LOG_DEBUG etc to ast_debug. Thanks,
- dimas! (closes issue #11572) Reported by: dimas Patches:
- dundilog-trunk.patch uploaded by dimas (license 88)
-
- * main/manager.c, CHANGES: Adding a new CLI command for "manager
- reload", which is important now that you need to reload after
- changes. Thanks YS. Reported by: ys Patches:
- trunk93163_manager_reload.c.diff uploaded by ys (license 281)
- (related to issue #11414)
-
- * main/manager.c, CHANGES: Change manager so that registered
- accounts are stored in memory. This opens for a manager realtime
- implementation. If you change accounts in manager.conf, you now
- need to reload to activate the changes (deletions, additions).
- This was not the case with 1.4. Reported by: ys Patches:
- trunk93163_manager_reload.c.diff uploaded by ys (license 281)
- (closes issue #11414)
-
- * CHANGES: Adding console_video to CHANGES. It's important that we
- keep this file up to date, even with experimental stuff.
-
- * channels/chan_unistim.c, main/udptl.c, configs/dundi.conf.sample,
- channels/chan_sip.c, include/asterisk/rtp.h,
- include/asterisk/netsock.h, channels/iax2-provision.c,
- UPGRADE.txt, doc/tex/qos.tex, configs/skinny.conf.sample,
- CHANGES, channels/chan_iax2.c, main/rtp.c, main/netsock.c,
- configs/h323.conf.sample, configs/iax.conf.sample,
- channels/chan_skinny.c, configs/mgcp.conf.sample,
- configs/unistim.conf.sample, channels/chan_h323.c,
- configs/iaxprov.conf.sample, pbx/pbx_dundi.c,
- configs/sip.conf.sample, channels/chan_mgcp.c: HUGE improvements
- to QoS/CoS handling by IgorG - Refer to the proper documentation
- - Implement separate signalling/media QoS/CoS in many channels
- using RTP - Improve warnings and verbose messages - Deprecate
- some old settings Minor modifications by me, a big effort from
- IgorG. Thanks! Reported by: IgorG Patches:
- qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
- Tested by: IgorG (closes issue #11145)
-
-2007-12-16 10:34 +0000 [r93162] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: use a simpler idiom for 'cmp -s ...'
-
-2007-12-16 09:37 +0000 [r93152-93161] Olle Johansson <oej@edvina.net>
-
- * main/asterisk.c: Don't drop the first character randomly in long
- listings in the CLI. Reported by: slavon Patches:
- asterisk-consolerefresh2.diff.txt uploaded by jamesgolovich
- (license 176) Tested by: eliel (closes issue #9325)
-
- * configs/sip.conf.sample, CHANGES: Update documentation
-
- * channels/chan_sip.c, configs/sip.conf.sample: Make more timers
- settable in SIP so that we can force timeout earlier on
- non-responsive SIP servers. Thanks, jcmoore, for the patch!
- Reported by: jcmoore Patches: peer_t1_timerb_trunk_v3.patch.txt
- uploaded by jcmoore (license 9) (closes issue #9771)
-
- * include/asterisk/file.h: Typo fixed earlier, that wasn't a typo
- after all. Didn't a clever guy once say "Compile before you
- commit" ? :-)
-
-2007-12-15 08:10 +0000 [r93151] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h: fix a typo from revision 93138
-
-2007-12-15 00:44 +0000 [r93138-93145] Luigi Rizzo <rizzo@icir.org>
-
- * configs/oss.conf.sample: configuration options related to video
- support.
-
- * channels/console_video.c (added): Bring in video console support
- for chan_oss (and later chan_alsa too). This is disabled in the
- default build, you need to explicitly enable it compiling with
- make COPTS=-DHAVE_VIDEO_CONSOLE In return, you will be able to do
- a video call with chan_oss, using the webcam (or X11 grabbing) as
- local source, and rendering the incoming stream on your screen.
- Currently supported formats are h261, h263, h263+, h264, mpeg4
- (all through the avcodec lib, part of ffmpeg). Incoming video is
- on the left, outgoing video is on the right, while the center
- displays a keypad (if configured so). Right clicking on the video
- windows increases the size, center clicking reduces the size.
- Dragging the mouse (with the left key) on the right window while
- the X11 grabber is active moves the grab area. This is the result
- of work by Sergio Fadda, Marta Carbone and myself, all properly
- disclaimed to digium. Note, there is a lot of work left to do in
- this module, including adding support for Video4LinuxV2 (I have
- patches from Matteo Brancaleoni which should be integrated), and
- making the GUI a lot more friendly than it is now (e.g.
- supporting merging or switching among multiple sources, a text
- window, and more).
-
- * channels/chan_oss.c: remove some redundant headers
-
- * include/asterisk/file.h: include mmap header if detected by
- configure
-
-2007-12-14 22:02 +0000 [r93094-93115] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Resolve a compiler warning
-
- * apps/app_voicemail.c: Change places where the name "INBOX" was
- hardcoded to use the imapfolder setting from voicemail.conf
- instead. This commit will help to get issue #11415 moving towards
- commitment.
-
-2007-12-14 21:09 +0000 [r93090] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile, channels/chan_unistim.c, codecs/ilbc/iLBC_define.h:
- Solaris compat fixes Reported by: snuffy Patch by:
- snuffy,tilghman (Closes issue #11315)
-
-2007-12-14 19:31 +0000 [r93067] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: make something static
-
-2007-12-14 19:27 +0000 [r93066] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_privacy.c, UPGRADE.txt, CHANGES,
- configs/privacy.conf.sample (removed): Remove use of privacy.conf
- by the Privacy app. Reported by: eliel Patch by: eliel (Closes
- issue #11344)
-
-2007-12-14 19:19 +0000 [r93042-93065] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, main/manager.c, funcs/func_timeout.c: I needed to
- increment the numbers used on the VERBOSITY_ATLEAST calls by 1.
- Thanks to kpfleming for pointing this out.
-
- * include/asterisk/logger.h, main/pbx.c, main/manager.c,
- funcs/func_timeout.c: Changed VERBOSITY_LEVEL to
- VERBOSITY_ATLEAST to be more accurate.
-
- * include/asterisk/logger.h, main/pbx.c, main/manager.c,
- funcs/func_timeout.c, main/logger.c: After reading Russell's
- e-mail to the dev list stating that checking option_verbose is
- not equivalent to the check done by ast_verb, I wrote a macro,
- VERBOSITY_LEVEL, which does this check. I did a quick look in the
- source and used this macro in some places where option_verbose
- was used. I also converted some verbose messages in logger.c to
- use ast_verb instead of ast_verbose.
-
-2007-12-14 18:24 +0000 [r93041] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_meetme.c: gcc 4.1.3 wants a union used here.
-
-2007-12-14 17:49 +0000 [r93001-93004] Russell Bryant <russell@digium.com>
-
- * main/config.c: Print an error message if a #included file does
- not exist
-
-2007-12-14 17:29 +0000 [r92999] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Publish the AGI events to manager. Reported by:
- moy Patch by: moy,tilghman (Closes issue #11337)
-
-2007-12-14 15:59 +0000 [r92976] Mark Michelson <mmichelson@digium.com>
-
- * funcs/func_timeout.c: Reintroduce an optimization that was lost
- when converting trunk to use ast_verb.
-
-2007-12-14 15:49 +0000 [r92939] Tilghman Lesher <tlesher@digium.com>
-
- * main/editline/sys.h: If malloc.h is included in a Solaris build,
- the compilation breaks. Reported by: snuffy Patch by: snuffy
- (Closes issue #11313)
-
-2007-12-14 15:18 +0000 [r92938] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 92937 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92937 | file | 2007-12-14 11:16:15 -0400 (Fri, 14 Dec 2007) | 4
- lines Up the length of the format on the SIP channel since it can
- now be rather long. (closes issue #11552) Reported by:
- francesco_r ........
-
-2007-12-14 15:14 +0000 [r92936] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 92933 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92933 | tilghman | 2007-12-14 09:01:10 -0600 (Fri, 14 Dec 2007)
- | 5 lines Change help documentation to match actual behavior
- (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
- (Closes issue #11548) ........
-
-2007-12-14 15:08 +0000 [r92935] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 92934 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r92934 | crichter | 2007-12-14 16:05:28 +0100 (Fr, 14
- Dez 2007) | 1 line fixed the sequencing of WAITING_4DIGS state
- setting and overlap_task thread starting. ........
-
-2007-12-14 14:48 +0000 [r92913] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, main/pbx.c, main/srv.c, channels/chan_skinny.c,
- res/res_features.c, apps/app_minivm.c, apps/app_amd.c,
- res/snmp/agent.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
- main/asterisk.c, main/netsock.c, apps/app_voicemail.c: Convert
- ast_verbose to ast_verb. Reported by: snuffy Patch by: snuffy
- (Closes issue #11547)
-
-2007-12-14 01:25 +0000 [r92876] Mark Michelson <mmichelson@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 92875 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r92875 | mmichelson | 2007-12-13 19:24:06 -0600 (Thu, 13
- Dec 2007) | 7 lines When compiling with DETECT_DEADLOCKS, don't
- spam the CLI with messages about possible deadlocks. Instead just
- print the intended single message every five seconds. (closes
- issue 11537, reported and patched by dimas) ........
-
-2007-12-13 23:10 +0000 [r92816-92855] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_meetme.c: When working with dates, use numeric form
- whenever possible, as it's faster. Also, a bunch of coding
- guidelines fixes.
-
- * channels/chan_zap.c, /: Merged revisions 92815 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92815 | tilghman | 2007-12-13 15:28:39 -0600 (Thu, 13 Dec 2007)
- | 5 lines Properly initialize polarity statuses, so that they are
- detected properly. Reported by: julianjm Patch by: julianjm
- (Closes issue #10238) ........
-
-2007-12-13 20:23 +0000 [r92811] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/app.h, include/asterisk/module.h, res/res_agi.c,
- apps/app_rpt.c: Move usage of the old LOCAL_USER_* macros to the
- new ast_module_user_* functions in a few documentation places.
- (closes issue #11533) Reported by: IgorG Patches:
- oldmacroclean.v1.diff uploaded by IgorG (license 20)
-
-2007-12-13 20:14 +0000 [r92810] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, /: Merged revisions 92809 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92809 | qwell | 2007-12-13 14:13:48 -0600 (Thu, 13 Dec 2007) | 1
- line Make application help text a little more clear about the use
- of extensions in a filename. ........
-
-2007-12-13 20:12 +0000 [r92806-92808] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 92807 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92807 | mmichelson | 2007-12-13 14:03:20 -0600 (Thu, 13 Dec
- 2007) | 3 lines Prevent another potential fd leak ........
-
- * /, apps/app_voicemail.c: Merged revisions 92803 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92803 | mmichelson | 2007-12-13 13:49:55 -0600 (Thu, 13 Dec
- 2007) | 3 lines Prevent a possible fd leak. ........
-
-2007-12-13 17:46 +0000 [r92779] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_adaptive_odbc.c: Don't use backslash as an escape
- character, unless it really is an escape character.
-
-2007-12-13 16:23 +0000 [r92758] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Remove remnants of a poorly merged commit.
- (92697)
-
-2007-12-13 15:40 +0000 [r92737] Doug Bailey <dbailey@digium.com>
-
- * apps/app_voicemail.c: Tag voicemails with UTC time as opposed to
- local time zone
-
-2007-12-13 00:18 +0000 [r92697] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_h323.c, main/config.c:
- Merged revisions 92696 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #10690) ........ r92696 | qwell | 2007-12-12 18:11:09 -0600
- (Wed, 12 Dec 2007) | 7 lines If a typo is found in a config file,
- we previous continued on with what was already loaded. We do not
- want to do this (see bug below for details). This makes it so
- that if a [ is found without a ], the entire config will fail,
- and nothing in it will be loaded. Issue 10690. ........
-
-2007-12-12 23:44 +0000 [r92676] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Revert an "optimization" that I added in
- revision 89887, as the user who reported issue #11449 has
- demonstrated that it actually was a performance hit on his
- machine. I think that it is possible that it could still be a
- benefit on systems under higher load, especially SMP systems, but
- I don't have enough time or interest to find out at the moment.
- (closes issue #11449)
-
-2007-12-12 21:22 +0000 [r92618] Jason Parker <jparker@digium.com>
-
- * /, apps/app_meetme.c, channels/ringtone.h: Merged revisions 92617
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11048) ........ r92617 | qwell | 2007-12-12 15:15:45 -0600
- (Wed, 12 Dec 2007) | 4 lines Don't increment user count until
- after name has been recorded (if enabled). Issue 11048, tested by
- pep. ........
-
-2007-12-12 20:05 +0000 [r92594] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, main/logger.c, main/utils.c,
- apps/app_mixmonitor.c: Conversions of free to ast_free, where
- applicable, and several other formatting fixes. Reported by:
- eliel Patch by: eliel,tilghman (Closes issue #11209)
-
-2007-12-12 19:50 +0000 [r92562] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Merged revisions 92556 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92556 | russell | 2007-12-12 13:40:02 -0600 (Wed, 12 Dec 2007) |
- 1 line resolve compiler warning ........
-
-2007-12-12 17:51 +0000 [r92511-92526] Mark Michelson <mmichelson@digium.com>
-
- * res/res_features.c: Same change to trunk as revision 92510. I'm
- not sure why I merged this way, but I did.
-
-2007-12-12 17:15 +0000 [r92476-92507] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: Correctly handle possible memory allocation
- failure Reported by: eliel Patch by: eliel (Closes issue #11512)
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 92463 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92463 | tilghman | 2007-12-12 10:52:56 -0600 (Wed, 12 Dec 2007)
- | 4 lines Test directly for the API that fixed AST-2007-026, to
- ensure that older versions of PostgreSQL are no longer
- acceptable. (Closes issue #11526) ........
-
-2007-12-12 16:11 +0000 [r92444] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 92443 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92443 | mmichelson | 2007-12-12 10:08:55 -0600 (Wed, 12 Dec
- 2007) | 3 lines Removing an unused variable. ........
-
-2007-12-11 22:20 +0000 [r92423] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/term.h, channels/misdn/isdn_msg_parser.c,
- channels/ringtone.h, include/asterisk/ulaw.h,
- include/jitterbuf.h, include/asterisk/manager.h,
- include/asterisk/transcap.h, channels/misdn/isdn_lib.c,
- channels/gentone.c, include/asterisk/zapata.h,
- channels/misdn/isdn_lib.h, include/asterisk/doxyref.h,
- channels/DialTone.h, channels/misdn/ie.c,
- channels/misdn/chan_misdn_config.h, channels/iax2.h,
- channels/misdn/portinfo.c, include/asterisk/udptl.h,
- main/cygload.c, include/asterisk/translate.h: Doxygen updates,
- formatting. misdn stuff needs a lot of doxygenification (Hello,
- Qwell :-) )
-
-2007-12-11 22:10 +0000 [r92422] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
- configure.ac: Trunk build would fail due to the nonexistence of
- zaptel hwgain structures missing. Patched configure to check for
- this stuff and put a #ifdef around the offending code in
- chan_zap. Thanks to file for overseeing this.
-
-2007-12-11 21:58 +0000 [r92421] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: We need to set the address we want to match
- against before we actually do the match.. Closes issue #11518.
-
-2007-12-11 21:46 +0000 [r92402] Mark Michelson <mmichelson@digium.com>
-
- * res/res_musiconhold.c: Removing a pointless memset. The memory
- was just calloc'd, so the memory is already zeroed out
-
-2007-12-11 21:17 +0000 [r92401] Jason Parker <jparker@digium.com>
-
- * apps/app_controlplayback.c: Add variable to show which key was
- pressed to stop playback. Issue #11377, initial patch by johan.
-
-2007-12-11 20:06 +0000 [r92364-92365] Joshua Colp <jcolp@digium.com>
-
- * res/res_monitor.c: Only look to see if options are set if some
- have been provided. (closes issue #11505) Reported by: Mike
- Anikienko
-
- * main/global_datastores.c, /: Merged revisions 92363 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r92363 | file | 2007-12-11 15:51:40 -0400 (Tue, 11 Dec
- 2007) | 6 lines Fix potential memory leak with the dialed
- interfaces list if another memory allocation fails. (closes issue
- #11507) Reported by: eliel Patches: global_datastores.c.patch
- uploaded by eliel (license 64) ........
-
-2007-12-11 17:44 +0000 [r92324] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 92323 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec
- 2007) | 10 lines Fixing autofill to be more accurate.
- Specifically, if calls ahead of the current caller were ringing
- members (but not yet bridged) there could be available members
- and waiting callers who would not get matched up. The member
- availability checker was correctly determining the number of
- available members in this scenario, but the queue itself did not
- parallelly reflect this status on the pending calls. This commit
- corrects the issue. (closes issue #11459, reported by
- equissoftware, patched by me) ........
-
-2007-12-11 16:29 +0000 [r92305] Russell Bryant <russell@digium.com>
-
- * include/asterisk/unaligned.h, main/event.c: * In unaligned.h,
- remove some unnecessary casts and mark the arg of the
- get_unaligned functions as const * In event.c, use
- get_unaligned_uint32() in a couple of places to fix issues on
- architectures that don't allow unaligned access
-
-2007-12-11 14:17 +0000 [r92267-92285] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/devicestate.h, include/asterisk/agi.h,
- include/asterisk/astobj2.h, include/asterisk/extconf.h,
- include/asterisk/io.h, include/asterisk/cdr.h,
- include/asterisk/aes.h, include/asterisk/_private.h,
- include/asterisk/localtime.h, include/asterisk/hashtab.h,
- include/asterisk/callerid.h, include/asterisk/logger.h,
- include/asterisk/doxyref.h, include/asterisk/app.h,
- include/asterisk/adsi.h, include/asterisk/event.h,
- include/asterisk/causes.h, include/asterisk/alaw.h,
- include/asterisk/ast_expr.h, include/asterisk/dsp.h,
- include/asterisk/mod_format.h, include/asterisk/ael_structs.h,
- include/asterisk/astdb.h: A lot of doxygen updates
-
- * include/asterisk/frame.h: Doxygen updates
-
-2007-12-10 20:18 +0000 [r92243] Doug Bailey <dbailey@digium.com>
-
- * channels/chan_zap.c: Add CLI commands to dynamically set hw and
- sw gains
-
-2007-12-10 16:48 +0000 [r92205-92206] Joshua Colp <jcolp@digium.com>
-
- * utils/check_expr.c: Add ast_atomic_fetchadd_int_slow to
- check_expr for platforms that need it. (closes issue #11484)
- Reported by: snuffy
-
- * /, main/rtp.c: Merged revisions 92204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6
- lines Add G729A as another possible payload name for G729. Some
- devices use this instead of G729, which is perfectly normal since
- the payload number itself is defined and can't be used by
- anything else so the name doesn't matter that much. (closes issue
- #11483) Reported by: revolution Patches: rtp.diff uploaded by
- revolution (license 346) ........
-
-2007-12-10 16:30 +0000 [r92203] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 92202 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec
- 2007) | 7 lines If there are no members in a queue, then the loop
- where the datastore for detecting duplicate dialed numbers will
- be skipped, meaning the datastore isn't created. This means that
- when we try to free it, there's a crash. This stops that crash
- from occurring. (closes issue #11499, reported by slavon, patched
- by eliel) ........
-
-2007-12-10 16:15 +0000 [r92199-92201] Joshua Colp <jcolp@digium.com>
-
- * res/res_agi.c: Only send a SIGHUP if the pid is greater than -1,
- otherwise all PIDs greater than -1 will get the SIGHUP... and
- that is bad. (closes issue #11453) Reported by: alanmcmillan
-
-2007-12-10 14:18 +0000 [r92140-92160] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Removing some LOG_DEBUG items
-
- * /, channels/chan_sip.c: Merged revisions 92158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16
- lines Avoid reinvite race situations with two Asterisks trying to
- reinvite each other in 1.4 and trunk. This patch implements
- support for the 491 error code that Asterisk 1.4 generates on
- situations where we get an incoming INVITE and already has one in
- progress. Thanks to mavetju for reporting and to Raj Jain for an
- excellent explanation of the problem. Patch by myself. Tested
- with 8 Asterisk servers connected to each other in a training
- network. Closes issue #10481 ........
-
- * doc/manager_1_1.txt, apps/app_voicemail.c: Add a few extra
- headers in the voicemail users listing in manager 1.1. Update
- documentation too. (closes issue #11495) Reported by: caio1982
- Patches: extra_vm_manager_info1.diff uploaded by caio1982
- (license 22)
-
-2007-12-10 09:00 +0000 [r91929-92122] Luigi Rizzo <rizzo@icir.org>
-
- * build_tools/make_version, build_tools/make_version_h:
- simplify/cleanup the scripts
-
- * utils/Makefile: remove relative paths and use ASTTOPDIR instead.
- Give a default value to ASTTOPDIR if unset so we can at least do
- a 'make clean' without too much trouble. The proper fix, however,
- is to partition the top level Makefile in a 'setup' and a 'main'
- part, in a way that the 'setup' part can be included from
- subdirs' Makefiles and allow targets to be built without going
- through the top level Makefile.
-
- * utils/clicompat.c: simplify this file
-
- * doc/CODING-GUIDELINES: add a bit of info on the build
- infrastructure
-
- * Makefile: Fix the detection of modules installed from this build.
- You can now add the path of local module subdirs from the command
- line with make LOCAL_MOD_SUBDIRS= ....
-
- * codecs/Makefile, apps/Makefile, Makefile.moddir_rules,
- cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile,
- formats/Makefile, funcs/Makefile: Put into Makefile.moddir_rules
- the common instructions used to generate loadable and embedded
- module lists. Individual Makefiles now are a lot simpler,
- possibly as simple as this: -include
- $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
- MODULE_PREFIX=cdr_ all: _all include
- $(ASTTOPDIR)/Makefile.moddir_rules and also more flexible because
- in a single directory we can combine various types of modules
- (app_, cdr_, func_, ... ) by simply listing them in the
- MODULE_PREFIX variable. The individual Makefiles can also create
- list of modules to be excluded by listing them in the variablel
- MODULE_EXCLUDE (see an example in channels/Makefile). With this
- change it becomes trivial to integrate a directory with locally
- created/modified sources into the main build.
-
- * Makefile, Makefile.moddir_rules: make the install target a bit
- less noisy
-
- * Makefile: document usage of several exported variables
-
- * utils/Makefile: add hashtab.c to the list of files deleted
-
- * Makefile.moddir_rules: another place where ../ should have been
- ASTTOPDIR
-
- * codecs/Makefile, utils/Makefile, apps/Makefile, cdr/Makefile,
- pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile,
- funcs/Makefile: normalize subdirs' Makefile by using ASTTOPDIR
- and not .. to reference the top level directory.
-
- * Makefile: Implement the outcome of a discussion on the -dev list
- re. the use of DESTDIR and INSTALL_PATH - many thanks to Tzafrir
- Cohen and Simon Perreault for extremely useful feedback: 1.
- comment out the [directories] section the created asterisk.conf ;
- you can set the correct defaults at build time using
- INSTALL_PATH, so the repetition here is redundant and often
- wrong. (The next step now is move asterisk.conf outside the
- Makefile to asterisk.conf.sample, because there is little if
- anything here that needs to be constructed at build/install
- time). 2. use DESTDIR?=$(INSTALL_PATH) so you only need to
- specify a path once if the two coincide. This should have no ill
- side effects, because if you don't specify DESTDIR, you really
- need INSTALL_PATH="" to set the correct defaults, and if you
- specify DESTDIR the value is not overridden. The second part
- required moving the 'export DESTDIR' right after the assignment
- to prevent DESTDIR getting set by the export (this is documented
- in the Makefile).o hopefully avoid the mistake)$ With this change
- you can now do something like this from your source tree: make
- INSTALL_PATH=/some/place install samples and then main/asterisk
- -vdc which will pick up the correct config files and libraries
- from /some/place - i.e. great for developers!
-
- * main/config.c: remove unused code, and simplify the logic for
- #include/#exec (still a lot of cleanup needed here).
-
- * main/config.c: Implement comment_buffer and lline_buffer in terms
- of the ast_str_*() API. I don't know if comment_buffers etc are
- actually used at all...
-
- * main/config.c: unify some common code
-
- * main/config.c: normalize formatting
-
- * main/config.c: document a nice technique to exit from a block in
- case of errors.
-
- * main/config.c: a little bit of documentation on how lines are
- parsed.
-
- * utils/ael_main.c: normalize header order, and add a comment on
- the need to clean up this file.
-
- * include/asterisk/network.h: some platforms (e.g. FreeBSD4) need
- netinet/in.h to be included before arpa/inet.h
-
-2007-12-07 23:32 +0000 [r91832-91891] Jason Parker <jparker@digium.com>
-
- * /, main/dsp.c: Merged revisions 91890 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11273) ........ r91890 | qwell | 2007-12-07 17:29:01 -0600
- (Fri, 07 Dec 2007) | 4 lines We need to make sure we free the
- input frame if we return a different frame in ast_dsp_process.
- Issue 11273, pointed out by dimas, with a patch by eliel.
- ........
-
- * pbx/pbx_lua.c, configs/extensions.lua.sample: Update
- documentation for pbx_lua. Closes issue #11492, patch by
- mnicholson.
-
-2007-12-07 21:25 +0000 [r91784-91831] Russell Bryant <russell@digium.com>
-
- * /, main/utils.c: Merged revisions 91830 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91830 | russell | 2007-12-07 15:24:33 -0600 (Fri, 07 Dec 2007) |
- 5 lines Make the lock protecting each thread's list of locks it
- currently holds recursive. I think that this will fix the
- situation where some people have said that "core show locks"
- locks up the CLI. (related to issue #11080) ........
-
- * /, include/asterisk/lock.h: Merged revisions 91828 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r91828 | russell | 2007-12-07 15:17:24 -0600 (Fri, 07
- Dec 2007) | 6 lines Fix another bug in the DEBUG_THREADS code.
- The ast_mutex_init() function had the mutex attribute object
- marked as static. This means that multiple threads initializing
- locks at the same time could step on each other and end up with
- improperly initialized locks. (found when tracking down locking
- issues related to issue #11080) ........
-
- * /, include/asterisk/lock.h: Merged revisions 91826 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r91826 | russell | 2007-12-07 15:11:08 -0600 (Fri, 07
- Dec 2007) | 6 lines I love fixing lock related errors in the lock
- debugging code. That's about as ironic as it gets in Asterisk
- programming land. Anyway, I spotted this bug while trying to
- track down why systems are locking up and acting weird in issue
- #11080. The mutex attribute object was marked as static in this
- function when it should not have been. ........
-
- * apps/app_dial.c, /: Merged revisions 91783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) |
- 6 lines * Add channel locking around datastore operations that
- expect the channel to be locked. * Document why we don't record
- Local channels in the dialed interfaces list. * Remove the dialed
- variable as it isn't needed. * Restructure some code for clarity
- and coding guidelines stuff ........
-
-2007-12-07 16:37 +0000 [r91782] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Fix a small typo in a comment. Closes issue
- #11490
-
-2007-12-07 16:28 +0000 [r91781] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 91780 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) |
- 7 lines * Add channel locking around datastore operations that
- expect the channel to be locked. * Document why we don't record
- Local channels in the dialed interfaces list. * Handle memory
- allocation failure. * Remove the dialed variable, as it wasn't
- actually needed. * Tweak some formatting to conform to coding
- guidelines. ........
-
-2007-12-07 16:11 +0000 [r91779] Jason Parker <jparker@digium.com>
-
- * doc/asterisk-mib.txt, main/pbx.c, res/snmp/agent.c,
- include/asterisk/pbx.h, main/cli.c: Add count of total number of
- calls processed by asterisk during it's lifetime. Add number of
- total calls and current calls to SNMP. Closes issue #10057, patch
- by jcmoore.
-
-2007-12-07 16:11 +0000 [r91778] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 91777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91777 | russell | 2007-12-07 10:08:35 -0600 (Fri, 07 Dec 2007) |
- 6 lines * Add a bit more of a verbose comment as to why a hangup
- frame needs to be queued up if autoservice gets a NULL return
- from ast_read(). * Make the process of queueing the hangup frame
- more efficient by putting the frame where it is going to end up
- and avoiding some locking and extra memory allocations and
- freeing. ........
-
-2007-12-07 15:40 +0000 [r91738] Mark Michelson <mmichelson@digium.com>
-
- * main/autoservice.c, /: Merged revisions 91737 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91737 | mmichelson | 2007-12-07 09:39:58 -0600 (Fri, 07 Dec
- 2007) | 7 lines Hangups that happen during autoservice were not
- processed appropriately. This is because a hangup actually causes
- a NULL frame to be received, not a hangup frame. Queueing a
- hangup if we receive a NULL frame during autoservice corrects
- this problem (closes issue #11467, reported by jmls, patched by
- me) ........
-
-2007-12-07 02:52 +0000 [r91676-91700] Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 91693 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) |
- 2 lines Don't unlock the dialed_interfaces list until we're done
- messing with the iterator. ........
-
- * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 91677 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) |
- 4 lines Allow dialing local channels from Queue() and Dial()
- again. There was a slight flaw in the code to prevent call
- forwards from looping that caused this problem. (related to issue
- #11486) ........
-
- * /, apps/app_queue.c: Merged revisions 91675 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) |
- 7 lines Fix in an issue in the call forwarding handling code that
- was causing crashes on every call into a queue. I'm not entirely
- sure about the logic in this part of the code, so I want to look
- at it some more tomorrow. However, this makes it safe and keeps
- it from crashing. (closes issue #11486, reported by adamg,
- patched by me) ........
-
-2007-12-07 00:58 +0000 [r91617-91638] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/rtp.c: Merged revisions 91637 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91637 | tilghman | 2007-12-06 18:52:17 -0600 (Thu, 06 Dec 2007)
- | 5 lines At the end of a call, when we're reporting, RTCP may
- already be partially torn down, so check for NULL dereference
- Reported by: blitzrage Patch by: tilghman (Closes issue #11450)
- ........
-
- * channels/chan_zap.c: Add a manager event for PRI events: this
- will help manager users detect when a D-channel goes down
-
- * main/cdr.c: If duration or billsec are not yet calculated,
- calculate them on demand.
-
-2007-12-06 21:57 +0000 [r91598] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_sqlite3_custom.c: Fix a problem with quoting in sqlite3
- cdr module.. Closes issue #11070, patch by seanbright.
-
-2007-12-06 21:03 +0000 [r91579] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Handle allocation failure of the heard and
- deleted arrays of the vm_state. (closes issue #11408, reported
- and patched by jaroth)
-
-2007-12-06 20:52 +0000 [r91561] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 90166,90736,90753 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90166 | tilghman | 2007-11-29 13:48:10 -0600 (Thu, 29 Nov 2007)
- | 3 lines Properly escape cdr->src and cdr->dst and ensure we use
- thread-safe escaping (Fixes AST-2007-026) ........ r90736 |
- tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines
- If both dbhost and dbsock were not set, a NULL deref could result
- Reported by: xrg Patch by: tilghman (Closes issue #11387)
- ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03
- Dec 2007) | 5 lines Solaris requires the inclusion of
- sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
- snuffy,tilghman (Closes issue #11430) ........
-
-2007-12-06 16:54 +0000 [r91472] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we clear these flags when libpri
- is not installed
-
-2007-12-06 16:51 +0000 [r91440-91458] Joshua Colp <jcolp@digium.com>
-
- * main/udptl.c, /: Merged revisions 91450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91450 | file | 2007-12-06 12:49:42 -0400 (Thu, 06 Dec 2007) | 6
- lines Fix various in the udptl implementation. It could return
- empty modem frames, have an incorrect sequence number on packets,
- and display the wrong sequence number in the debug messages.
- (closes issue #11228) Reported by: Cache Patches: udptl-4.patch
- uploaded by dimas (license 88) ........
-
- * /, channels/chan_sip.c: Merged revisions 91439 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4
- lines Add support for accepting and sending T.38 in the initial
- INVITE. (closes issue #9402) Reported by: thdei ........
-
-2007-12-06 15:56 +0000 [r91347-91438] Olle Johansson <oej@edvina.net>
-
- * doc/manager_1_1.txt (added), UPGRADE.txt: Adding documentation
- for the massive manager changes to manager version 1.1 -
- hopefully a more consistent manager interface.
-
- * main/manager.c: - The Ping Action - Now use Response: success -
- New header "Ping: pong" :-) - The Events action - Now use
- Response: Success - The new status is reported as "Events: On" or
- "Events: Off" - Report if manager is enabled in the reload event
- Small cleanups... From moremanager
-
- * main/channel.c: Changes to manager events in channel.c - Newstate
- event - Now has "CalleridNum" for numeric caller id, like
- Newchannel - The event does not send "<unknown>" for unknown
- caller IDs just an empty field - Newstate and Newchannel events -
- these have changed headers "State" -> ChannelStateDesc Text based
- channel state -> ChannelState Numeric channel state - The events
- does not send "<unknown>" for unknown caller IDs just an empty
- field - Newstate event - Now has "CalleridNum" for numeric caller
- id, like Newchannel - The event does not send "<unknown>" for
- unknown caller IDs just an empty field - Link and Unlink events -
- The "Link" and "Unlink" bridge events in channel.c are now
- renamed to "Bridge" - The link state is in the bridgestate:
- header as "Link" or "Unlink" - For channel.c bridges,
- "Bridgetype: core" is added. This opens up for bridge events in
- rtp.c and channel drivers - The "Rename" manager event has a
- renamed header, to use the same terminology for the current
- channel as other events - Oldname -> Channel (Moremanager)
-
- * main/cdr.c: New manager event when a channel changes account
- code. Maybe belongs in the new cdr category? ---moremanager---
- Event: NewAccountCode Modules: cdr.c Purpose: To report a change
- in account code for a live channel Example: Event: NewAccountCode
- Privilege: call,all Channel: SIP/olle-01844600 Uniqueid:
- 1177530895.2 AccountCode: Stinas account 1234848484
- OldAccountCode: Olles Account 12345
-
- * apps/app_dial.c: - Dial event - Event Dial has new headers, to
- comply with other events - Source -> Channel Channel name
- (caller) - SrcUniqueID -> UniqueID Uniqueid (new) -> Dialstring
- Dialstring in app data (moremanager)
-
- * apps/app_meetme.c: Adding small missing but important comma...
-
- * apps/app_meetme.c: A big oops...
-
- * apps/app_meetme.c: The MeetmeJoin now has caller ID name and
- Caller ID number fields (like MeetMeLeave) (Moremanager)
-
- * channels/chan_zap.c: Update ZapShowChannels so that you can
- specify one channel. Action ZapShowChannels Header changes -
- Channel: -> ZapChannel For active channels, the Channel: and
- Uniqueid: headers are added You can now add a "ZapChannel: "
- argument to zapshowchannels actions to only get information about
- one channel. From the moremanager branch
-
- * main/logger.c: Doxygen updates
-
- * include/asterisk/logger.h, /, main/logger.c, main/loader.c:
- Merged revisions 91366 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91366 | oej | 2007-12-06 13:54:11 +0100 (Tor, 06 Dec 2007) | 4
- lines Make sure logger is reloaded at general reload in the cli.
- (Discovered during Asterisk training in Portugal) ........
-
- * main/manager.c: Change description of new manager command
-
- * main/manager.c, CHANGES: Add manager command for showing all
- current channels. Thanks, eliel, for writing the original patch.
- Modified by me to follow other manager events and the new
- "moremanager" style. (closes issue #11478) Reported by: eliel
- Patches: manager.c.patch uploaded by eliel (license 64)
-
-2007-12-06 04:37 +0000 [r91328] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Instead of iterating through the entire epoll
- events array just look at the ones that will actually contain
- data. (props to eliel on IRC for this)
-
-2007-12-05 22:57 +0000 [r91291-91293] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 91292 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91292 | mmichelson | 2007-12-05 16:57:13 -0600 (Wed, 05 Dec
- 2007) | 3 lines Reverting extra stuff I didn't mean to commit
- ........
-
- * apps/app_dial.c, /, apps/app_voicemail.c: Merged revisions 91273
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec
- 2007) | 11 lines The 'G' option for Dial() did not properly
- handle the case where only a label was provided. This was due to
- the fact that the answering channel did not have an extension
- set, so ast_parseable_goto would fail. This fix eliminates the
- call to ast_parseable_goto on the answering channel since it is a
- wasteful call. The answering channel and the calling channel are
- both directed to the same extension and context, just different
- priorities, so we can just copy the values from the calling
- channel to the answering channel and increment the answering
- channel's priority. (closes issue #11382, reported by jon, patch
- by me with correction by jon) ........
-
-2007-12-05 21:46 +0000 [r91238] Tilghman Lesher <tlesher@digium.com>
-
- * /, sounds/Makefile: Merged revisions 91237 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91237 | tilghman | 2007-12-05 15:38:13 -0600 (Wed, 05 Dec 2007)
- | 2 lines Upgrade to the latest version of extra sounds ........
-
-2007-12-05 17:49 +0000 [r91193-91197] Russell Bryant <russell@digium.com>
-
- * /, main/threadstorage.c: Merged revisions 91192 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91192 | russell | 2007-12-05 11:31:42 -0600 (Wed, 05 Dec 2007) |
- 10 lines Make the lock in the threadstorage debugging code
- untracked to avoid a deadlock on thread destruction. (closes
- issue #11207) Reported by: ys Patches: threadstorage.c.diff
- uploaded by ys (license 281) Also fixes an open bug report:
- (closes issue #11446) ........
-
- * apps/app_directory.c: Resolve compiler warnings.
-
-2007-12-05 16:46 +0000 [r91172-91173] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, UPGRADE.txt, configs/manager.conf.sample,
- CHANGES, include/asterisk/manager.h, cdr/cdr_manager.c: Change
- cdr_manager to use a "CDR" level, rather than the (overcrowded)
- "call" level. (Closes issue #11015)
-
- * CHANGES, apps/app_directory.c: Added multiple name listing.
- (Closes issue #10413)
-
-2007-12-05 16:14 +0000 [r91171] Joshua Colp <jcolp@digium.com>
-
- * configs/http.conf.sample: Remove second prefix line. Only need it
- documented once in the same file. (closes issue #11472) Reported
- by: eserra Patches: http.conf.sample.diff uploaded by eserra
- (license 45)
-
-2007-12-05 13:09 +0000 [r91151-91152] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample: Rename
- "username" to "defaultuser" to match with "defaultip". "Username"
- still works, but is deprecated.
-
- * channels/chan_sip.c: Remove the cseqs from "sip show channel" and
- make more place for the call ID.
-
-2007-12-05 03:48 +0000 [r91133] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: revert part of my changes from earlier today
- since this code is no longer dependent on libpri.h
-
-2007-12-05 03:34 +0000 [r91029-91131] Russell Bryant <russell@digium.com>
-
- * res/res_odbc.c: Use ast_free() instead of free(). (closes issue
- #11309) Reported by: Laureano Patches: res_odbc.c.patch uploaded
- by Laureano (license 265)
-
- * /, include/asterisk/lock.h: Merged revisions 91070 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r91070 | russell | 2007-12-04 18:35:31 -0600 (Tue, 04
- Dec 2007) | 11 lines Fix some crashes in chan_iax2 that were
- reported as happening on Mac systems. It turns out that the
- problem was the Mac version of the ast_atomic_fetchadd_int()
- function. The Mac atomic add function returns the _new_ value,
- while this function is supposed to return the old value. So, the
- crashes happened on unreferencing objects. If the reference count
- was decreased to 1, ao2_ref() thought that it had been decreased
- to zero, and called the destructor. However, there was still an
- outstanding reference around. (closes issue #11176) (closes issue
- #11289) ........
-
- * /, main/utils.c: Merged revisions 91074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r91074 | russell | 2007-12-04 18:48:47 -0600 (Tue, 04 Dec 2007) |
- 4 lines When DEBUG_THREADS is enabled, we only have the details
- about who is holding a lock that we are waiting on for a mutex,
- not rwlocks. This should fix the problem where people have
- reported "core show locks" crashing sometimes. ........
-
- * channels/chan_zap.c: Fix mwimonitornotify on reload ... again.
- This option was only read at startup so a reload would erase it
- and not reset it. (pointed out by tzafrir)
-
- * utils/astman.c: Fix the build of astman. Any file that includes
- any asterisk sub-headers needs to first include asterisk.h.
- (closes issue #11394)
-
-2007-12-04 22:44 +0000 [r91012] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Don't error when we don't have libpri
- installed with libss7 support. Also, print the debug message
- anyway if we can't find the right PRI
-
-2007-12-04 22:07 +0000 [r91010-91011] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /: Merged revisions 90967 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) |
- 7 lines Make some changes to some additions I made recently for
- doing channel autoservice when looking up extensions. This code
- was added to handle the case where a dialplan switch was in use
- that could block for a long time. However, the way that I added
- it, it did this for all extension lookups. However, lookups in
- the in-memory tree of extensions should _not_ take long enough to
- matter. So, move the autoservice stuff to be only around
- executing a switch. ........
-
- * channels/chan_zap.c: Fix resetting mwimonitornotify on reload. I
- guess I only added this line in my head. (thanks to tzafrir for
- pointing it out)
-
-2007-12-04 21:46 +0000 [r90993] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_usbradio.c: Coding guidelines fixups (Closes issue
- #11412)
-
-2007-12-04 21:23 +0000 [r90991] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c, CHANGES: Add manager action
- 'sipshowregistry'. Closes issue #11464, patch by eliel.
-
-2007-12-04 19:08 +0000 [r90949] Russell Bryant <russell@digium.com>
-
- * include/asterisk/callerid.h, channels/chan_zap.c,
- main/callerid.c, CHANGES, configs/zapata.conf.sample: Add support
- for monitoring MWI on FXO lines. This introduces two new options
- for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor
- option enables MWI monitoring. When the MWI state on a line
- changes, then the script specified by mwimonitornotify will be
- executed for custom handling of the state change, similar to the
- externnotify option of voicemail.conf. Also, when the MWI state
- on an FXO line changes, an internal Asterisk event is generated
- to indicate the new state of the associated mailbox. That may,
- any module that cares about MWI information will get notified and
- can handle it just as if app_voicemail had sent this
- notification. (BE-253, original patch from markster, with some
- minor modifications by me to add comments, documentation, and
- internal event support)
-
-2007-12-04 18:29 +0000 [r90930] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Kevin suggested doing the reverse of my
- last commit, since imap_retrieve_file does not modify the
- contents of the "mailbox" string. In other words, I'm changing
- the imap_retrieve_file function to take a const char* as the
- third argument so that I don't need to cast const char*'s as
- char*'s to suppress compiler warnings.
-
-2007-12-04 18:15 +0000 [r90929] Jason Parker <jparker@digium.com>
-
- * Makefile: Add Makefile alias target 'pdf' which does the same
- thing as asterisk.pdf. Issue 11452, reported by blitzrage.
-
-2007-12-04 18:14 +0000 [r90928] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Suppress a compiler warning due to
- discarding a "const" qualifier
-
-2007-12-04 18:09 +0000 [r90927] Jason Parker <jparker@digium.com>
-
- * main/global_datastores.c: Fix build, that some people aren't
- seeing for some reason.
-
-2007-12-04 17:51 +0000 [r90899] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Wrong locking style got merged from 1.4 to
- trunk. My mistake.
-
-2007-12-04 17:40 +0000 [r90880] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: fix build of this module when libpri and/or
- libss7 are or are not present
-
-2007-12-04 17:38 +0000 [r90879] Jason Parker <jparker@digium.com>
-
- * main/channel.c, /: Merged revisions 90876 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11454) ........ r90876 | qwell | 2007-12-04 11:28:08 -0600
- (Tue, 04 Dec 2007) | 4 lines If we fail to create a channel after
- allocating a timing fd, we need to make sure to close it. Issue
- 11454, patch by eliel. ........
-
-2007-12-04 17:36 +0000 [r90878] Russell Bryant <russell@digium.com>
-
- * main/Makefile: Fix a silly little typo :)
-
-2007-12-04 17:35 +0000 [r90877] Jason Parker <jparker@digium.com>
-
- * apps/app_dial.c: Fix build in trunk. This was fixed in 1.4, but
- blocked in trunk since this hadn't been merged yet.
-
-2007-12-04 17:08 +0000 [r90873] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, main/global_datastores.c (added),
- channels/chan_local.c, /, main/Makefile,
- include/asterisk/channel.h, include/asterisk/global_datastores.h
- (added), apps/app_queue.c: Merged revisions 90735 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03
- Dec 2007) | 22 lines A big one... This is the merge of the
- forward-loop branch. The main change here is that call-forwards
- can no longer loop. This is accomplished by creating a datastore
- on the calling channel which has a linked list of all devices
- dialed. If a forward happens, then the local channel which is
- created inherits the datastore. If, through this progression of
- forwards and datastore inheritance, a device is attempted to be
- dialed a second time, it will simply be skipped and a warning
- message will be printed to the CLI. After the dialing has been
- completed, the datastore is detached from the channel and
- destroyed. This change also introduces some side effects to the
- code which I shall enumerate here: 1. Datastore inheritance has
- been backported from trunk into 1.4 2. A large chunk of code has
- been removed from app_dial. This chunk is the section of code
- which handles the call forward case after the channel has been
- requested but before it has been called. This was removed because
- call-forwarding still works fine without it, it makes the code
- less error-prone should it need changing, and it made this set of
- changes much less painful to just have the forwarding handled in
- one place in each module. 3. Two new files, global_datastores.h
- and .c have been added. These are necessary since the datastore
- which is attached to the channel may be created and attached in
- either app_dial or app_queue, so they need a common place to find
- the datastore info. This approach was taken in case similar
- datastores are needed in the future, there will be a common place
- to add them. ........
-
-2007-12-04 15:16 +0000 [r90852-90854] Olle Johansson <oej@edvina.net>
-
- * apps/app_queue.c: (closes issue #11431) Reported by: Laureano
- Patches: app_queue.c.patch uploaded by Laureano (license 265)
-
- * main/pbx.c, CHANGES: (closes issue #11422) Reported by: eliel
- Patches: core.show.hint.patch uploaded by eliel (license 64)
-
- * CHANGES: (closes issue #11462) Reported by: eliel Patches:
- CHANGES.patch uploaded by eliel (license 64)
-
-2007-12-04 15:01 +0000 [r90851] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Pass the Asterisk version to AGI scripts as part
- of the initial dump of info Reported by: acunningham Patch by:
- acunningham (Closes issue #11398)
-
-2007-12-04 11:50 +0000 [r90834] Luigi Rizzo <rizzo@icir.org>
-
- * res/Makefile: fix build on cygwin
-
-2007-12-03 23:52 +0000 [r90760] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/compat.h: Merged revisions 90753 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03
- Dec 2007) | 5 lines Solaris requires the inclusion of
- sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
- snuffy,tilghman (Closes issue #11430) ........
-
-2007-12-03 23:49 +0000 [r90746] Steve Murphy <murf@digium.com>
-
- * main/hashtab.c: A small fix from snuffy
-
-2007-12-03 23:48 +0000 [r90738] Jason Parker <jparker@digium.com>
-
- * res/res_monitor.c: Add manager events for when a monitor is
- started or stopped. Closes issue #10191, patch by dgradecak.
-
-2007-12-03 23:29 +0000 [r90737] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c, /: Merged revisions 90736 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03
- Dec 2007) | 5 lines If both dbhost and dbsock were not set, a
- NULL deref could result Reported by: xrg Patch by: tilghman
- (Closes issue #11387) ........
-
-2007-12-03 22:07 +0000 [r90697] Jason Parker <jparker@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 90696 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #11383) ........ r90696 | qwell | 2007-12-03 16:06:36 -0600
- (Mon, 03 Dec 2007) | 4 lines Make sure we always close the
- conference fd if we have an open one. Issue 11383, reported by
- markmhy, patch by eliel. ........
-
-2007-12-03 21:24 +0000 [r90670] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Replacing some calls to free() with
- ast_free(). (closes issue #11448, reported and patched by jaroth)
-
-2007-12-03 21:03 +0000 [r90656] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/agi.h, res/res_agi.c, CHANGES: Add AGI commands
- for speech recognition. These mirror the dialplan applications
- mostly but present the information in a nicer fashion. The SPEECH
- RECOGNIZE command for example will return the results instead of
- having to query the dialplan functions.
-
-2007-12-03 21:00 +0000 [r90644] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_mgcp.c: Merged revisions 90639 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec
- 2007) | 5 lines Changing some bad logic when calculating the
- interdigit timeout. (closes issue #11402, reported and patched by
- eferro) ........
-
-2007-12-03 20:58 +0000 [r90631] Jason Parker <jparker@digium.com>
-
- * /, res/res_features.c: Merged revisions 90607 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
- issue #11436) ........ r90607 | qwell | 2007-12-03 14:51:17 -0600
- (Mon, 03 Dec 2007) | 4 lines Fix crash in ParkAndAnnounce
- application. Issue #11436, reported by lytledd, patch by eliel.
- ........
-
-2007-12-03 20:30 +0000 [r90591] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c: Avoid an additional function call. Reported by:
- eliel Patch by: eliel (Closes issue #11438)
-
-2007-12-03 20:07 +0000 [r90550-90589] Joshua Colp <jcolp@digium.com>
-
- * /, main/rtp.c: Merged revisions 90588 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2
- lines Do not create a smoother for G723.1 frames, they need to be
- left alone to their native 20/24 byte size. ........
-
- * main/channel.c, /, include/asterisk/channel.h, .cleancount:
- Merged revisions 90548 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2
- lines Preserve the indication currently playing on a channel when
- a masquerade operation happens. (issue #BE-88) ........
-
-2007-12-03 16:46 +0000 [r90528] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample: Updating sample queues.conf file to
- show how multiple periodic announcements may be specified since
- this was not documented previously (closes issue #11432, reported
- and patched by Laureano)
-
-2007-12-03 14:14 +0000 [r90508] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Remove the file descriptors from the main poll
- channel when the channel is hung up during the dialing attempt,
- and make sure a channel exists before trying to remove it at the
- end. (closes issue #11441) Reported by: blitzrage
-
-2007-12-02 18:20 +0000 [r90471] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 90470 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) |
- 6 lines The other day when I went through making changes as a
- result of the ao2_link() change, I added some code to set
- pointers to NULL after they were unreferenced. This pointed out
- that in this place, the object was unreferenced before the code
- was done using it. So, move the unref down a little bit. (crash
- reported by jmls on IRC) ........
-
-2007-12-02 09:42 +0000 [r90433] Tilghman Lesher <tlesher@digium.com>
-
- * main/autoservice.c, /: Merged revisions 90432 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007)
- | 7 lines Clarify the return value on autoservice. Specifically,
- if you started autoservice and autoservice was already on, it
- would erroneously return an error. Reported by: adiemus Patch by:
- dimas (Closes issue #11433) ........
-
-2007-12-01 01:37 +0000 [r90410] Jason Parker <jparker@digium.com>
-
- * res/res_adsi.c: Only reload if the config file has changed.
- Closes issue #11281, patch by eliel.
-
-2007-11-30 21:19 +0000 [r90388] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, include/asterisk/app.h,
- include/asterisk/audiohook.h, res/res_features.c,
- include/asterisk/channel.h, main/audiohook.c, apps/app_queue.c,
- configs/features.conf.sample: Adding support for the
- "automixmonitor" dial and queue options. This works in much the
- same way as the automonitor, except that instead of using the
- monitor app, it uses the mixmonitor app. By providing an 'x' or
- 'X' as a dial or queue option, a DTMF sequence may be entered (as
- defined in features.conf) to start the one-touch mixmonitor. This
- patch also introduces some new API calls to the audiohooks code
- for searching for an audiohook by type and for searching for a
- running audiohook by type. Big thanks to joetester for writing
- the initial patch, testing it and patiently waiting for it to be
- committed. (closes issue #10185, reported and patched by
- xmarksthespot)
-
-2007-11-30 19:34 +0000 [r90311-90351] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /, include/asterisk/astobj2.h, apps/app_queue.c,
- channels/chan_iax2.c, main/astobj2.c, main/config.c: Merged
- revisions 90348 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) |
- 8 lines Change the behavior of ao2_link(). Previously, in
- inherited a reference. Now, it automatically increases the
- reference count to reflect the reference that is now held by the
- container. This was done to be more consistent with ao2_unlink(),
- which automatically releases the reference held by the container.
- It also makes it so it is no longer possible for a pointer to be
- invalid after ao2_link() returns. ........
-
- * /, include/asterisk/astobj2.h: Merged revisions 90310 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90310 | russell | 2007-11-30 12:46:46 -0600 (Fri, 30 Nov 2007) |
- 2 lines Add some notes on the behavior of ao2_unlink() after a
- discussion with Tilghman ........
-
-2007-11-30 14:45 +0000 [r90270] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 90269 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6
- lines Fix locking issues under one legged replaces scenarios.
- (closes issue #11420) Reported by: irroot Patches:
- chan_sip_oneleg.patch uploaded by irroot (license 52) ........
-
-2007-11-30 00:16 +0000 [r90164-90232] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_mgcp.c: Merged revisions 90231 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov
- 2007) | 5 lines Clear the DTMF buffer if the call times out.
- (closes issue #11418, reported and patched by eferro) ........
-
- * /, apps/app_queue.c: Merged revisions 90163 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov
- 2007) | 6 lines This patch handles the case where a queue member
- with a negative penalty is added via the manager. If a negative
- value is submitted for a member penalty, we set it to 0. (closes
- issue #11411, reported and patched by Laureano) ........
-
-2007-11-29 19:35 +0000 [r90156-90162] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c, /: Merged revisions 90160 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r90160 | tilghman | 2007-11-29 13:24:11 -0600 (Thu, 29
- Nov 2007) | 2 lines Properly escape input buffers (Fixes
- AST-2007-025) ........
-
- * /, formats/format_wav.c, formats/format_pcm.c,
- formats/format_ogg_vorbis.c, main/file.c,
- include/asterisk/mod_format.h, formats/format_h263.c,
- formats/format_h264.c, formats/format_wav_gsm.c,
- formats/format_g726.c: Merged revisions 90155 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007)
- | 5 lines Use of "private" as a field name in a header file
- messes with C++ projects Reported by: chewbacca Patch by: casper
- (Closes issue #11401) ........
-
- * include/asterisk/lock.h: Fix build of trunk
-
- * /, sounds/Makefile: Merged revisions 90154 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90154 | tilghman | 2007-11-29 11:18:09 -0600 (Thu, 29 Nov 2007)
- | 2 lines Upgrade the core sounds release version ........
-
-2007-11-29 13:38 +0000 [r90149-90150] Kevin P. Fleming <kpfleming@digium.com>
-
- * utils/Makefile, utils/hashtest.c: let's try this again... *all*
- compilation and linking in Asterisk should be done using the
- standard compilation rules, not manually created ones. changing
- hashtest.c to use these rules caused the compiler to notice a
- large number of coding guidelines violations, so those are fixed
- too.
-
- * main/manager.c: restore behavior from the 1.4 branch... manager
- users created via users.conf should default to *all* permissions,
- not none
-
-2007-11-29 00:37 +0000 [r90139-90148] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /, include/asterisk/channel.h,
- funcs/func_callerid.c: Merged revisions 90145 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) |
- 5 lines This set of changes is to make some callerID handling
- thread-safe. The ast_set_callerid() function needed to lock the
- channel. Also, the handlers for the CALLERID() dialplan function
- needed to lock the channel when reading or writing callerid
- values directly on the channel structure. ........
-
- * include/asterisk/file.h, /, main/file.c: Merged revisions 90142
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) |
- 4 lines Merge a change from team/russell/chan_refcount ... This
- makes ast_stopstream() thread-safe. ........
-
- * include/asterisk/audiohook.h: Merge another small doxygen change
- from team/russell/chan_refcount to indicate that a channel
- doesn't need to be locked before calling a certain function.
-
- * include/asterisk/channel.h: Merge some channel.h doxygen updates
- from team/russell/chan_refcount This was mostly to note whether a
- channel needed to be locked or not before calling these
- functions. However, I added some other things, too.
-
-2007-11-28 23:03 +0000 [r90102] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c, apps/app_queue.c: Merged revisions
- 90101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6
- lines Fix a few memory leaks. (closes issue #11405) Reported by:
- eliel Patches: load_realtime.patch uploaded by eliel (license 64)
- ........
-
-2007-11-28 22:44 +0000 [r90100] Kevin P. Fleming <kpfleming@digium.com>
-
- * configs/users.conf.sample, main/manager.c, /: Merged revisions
- 90098 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007)
- | 2 lines it is impossible to set permissions for manager
- accounts created by users.conf (reported internally, patched by
- me) ........
-
-2007-11-28 22:32 +0000 [r90099] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c: file says... compile before you commit!
-
-2007-11-28 22:17 +0000 [r90060-90061] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Removing a pointless check of option_debug
-
- * main/pbx.c, /: Merged revisions 90059 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov
- 2007) | 13 lines Removing some seemingly pointless code. This
- sets a channel variable for every priority executed in the
- dialplan if you have debug set to anything non-zero. This seems
- pointless due to the fact that these channel variables are not
- referenced anywhere else in the code and their names are esoteric
- enough that they would not be practical to reference in the
- dialplan. Plus the fact that this behavior isn't documented
- anywhere means that the change is not likely to cause any
- disruption. If anything, this may actually cause a slight
- performance increase if running with debug on. The motivating
- influence for this code change is the eventwhencalled option for
- queues. If set to vars, all channel variables will be output to
- the manager. These unnecessary channel variables make the output
- a lot more difficult to deal with. ........
-
-2007-11-28 20:33 +0000 [r90039] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2f.c, main/ast_expr2.fl: Made expr2 parser 8-bit
- transparent
-
-2007-11-28 20:27 +0000 [r90038] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, res/res_crypto.c, include/asterisk/cli.h, main/cli.c:
- Remove "old"-style CLI handler, since nothing uses it anymore.
- Closes issue #11403, patch by eliel. This also completes the
- janitor project.
-
-2007-11-28 15:48 +0000 [r89981-89982] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c: Hide CLI commands starting with _ from tab completion
- as was done previously. (closes issue #11395) Reported by: eliel
- Patches: cli.c.patch uploaded by eliel (license 64)
-
- * main/abstract_jb.c, res/res_agi.c: Fix a few log messages.
- (closes issue #11396) Reported by: IgorG Patches: spell.v1.diff
- uploaded by IgorG (license 20)
-
-2007-11-28 00:49 +0000 [r89947] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Merge some little changes from
- team/russell/chan_refcount to help reduce the diff to trunk. This
- just removes some checks on the return value of alloca(), as
- behavior is undefined if it runs out of stack space, and we don't
- check it anywhere else.
-
-2007-11-28 00:47 +0000 [r89946] Mark Michelson <mmichelson@digium.com>
-
- * configs/musiconhold.conf.sample, configs/extconfig.conf.sample,
- res/res_musiconhold.c, CHANGES: Adding support for realtime music
- on hold. The following are the main points: 1. When moh is
- started, we search first in memory to find the class. If we do
- not find it in memory, we search realtime instead. 2. When moh is
- restarted (as in, it had been started on this particular channel,
- stopped, and now we're starting it again), if using the "files"
- mode, then realtime will always be rechecked. If you are using
- other modes, however, we will simply reattach to the external
- running process which was playing moh earlier in the call. This
- is a necessary compromise so that we don't end up with too many
- background processes. 3. musiconhold.conf has a general section
- now. It has one option: cachertclasses. If set to yes, then moh
- classes found in realtime will be added to the in-memory list.
- This has the advantage of not requiring database lookups each
- time moh is started, but it has the disadvantage of not truly
- being realtime. I have tested this for functionality, and it
- passes. I also tested this under valgrind and there are no memory
- problems reported under typical use. Special thanks to Sergee for
- implementing this feature and enduring my complaints on the
- bugtracker! (closes issue #11196, reported and patched by sergee)
-
-2007-11-28 00:24 +0000 [r89840-89915] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 89893 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) |
- 4 lines - update documentation for some of the goto functions to
- note that they handle locking the channel as needed - update
- ast_explicit_goto() to lock the channel as needed ........
-
- * include/asterisk/channel.h: Document that the channel is not
- locked when the send_digit_begin and end callbacks get called.
-
- * main/autoservice.c, /: Merged revisions 89886 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) |
- 2 lines Don't do frame processing if ast_read() returned NULL.
- ........
-
- * channels/chan_iax2.c: Merge changes from
- team/russell/iax2_frame_queue This patch is an optimization for
- chan_iax2. This module is now heavily multi-threaded. However,
- there is still a good number of globally shared resources that
- prevent things from happen asynchronously. One of those things
- was the global IAX frame queue. This queue was used to hold
- frames that have been deferred for transmitting by another
- thread, and frames that may need to get retransmitted. I changed
- the frame queue to be per-call, since almost all of the frame
- queue handling only cares about frames specific to a call number.
-
- * /, apps/app_queue.c: Merged revisions 89844 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) |
- 3 lines Instead of depending on the return value of ast_true(),
- explicitly set the eventwhencalled variable to 1. ........
-
- * main/pbx.c, /: Merged revisions 89839 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) |
- 2 lines Don't start/stop autoservice in pbx_extension_helper()
- unless a channel exists ........
-
-2007-11-27 23:11 +0000 [r89838] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 89837 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov
- 2007) | 12 lines Two changes with regards to the
- 'eventwhencalled' option of queues.conf 1) Due to some signed vs.
- unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
- did exactly the same thing. Thus the sign change of the ast_true
- call. 2) The vars2manager function overwrote a \n for every
- channel variable it parsed, resulting in bizarre output for the
- channel variables. This patch remedies this. (related to issue
- #11385, however I'm not sure if this will actually be enough to
- close it) ........
-
-2007-11-27 22:42 +0000 [r89835] Russell Bryant <russell@digium.com>
-
- * channels/chan_misdn.c: Bring in a small change from
- team/russell/chan_refcount This replaces tab completion code with
- the use of a public function that does the same thing
-
-2007-11-27 22:14 +0000 [r89792] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, pbx/pbx_config.c: closes issue #11294; missed the
- conditional unlock of the contexts when the hash table is used
- instead; also, used the ast_free_ptr as advised.
-
-2007-11-27 22:05 +0000 [r89791] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, main/pbx.c, /: Merged revisions 89790 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) |
- 41 lines Merge changes from team/russell/autoservice_1.4 This set
- of changes fixes an issue that was reported to me on IRC
- yesterday. The user, d1mas, was using chan_zap for incoming calls
- and was having DTMF recognition issues in some situations.
- Specifically, he noticed that the problem occurred when using
- DISA or WaitExten. He also noticed that when using Read, the
- problem did not occur. His system also used DUNDi for dialplan
- lookups. So, he theorized that if the DUNDi lookups blocked for
- some period of time, that audio from the zap channel could get
- lost. If the audio got lost, then it wouldn't be run through the
- DTMF detector, and digits could get lost. He was correct, and the
- following set of changes fixes the problem. However, the changes
- go a little bit further than what was necessary to fix this exact
- problem. 1) I updated pbx_extension_helper() to autoservice the
- associated channel to handle cases where extension lookups may
- take a long time. This would normally be a dialplan switch that
- does some lookup over the network, such as the DUNDi or IAX2
- switches. This ensures that even while a DUNDi lookup is
- blocking, the channel will be continuously serviced. 2) I made a
- change to the autoservice code. This is actually something that
- has bothered me for a long time. When a channel is in
- autoservice, _all_ frames get thrown away. However, some frames
- really shouldn't be thrown away. The most notable examples are
- signalling (CONTROL) frames, and DTMF. So, this patch queues up
- important frames while a channel is in autoservice. When
- autoservice is stopped on the channel, the queued up frames get
- stuck back on the channel so that they can get processed instead
- of thrown away. 3) I made another change to the autoservice code
- to handle the case where autoservice is started on channels
- recursively. Previously, you could call ast_autoservice_start()
- multiple times on a channel, and it would stop the first time
- ast_autoservice_stop() gets called. Now, it will ensure that
- autoservice doesn't actually stop until the final call to
- ast_autoservice_stop(). ........
-
-2007-11-27 21:10 +0000 [r89769-89772] Olle Johansson <oej@edvina.net>
-
- * main/dnsmgr.c, res/res_jabber.c, main/enum.c, main/asterisk.c: A
- few more "moremanager" fixes
-
- * include/asterisk.h, main/asterisk.c, main/loader.c: More
- "moremanager" fixes. Manager commands to check module status.
-
- * include/asterisk/manager.h: More "moremanager" changes - doxygen
- docs and changing manager version (finally) before making more
- dramatic changes.
-
- * channels/chan_iax2.c: More additions from the "moremanager"
- branch, this time for IAX2.
-
-2007-11-27 20:21 +0000 [r89721] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/app.c: Merged revisions 89709 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007)
- | 2 lines on second thought... revert all the other changes i've
- made in app options parsing leaving only one: if an empty
- argument is supplied for an option, set that argument pointer to
- point to an empty string rather than NULL, so that the
- application can do normal checks on it without worrying about it
- being NULL ........
-
-2007-11-27 20:17 +0000 [r89710] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: remove a duplicate manager event
-
-2007-11-27 19:50 +0000 [r89706] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Manager events from the "moremanager"
- branch
-
-2007-11-27 19:47 +0000 [r89704] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/app.c: Merged revisions 89701 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007)
- | 2 lines generate a warning when an application option that
- requires an argument is ignored due to lack of an argument
- ........
-
-2007-11-27 19:45 +0000 [r89698-89702] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Starting to merge changes from the
- "moremanager" branch. Documentation will follow.
-
- * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: The
- following patch with updates for trunk. Works much better in
- trunk. Also by accident fixed a bad typo by a previous committer,
- which actually made video calls not work fully... Merged
- revisions 89630 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12
- lines If we get a codec offer using a well-known payload type,
- but using it for another codec that we don't know, Asterisk did
- not remove that codec from the list. With this patch, we remove
- the codec from audio and video rtp objects and deny it ever
- existed. Thanks to lasse for testing. (closes issue #11376)
- Reported by: lasse Patches: bug11376.txt uploaded by oej (license
- 306) Tested by: lasse ........
-
-2007-11-27 19:12 +0000 [r89683] Jason Parker <jparker@digium.com>
-
- * include/asterisk/strings.h: Add an S_COR macro, which is similar
- to the existing S_OR macro, except with an additional boolean
- arg. A hack such as: foo ? S_OR(bar, "baz") : "baz" becomes:
- S_COR(foo, bar, "baz")
-
-2007-11-27 18:50 +0000 [r89682] Steve Murphy <murf@digium.com>
-
- * res/ael/ael.y, pbx/ael/ael-test/ref.ael-test11,
- pbx/ael/ael-test/ref.ael-test20, pbx/ael/ael-test/ref.ael-test14,
- pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
- pbx/ael/ael-test/ref.ael-test16, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-test19,
- pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c,
- pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
- pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22,
- res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test3,
- pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
- pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex,
- pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8:
- made AEL 8-bit transparent; mainly the lexer was tossing chars
- with the hi-order bit set. Not nice. Also, allow @ in extension
- names, and a backslash, also.
-
-2007-11-27 17:01 +0000 [r89637] Joshua Colp <jcolp@digium.com>
-
- * main/utils.c: Ensure the value returned from ast_random is
- between 0 and RAND_MAX on 64-bit platforms. (closes issue #11348)
- Reported by: sperreault
-
-2007-11-27 16:13 +0000 [r89635] Russell Bryant <russell@digium.com>
-
- * /, configs/voicemail.conf.sample: Merged revisions 89634 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) |
- 3 lines Add a note to the sample voicemail config noting that
- when using IMAP storage, only the first format specified will be
- attached to the message. ........
-
-2007-11-27 15:41 +0000 [r89632] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_env.c: Merged revisions 89631 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007)
- | 3 lines Default result of STAT should be "0" not "". Reported
- via the -users mailing list, fixed by me. ........
-
-2007-11-27 07:36 +0000 [r89625] Olle Johansson <oej@edvina.net>
-
- * /, configs/sip.conf.sample: Merged revisions 89624 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov
- 2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304)
- Reported by: pj ........
-
-2007-11-27 06:47 +0000 [r89623] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, main/cdr.c, /, configs/cdr.conf.sample,
- include/asterisk/cdr.h: Merged revisions 89622 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1
- line closes issue #11379; OK, this is an attempt to make both
- sides happy. To the cdr.conf file, I added the option
- 'unanswered', which defaults to 'no'. In this mode, you will see
- a cdr for a call, whether it was answered or not. The disposition
- will be NO ANSWER or ANSWERED, as appropriate. The src is as
- you'd expect, the destination channel will be one of the channels
- from the Dial() call, usually the last in the list if more than
- one chan was specified. With unanswered set to 'yes', you will
- still see this cdr entry in both cases. But in the case where the
- dial timed out, you will also see a cdr for each line attempted,
- marked NO ANSWER, with no destination channel name. The new
- option defaults to 'no', so you don't see the pesky extra cdr's
- by default, and you will not see the irritating 'not posted'
- messages. ........
-
-2007-11-26 23:15 +0000 [r89617-89621] Mark Michelson <mmichelson@digium.com>
-
- * pbx/ael/ael-test/ael-test19/extensions.ael,
- pbx/ael/ael-test/ael-vtest13/extensions.ael, doc/osp.txt,
- pbx/ael/ael-test/ael-test3/extensions.ael,
- pbx/ael/ael-test/ref.ael-vtest13,
- pbx/ael/ael-test/ael-test7/extensions.ael: Change all instances
- of "CALLERID(number)" to "CALLERID(num)" for consistency's sake
- (closes issue #11381, reported and patched by jon)
-
- * /, apps/app_playback.c: Merged revisions 89618 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov
- 2007) | 7 lines After issuing a "say load new", if a caller hangs
- up during the middle of playback of a number, app_playback will
- continue to try to play the remaining files. With this change, no
- more files will be played back upon hangup. (closes issue #11345,
- reported and patched by IgorG) ........
-
-2007-11-26 22:52 +0000 [r89615] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Update the configure script check for
- libpri to check for the newest function that was just added.
- Cresl1n, please keep this in mind when making these changes to
- libpri or libss7.
-
-2007-11-26 21:23 +0000 [r89613] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, configs/sip.conf.sample: Rename
- "limitonpeer" to "counteronpeer" since the call-limit is
- deprecated. Both still works in this version.
-
-2007-11-26 21:14 +0000 [r89612] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c, /: Merged revisions 89610 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2
- lines Fix issues with async dialing with an application
- executing. The application has to be terminated and control
- returned to the thread before hanging things up. (issue #BE-252)
- ........
-
-2007-11-26 21:12 +0000 [r89606-89611] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Formatting, doxygenification
-
- * channels/chan_sip.c: Formatting changes, cleaning up some code
-
- * include/asterisk/doxyref.h, channels/chan_sip.c: Start using
- Doxygen groupings to group variables and defines.
-
- * apps/app_meetme.c, UPGRADE.txt, CHANGES, main/cli.c: - Mark
- "concise" as deprecated - Restructure other changes to
- UPGRADE.txt and CHANGES We're still looking for scripts that
- replace asterisk -rx "show shannels concise" by using the manager
- interface, but still produces the same output. Anyone?
-
-2007-11-26 18:11 +0000 [r89600-89602] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c, apps/app_queue.c: Perform some module use
- counting audits. This is now done outside the scope of the
- application/dialplan function so they do not need to worry about
- it.
-
- * /, res/res_features.c: Merged revisions 89599 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6
- lines Add module counting removal for error conditions. (closes
- issue #11333) Reported by: Laureano Patches:
- res_features_v2.c.patch uploaded by Laureano (license 265)
- ........
-
-2007-11-26 17:49 +0000 [r89596] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /: Merged revisions 89594 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) |
- 3 lines Add channel locking to a function that needed to be doing
- it. This is just a little something I noticed while working on a
- completely unrelated issue. ........
-
-2007-11-26 17:46 +0000 [r89595] Steve Murphy <murf@digium.com>
-
- * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c: closes
- issue #11341; made changes to make utils again right with the
- MTX_PROFILE world.
-
-2007-11-26 17:38 +0000 [r89593] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 89592 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6
- lines Use ast_free to free memory, or else we shall implode if
- MALLOC_DEBUG is enabled. (closes issue #11347) Reported by: ys
- Patches: pbx.pbx_config.c.diff uploaded by ys (license 281)
- ........
-
-2007-11-26 17:26 +0000 [r89591] Steve Murphy <murf@digium.com>
-
- * main/hashtab.c: closes issue #11356; Many thanks to snuffy for
- his code review and changes to cut down duplication. I tested
- this against hashtest, and it passes. I reviewed the changes, and
- they look reasonable. I had to remove a few const decls to make
- things compile on my workstation,
-
-2007-11-26 17:25 +0000 [r89590] Russell Bryant <russell@digium.com>
-
- * Makefile: make sure we check to see if the configure script has
- been executed on a new checkout or after a distclean
-
-2007-11-26 17:23 +0000 [r89589] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_mixmonitor.c: Merged revisions 89587 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov
- 2007) | 6 lines Close the audio file before sending it to the
- post processing application. (closes issue #11357) Reported by:
- reformed Patches: mixmonitor.patch uploaded by reformed (license
- 330) ........
-
-2007-11-26 17:21 +0000 [r89588] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/app.c, apps/app_controlplayback.c: Merged revisions 89586
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007)
- | 2 lines when parsing application options that take arguments,
- don't indicate that the option was supplied unless a
- non-zero-length argument was found for it ........
-
-2007-11-26 16:24 +0000 [r89583] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, CHANGES, configs/extensions.conf.sample: Thanks to
- pnlarsson for noting the spelling error in the cli commands.
- Also, added some verbage about the new algorithm to CHANGES.
-
-2007-11-26 16:20 +0000 [r89582] Joshua Colp <jcolp@digium.com>
-
- * main/utils.c: Revert change for 11348 until it can be looked at
- even more.
-
-2007-11-26 15:50 +0000 [r89581] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 89580 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov
- 2007) | 6 lines Revert vmu->email back to an empty string if it
- was empty when imap_store_file was called. This prevents sending
- a duplicate e-mail. (closes issue #11204, reported by spditner,
- patched by me) ........
-
-2007-11-26 15:36 +0000 [r89570-89578] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 89577 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6
- lines If channel allocation fails because the alert pipe could
- not be created also free the scheduler context. (closes issue
- #11355) Reported by: eliel Patches: main.channel.c.patch uploaded
- by eliel (license 64) ........
-
- * main/utils.c: Make the behavior of using /dev/urandom for random
- numbers the same as random(). (closes issue #11348) Reported by:
- sperreault Patches: ast_random2.diff uploaded by sperreault
- (license 252)
-
- * channels/chan_sip.c: Instead of printing out one codec in sip
- show channels print out all of the native ones (this is for
- video). (closes issue #11366) Reported by: ovi
-
- * /, apps/app_meetme.c: Merged revisions 89571 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4
- lines When unloading app_meetme destroy any auto created contexts
- created by SLA. (closes issue #11367) Reported by: eliel ........
-
- * apps/app_controlplayback.c: Don't crash if the 'o' option of
- ControlPlayback is used without any value. (closes issue #11375)
- Reported by: johan
-
-2007-11-25 21:12 +0000 [r89564-89566] Olle Johansson <oej@edvina.net>
-
- * channels/chan_usbradio.c: Formatting changes
-
- * main/channel.c, include/asterisk/channel.h: Try to get channel.h
- and channel.c aligned in regards to ast_set_callerid as well as
- change name of variables to follow the rest of the naming.
-
-2007-11-25 17:50 +0000 [r89560-89561] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h, res/res_config_odbc.c, /,
- res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions
- 89559 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007)
- | 14 lines We previously attempted to use the ESCAPE clause to
- set the escape delimiter to a backslash. Unfortunately, this does
- not universally work on all databases, since on databases which
- natively use the backslash as a delimiter, the backslash itself
- needs to be delimited, but on other databases that have no
- delimiter, backslashing the backslash causes an error. So the
- only solution that I can come up with is to create an option in
- res_odbc that explicitly specifies whether or not backslash is a
- native delimiter. If it is, we use it natively; if not, we use
- the ESCAPE clause to make it one. Reported by: elguero Patch by:
- tilghman (Closes issue #11364) ........
-
- * channels/chan_sip.c: Typo (someone needs to test compile before
- committing his changes)
-
-2007-11-25 12:18 +0000 [r89551-89557] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: More doxygen changes
-
- * channels/chan_sip.c: Housekeeping
-
- * channels/chan_sip.c: Formatting, doxygen updates
-
- * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: -
- Deprecate "call-limit" in chan_sip. No other channel driver
- enforces call-limits and we now have the groupcount system to
- implement call-limits in the dialplan. You can use the "setvar"
- option in realtime/sip.conf to set limits per device. - Implement
- "callcounter" as a new option to enable the call counting we need
- to report device status to queue, manager and SIP subscriptions.
- The call counter setting is now enabled in the code by setting
- the device call-limit to 999. When we remove the call limit, we
- can simply enable this with a boolean setting.
-
- * channels/chan_sip.c, include/asterisk/channel.h: Housekeeping...
- - Fix typo in chan_sip - Remove changes to caller ID structure,
- moving it to branch (russellb)
-
-2007-11-24 21:00 +0000 [r89547] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c,
- configs/extensions.conf.sample: closes issue #11363; where the
- pattern _20x. buried in an included context, didn't match 2012;
- There were a small set of problems to fix: 1. I needed NOT to
- score patterns unless you are at the end of the data string. 2.
- Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize
- the patterns in the trie to caps. 3. When a pattern ends with dot
- or exclamation, CANMATCH/MATCHMORE should always report this
- pattern, no matter the length. With this commit, I also supplied
- the wish of Luigi, where the user can select which pattern
- matching algorithm to use, the old (legacy) pattern matcher, or
- the new, trie based matcher. The OLD matcher is the default. A
- new [general] section variable, extenpatternmatchnew, is added to
- the extensions.conf, and the example config has it set to false.
- If true, the new matcher is used. In all other respects, the
- context/exten structs are the same; the tries and hashtabs are
- formed, but in the new mode the tries are not used. A new CLI
- command 'dialplan set extenpatternmatch true/false' is provided
- to allow switching at run time. I beg users that are forced to
- return to the old matcher to please report the reason in the bug
- tracker. Measured the speed benefit of the new matcher against an
- impossibly large context with 10,000 extensions: the new matcher
- is 374 times faster.
-
-2007-11-24 17:07 +0000 [r89546] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_adsi.c: Merged revisions 89545 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89545 | tilghman | 2007-11-24 10:59:59 -0600 (Sat, 24 Nov 2007)
- | 5 lines Free some frames that would otherwise leak on error.
- Reported by: Laureano Patch by: Laureano,tilghman (Closes issue
- #11351) ........
-
-2007-11-24 16:53 +0000 [r89544] Steve Murphy <murf@digium.com>
-
- * main/app.c: Added <sys/file.h> include to allow trunk to compile.
- Hope this doesn't louse thing up.
-
-2007-11-24 13:57 +0000 [r89542-89543] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_h323.c: remove a DEBUG_THREADS message that
- accesses private lock fields. If needed, the code to extract this
- information should be implemented in some generic header or
- library and the function called here. (closed bug #11362)
-
- * main/acl.c, main/http.c, main/app.c: remove some unnecessary
- includes
-
-2007-11-24 06:24 +0000 [r89535-89541] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/app.c, apps/app_voicemail.c: Merged revisions 89540 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007)
- | 9 lines Currently, zero-length voicemail messages cause a
- hangup in VoicemailMain. This change fixes the problem, with a
- multi-faceted approach. First, we do our best to avoid these
- messages from being created in the first place, and second, if
- that fails, we detect when the voicemail message is zero-length
- and avoid exiting at that point. Reported by: dtyoo Patch by:
- gkloepfer,tilghman (Closes issue #11083) ........
-
- * main/manager.c, /: Merged revisions 89536 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89536 | tilghman | 2007-11-23 11:18:26 -0600 (Fri, 23 Nov 2007)
- | 10 lines Up until this point, the XML output of the manager has
- been technically invalid, due to the repetition of certain
- parameters in a single event. This caused various issues for XML
- parsers, some of which refused to parse at all, given the
- invalidity of the rendered XML. So this commit fixes the XML
- output, ensuring that each entity parameter has a unique name,
- thus ensuring valid XML. Reported by: msetim Patch by: tilghman
- (Closes issue #10220) ........
-
- * res/res_config_odbc.c, /: Merged revisions 89534 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89534 | tilghman | 2007-11-23 11:05:10 -0600 (Fri, 23
- Nov 2007) | 5 lines Use ESCAPE clause for the first parameter,
- not just 2nd-Nth parameters. Reported by: apsaras Patch by:
- tilghman (Closes issue #11353) ........
-
-2007-11-23 15:54 +0000 [r89532-89533] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_oss.c: put in the necessary hooks for video support
- in the console. This is a NOP as far as the current code is
- concerned, but there is already support in ./configure and the
- Makefiles for the various libraries used by console_video.c (not
- yet in the tree) so addition is trivial.
-
- * channels/chan_sip.c: set rtpmap video info according to what is
- read from SDP; make the format explicit in a debug message; print
- the audio instead of aggregated peer capability in a debugging
- msg.
-
-2007-11-23 09:40 +0000 [r89531] Olle Johansson <oej@edvina.net>
-
- * include/asterisk/channel.h: Let's start with implementing the
- base architecture for UTF8 caller ID's so we can handle multiple
- formats properly. This is not carved in stone, but a proposal to
- start with. We need to add support for transliterations as well
- as UTF8 handling, propably with libiconv. Murf is looking into
- that for the dialplan.
-
-2007-11-23 09:03 +0000 [r89530] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/image.h, formats/format_jpeg.c: formatting
- cleanup on the header, normalization of the assignment of
- descriptor fields.
-
-2007-11-23 02:37 +0000 [r89529] Russell Bryant <russell@digium.com>
-
- * configs/agents.conf.sample, /: Merged revisions 89527 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89527 | russell | 2007-11-22 12:29:41 -0500 (Thu, 22 Nov 2007) |
- 3 lines mvanbaak pointed out a spelling error in this sample
- configuration file. While I was at it, I went ahead and tweaked
- it a little bit more. ........
-
-2007-11-22 07:10 +0000 [r89514-89526] Luigi Rizzo <rizzo@icir.org>
-
- * doc/CODING-GUIDELINES: new info on the management of headers
-
- * apps/app_echo.c, apps/app_sendtext.c, apps/app_verbose.c,
- apps/app_milliwatt.c: more header removal
-
- * include/asterisk/channel.h: formatting cleanup
-
- * include/asterisk.h, apps/app_read.c, apps/app_record.c,
- apps/app_echo.c, apps/app_readexten.c,
- include/asterisk/channel.h, apps/app_system.c,
- apps/app_transfer.c, res/ael/pval.c, include/asterisk/app.h,
- apps/app_dumpchan.c, include/asterisk/module.h, apps/app_url.c,
- include/asterisk/pbx.h, apps/app_senddtmf.c, pbx/pbx_config.c,
- apps/app_mixmonitor.c, apps/app_stack.c, apps/app_verbose.c,
- apps/app_milliwatt.c, apps/app_cdr.c, apps/app_while.c: shuffle a
- little bit the content of header files to reduce dependencies. In
- this commit: - move the ast_register/unregister_app functions to
- module.h to avoid the need to include pbx.h for the simpler apps;
- - move the ast_group structure to channel.h to remove the
- dependency of app.h on linkedlists.h Note, this is a long process
- that I am doing in small steps. The main difficulty is that now
- for each subsystem we have a single header (e.g. channel.h)
- included by the subsystem provider (usually one file, e.g.
- channel.c) and by its clients (dozens of them, e.g. we have some
- 70+ apps and 30+ functions). This requires the clients to include
- all the extra headers required by the provider (eg. lock.h,
- linkedlists.h, definitions of substructures...) even though many
- of the clients would be just happy with opaque struct
- declarations and function prototypes. The long term plan is to
- eventually rectify this structure so that the compilation can
- become faster, and also APIs are more stable.
-
- * funcs/func_md5.c, funcs/func_module.c, funcs/func_blacklist.c,
- apps/app_url.c, funcs/func_sha1.c, funcs/func_global.c: remove
- some useless includes
-
- * include/asterisk/audiohook.h, apps/app_dictate.c,
- apps/app_readexten.c, apps/app_directory.c, apps/app_senddtmf.c,
- apps/app_mixmonitor.c, apps/app_stack.c,
- apps/app_controlplayback.c: more removal of redundant headers
-
- * apps/app_read.c, apps/app_echo.c, apps/app_record.c,
- apps/app_userevent.c, apps/app_image.c, apps/app_system.c,
- apps/app_verbose.c, apps/app_milliwatt.c, apps/app_playback.c,
- apps/app_while.c: remove redundant headers
-
- * main/file.c, main/netsock.c: more removal of fcntl.h and other
- system headers
-
- * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c,
- codecs/codec_speex.c, codecs/codec_alaw.c, codecs/codec_adpcm.c,
- res/res_crypto.c, codecs/codec_g726.c, apps/app_test.c,
- formats/format_ogg_vorbis.c, codecs/codec_gsm.c, res/res_agi.c,
- apps/app_mp3.c, main/app.c, codecs/codec_ulaw.c,
- codecs/codec_ilbc.c: remove a number of #include <fcntl.h> which
- are either useless or done elsewhere
-
- * formats/format_sln.c, formats/format_wav.c,
- formats/format_ogg_vorbis.c, include/asterisk/_private.h,
- formats/format_wav_gsm.c, formats/format_ilbc.c,
- include/asterisk/file.h, formats/format_vox.c,
- formats/format_pcm.c, main/file.c, formats/format_h263.c,
- formats/format_g723.c, formats/format_h264.c,
- include/asterisk/frame.h, formats/format_jpeg.c,
- formats/format_g726.c, formats/format_gsm.c,
- formats/format_g729.c: implement the split of file.h and
- mod_format.h
-
- * include/asterisk/mod_format.h (added): Add a specific header for
- providers of file and format handling routines, moving here
- structs and function declarations formerly in file.h
-
-2007-11-21 23:54 +0000 [r89513] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, channels/chan_sip.c, channels/chan_skinny.c,
- res/res_features.c, apps/app_queue.c, channels/chan_iax2.c:
- closes issue #11285, where an unload of a module that creates a
- dialplan context, causes a crash when you do a 'dialplan show' of
- that context. This is because the registrar string is defined in
- the module, and the stale pointer is traversed. The reporter
- offered a patch that would always strdup the registrar string,
- which is practical, but I preferred to destroy the created
- contexts in each module where one is created. That seemed more
- symmetric. There were only 6 place in asterisk where this is
- done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial,
- and app_queue. The two apps destroyed the context, but left the
- contexts. All is fixed now and unloads should be dialplan
- friendly.
-
-2007-11-21 23:24 +0000 [r89511-89512] Luigi Rizzo <rizzo@icir.org>
-
- * funcs/func_rand.c, cdr/cdr_sqlite3_custom.c, apps/app_readfile.c,
- channels/chan_local.c, apps/app_record.c, funcs/func_strings.c,
- apps/app_sayunixtime.c, apps/app_test.c,
- apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c,
- apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c,
- channels/chan_iax2.c, apps/app_exec.c, pbx/pbx_loopback.c,
- pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_dumpchan.c,
- apps/app_zapscan.c, apps/app_zapras.c, pbx/pbx_realtime.c,
- channels/chan_alsa.c, apps/app_amd.c, apps/app_url.c,
- apps/app_externalivr.c, cdr/cdr_odbc.c, apps/app_dial.c,
- funcs/func_timeout.c, apps/app_page.c, apps/app_privacy.c,
- channels/chan_agent.c, apps/app_disa.c, apps/app_morsecode.c,
- channels/iax2-provision.c, funcs/func_cut.c,
- apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c,
- apps/app_playback.c, funcs/func_curl.c, channels/chan_misdn.c,
- apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c,
- channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c,
- pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
- apps/app_voicemail.c, channels/chan_unistim.c,
- channels/chan_vpb.cc, apps/app_meetme.c, apps/app_authenticate.c,
- apps/app_readexten.c, funcs/func_vmcount.c,
- channels/chan_gtalk.c, cdr/cdr_pgsql.c, apps/app_followme.c,
- cdr/cdr_radius.c, apps/app_controlplayback.c, cdr/cdr_csv.c,
- channels/chan_phone.c, funcs/func_enum.c, apps/app_osplookup.c,
- funcs/func_odbc.c, apps/app_mp3.c, apps/app_minivm.c,
- apps/app_rpt.c, channels/chan_mgcp.c, apps/app_parkandannounce.c,
- apps/app_while.c, apps/app_adsiprog.c, apps/app_nbscat.c,
- funcs/func_version.c, funcs/func_db.c, channels/chan_zap.c,
- apps/app_read.c, channels/chan_sip.c, apps/app_festival.c,
- apps/app_waitforsilence.c, funcs/func_lock.c, pbx/pbx_lua.c,
- apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c,
- channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c,
- channels/chan_jingle.c, channels/chan_usbradio.c,
- apps/app_channelredirect.c, apps/app_flash.c,
- apps/app_directed_pickup.c, funcs/func_blacklist.c,
- channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_sms.c,
- channels/chan_nbs.c, apps/app_senddtmf.c, funcs/func_callerid.c,
- apps/app_verbose.c, apps/app_stack.c, pbx/pbx_gtkconsole.c:
- remove another set of redundant #include "asterisk/options.h"
-
- * main/udptl.c, main/autoservice.c, main/frame.c, res/res_snmp.c,
- main/say.c, res/res_features.c, main/devicestate.c, main/utils.c,
- res/res_musiconhold.c, res/res_jabber.c, main/indications.c,
- main/enum.c, res/res_config_sqlite.c, main/config.c,
- main/loader.c, main/term.c, main/cli.c, main/io.c,
- main/channel.c, main/cdr.c, main/dial.c, res/res_smdi.c,
- res/res_config_odbc.c, main/manager.c, res/res_agi.c,
- main/http.c, main/logger.c, res/res_realtime.c, main/app.c,
- main/image.c, main/dns.c, main/db.c, res/res_speech.c,
- main/sched.c, main/pbx.c, res/res_config_pgsql.c, main/dnsmgr.c,
- main/translate.c, res/res_crypto.c, res/res_adsi.c,
- main/jitterbuf.c, main/acl.c, formats/format_ogg_vorbis.c,
- res/res_ael_share.c, res/res_monitor.c, main/rtp.c,
- main/netsock.c, main/srv.c, main/hashtab.c, main/privacy.c,
- main/adsistub.c, main/abstract_jb.c, main/file.c,
- main/callerid.c, main/astmm.c, main/audiohook.c,
- formats/format_g726.c, main/asterisk.c, res/res_odbc.c,
- main/dsp.c: remove a bunch of useless #include "options.h"
-
-2007-11-21 22:37 +0000 [r89509-89510] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Remove unneccessary explicit case for BRI
-
- * channels/chan_zap.c: Take some debug code out :-)
-
-2007-11-21 22:20 +0000 [r89508] Luigi Rizzo <rizzo@icir.org>
-
- * main/cygload.c: add a missing return
-
-2007-11-21 22:07 +0000 [r89507] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add BRI support to chan_zap
-
-2007-11-21 21:30 +0000 [r89506] Luigi Rizzo <rizzo@icir.org>
-
- * utils/Makefile, configure, configure.ac: enable support for stack
- backtrace for stuff built in utils/ (this was present in the main
- tree but forgotten here).
-
-2007-11-21 20:38 +0000 [r89505] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: closes issue #11290; the proposed patch was a good
- guess, and would solve the bug to some extent, but was really
- masking the real issue, that there were bad entries in the table.
- This fix removes the condition that the hashtab updates be done
- on exten removal only when the pattern_tree was present, which is
- silly. The operations that apply to the pattern tree are instead
- made conditional. Also, threw back in routines that kpfleming
- deleted because of probs in the 64-bit world. Tested on both 32
- and 64-bit machines (compile). Tested the reload problem with
- over 20 reloads, and no problems. If you find more problems,
- please reopen 11290.
-
-2007-11-21 20:22 +0000 [r89504] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: Simplify comparison in parking fix
-
-2007-11-21 19:28 +0000 [r89494-89496] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 89495 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89495 | mmichelson | 2007-11-21 13:27:51 -0600 (Wed, 21 Nov
- 2007) | 3 lines Fix a small error I made in my previous commit
- ........
-
- * /, apps/app_queue.c: Merged revisions 89493 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov
- 2007) | 5 lines Changing an inaccurate debug message to be less
- inaccurate. Under the circumstances, this message would always
- report that there were 0 members available, even though that may
- not be true. ........
-
-2007-11-21 19:20 +0000 [r89492] Terry Wilson <twilson@digium.com>
-
- * /, res/res_features.c: Merged revisions 89491 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89491 | twilson | 2007-11-21 12:59:27 -0600 (Wed, 21 Nov 2007) |
- 4 lines If a channel gets masqueraded in the middle of a park,
- don't play the announcement to the masqueraded channel, and dial
- back to the original channel on timeout. ........
-
-2007-11-21 18:52 +0000 [r89490] Russell Bryant <russell@digium.com>
-
- * main/dsp.c: Remove obsolete OLD_DSP_ROUTINES code. Also, remove
- the FAX_DETECT define and only do the calculations if fax
- detection is enabled on the dsp. (closes issue #11331) Reported
- by: dimas Patches: dsp.patch uploaded by dimas (license 88)
-
-2007-11-21 18:38 +0000 [r89489] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_read.c, UPGRADE.txt, CHANGES: Change Read to set
- READSTATUS as an indication of the result Also, some cleanup to
- CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman
- (Closes issue #11004)
-
-2007-11-21 18:24 +0000 [r89488] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: fix a small gramatical error in a comment
-
-2007-11-21 18:19 +0000 [r89487] Mark Michelson <mmichelson@digium.com>
-
- * main/utils.c: There existed about a 1 in 4 billion chance that
- reading from /dev/urandom would return LONG_MIN (1 in 9
- quintillion if using 64-bit longs). Since there is no positive
- equivalent of LONG_MIN, the result of labs() in this case is
- unpredictable. This fixes that situation. (closes issue #11336,
- reported and patched by sperreault)
-
-2007-11-21 16:24 +0000 [r89484] Russell Bryant <russell@digium.com>
-
- * channels/chan_unistim.c: Fix some code that was supposed to
- ensure that a buffer was terminated, but was writing to the wrong
- byte. Also, remove some non-thread safe test code. (closes issue
- #11317) Reported by: IgorG Patches: unistim-2.patch uploaded by
- IgorG (license 20) - additional changes by me
-
-2007-11-21 16:08 +0000 [r89483] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: I introduced a deadlock avoidance into 1.4, which I
- attempted to port to trunk as well. Unfortunately, since trunk
- uses read/write locks for the context lock, it means that I have
- actually *introduced* a deadlock condition since they are not
- recursive. Removing this change for now and will look into
- introducing a different one.
-
-2007-11-21 16:07 +0000 [r89480-89482] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk.h, include/asterisk/compat.h, utils/ael_main.c,
- utils/conf2ael.c: move these forward declarations back to
- asterisk.h where they belong... even though asterisk.h includes
- compat.h, these declarations have nothing to do with the being
- platform-compatible and are directly related to being part of
- Asterisk
-
- * channels/chan_usbradio.c: get this to actually compile...
-
- * main/pbx.c: remove some debugging code that doesn't compile on
- 64-bit platforms
-
-2007-11-21 15:17 +0000 [r89478-89479] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: OOOps! All the debug stuff I inserted was
- accidentally committed. I hereby revert it.
-
- * main/hashtab.c, res/res_features.c: closes issue #11265; Thanks
- to snuffy for his work on neatening up the code and removing
- duplicated code.
-
-2007-11-21 08:28 +0000 [r89475-89477] Luigi Rizzo <rizzo@icir.org>
-
- * channels/gentone-ulaw.c (removed): remove this file, it is not
- used anywhere.
-
- * main/astmm.c: add missing paths.h
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: add
- check for video4linux
-
-2007-11-21 01:09 +0000 [r89474] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: A free in add_pri was ultimately the source of the
- grief we were having with parking. This set of changes fixes that
- problem, and introduces some more error messages, and puts debugs
- into ifdefs for what could be short-term usage. Txs to Terry W.
- for his help, guidance, and especially patience.
-
-2007-11-21 00:23 +0000 [r89472-89473] Luigi Rizzo <rizzo@icir.org>
-
- * main/sha1.c, agi/eagi-test.c, utils/smsq.c, utils/hashtest2.c,
- main/minimime/mm.h, utils/check_expr.c: more header
- removal/normalization
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: X11
- checks (at least some - for other platforms with unusual X11
- locations you might need to add more directories)
-
-2007-11-21 00:21 +0000 [r89470] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c, CHANGES: Merge changes from
- team/russell/sla_trunk_moh ... * Added the ability to specify the
- music on hold class used to play into the conference when there
- is only one member and the M option is used. * Added the ability
- to specify a music on hold class to play instead of ringing for
- the SLATrunk application. (patched by me, and tested internally)
-
-2007-11-21 00:20 +0000 [r89469] Luigi Rizzo <rizzo@icir.org>
-
- * makeopts.in: complete support for X11
-
-2007-11-20 23:29 +0000 [r89467-89468] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_meetme.c, cdr/cdr_sqlite.c, pbx/pbx_lua.c: Make trunk
- build again
-
- * main/say.c: Add support for new recorded character sounds Closes
- issue #5208
-
-2007-11-20 23:16 +0000 [r89465-89466] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c,
- apps/app_dictate.c, apps/app_test.c, apps/app_ices.c,
- apps/app_followme.c, channels/chan_iax2.c, main/config.c,
- main/loader.c, main/cli.c, cdr/cdr_csv.c, main/channel.c,
- main/manager.c, pbx/pbx_spool.c, include/asterisk/compat.h,
- res/res_agi.c, apps/app_minivm.c, main/logger.c, main/http.c,
- main/app.c, main/image.c, apps/app_directory.c, main/db.c,
- cdr/cdr_custom.c, apps/app_adsiprog.c, apps/app_dial.c,
- include/asterisk/utils.h, include/asterisk.h, main/pbx.c,
- channels/chan_sip.c, res/res_crypto.c,
- include/asterisk/channel.h, res/res_monitor.c,
- include/asterisk/paths.h, main/file.c, apps/app_sms.c,
- include/asterisk/ael_structs.h, pbx/pbx_config.c,
- apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c:
- move asterisk/paths.h outside asterisk.h and into those files who
- really need it.
-
- * main/pbx.c, include/asterisk.h, main/frame.c, main/dnsmgr.c,
- main/threadstorage.c, main/devicestate.c,
- include/asterisk/_private.h (added), main/astobj2.c,
- main/loader.c, main/term.c, main/cli.c, main/channel.c,
- main/manager.c, main/logger.c, build_tools/strip_nonapi,
- main/event.c, main/asterisk.c, main/db.c: move internal function
- declarations to include/asterisk/_private.h
-
-2007-11-20 19:29 +0000 [r89464] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: i got a little carried away with commas
- ...
-
-2007-11-20 19:28 +0000 [r89463] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/module.h, build_tools/make_buildopts_h,
- main/loader.c: switch compile-time option checking to string
- storage mode in this branch too
-
-2007-11-20 19:11 +0000 [r89460] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: fix the zaptel configure script check
-
-2007-11-20 18:20 +0000 [r89459] Luigi Rizzo <rizzo@icir.org>
-
- * acinclude.m4: the 'version' is now $7 not $6 (wait a bit before
- regenerating configure, i have more changes)
-
-2007-11-20 17:59 +0000 [r89458] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c, /: Merged revisions 89457 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89457 | mmichelson | 2007-11-20 11:50:31 -0600 (Tue, 20 Nov
- 2007) | 9 lines According to comments in main/pbx.c, it is
- essential that if we are going to lock the conlock as well as the
- hints lock, it must be locked in that respective order. In order
- to prevent a potential deadlock, we need to lock the conlock
- prior to locking the hints lock in ast_hint_state_changed (see
- the call stack example on issue #11323 for how this can happen).
- (closes issue #11323, reported by eelcob, suggestion for patch by
- eelcob, patch by me) ........
-
-2007-11-20 17:11 +0000 [r89454-89455] Luigi Rizzo <rizzo@icir.org>
-
- * makeopts.in: prepare to support console_video
-
- * apps/Makefile, Makefile.moddir_rules, pbx/Makefile, res/Makefile,
- channels/Makefile: Fix building of modules under cygwin. After
- this commit we can actually load modules under windows, and we
- can start debugging more interesting problems related to the load
- order and functionality of modules.
-
-2007-11-20 16:11 +0000 [r89453] Mark Michelson <mmichelson@digium.com>
-
- * configs/sip.conf.sample: Changed occurrences of "busy-level" to
- "busylevel" in sip.conf.sample in light of commit 89441. Thanks
- to pj for pointing out the need for this (closes issue #11307,
- reported by pj)
-
-2007-11-20 15:39 +0000 [r89452] Luigi Rizzo <rizzo@icir.org>
-
- * configure, configure.ac, acinclude.m4: add an argument for extra
- headers to AC_EXT_LIB_CHECK, and on passing simplify the code.
- Too bad that every time we need to regenerate configure...
-
-2007-11-20 15:30 +0000 [r89451] Steve Murphy <murf@digium.com>
-
- * /, doc/tex/queues-with-callback-members.tex: Merged revisions
- 89450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1
- line closes issue #11324; break statements missing in switch
- cases. ........
-
-2007-11-20 15:00 +0000 [r89449] Joshua Colp <jcolp@digium.com>
-
- * main/translate.c: Minor documentation tweak and if an incorrect
- parameter is given to core show translation return the usage
- information. (closes issue #11316) Reported by: eliel Patches:
- translate.c.patch uploaded by eliel (license 64)
-
-2007-11-20 14:54 +0000 [r89448] Luigi Rizzo <rizzo@icir.org>
-
- * configure, acinclude.m4: comment a bit the code in acinclude.m4
- There is still a lot of code to clean up there, but hopefully
- this should clarify what goes on in there.
-
-2007-11-20 14:49 +0000 [r89447] Joshua Colp <jcolp@digium.com>
-
- * channels/h323/ast_h323.cxx: Include the compatibility header file
- in ast_h323.cxx for compatibility reasons. (closes issue #11311)
- Reported by: falves11
-
-2007-11-20 14:44 +0000 [r89444-89446] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Fix sip show history. Closes issue #11312
-
- * channels/chan_sip.c: Change terminology a bit for CLI commands
- handling SIP channels/calls/dialogs/whatever. Closes issue #11312
-
-2007-11-20 07:42 +0000 [r89443] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile, main/Makefile, Makefile.moddir_rules: initial makefile
- changes to build loadable modules under cygwin (not complete yet
- - still need to sort out dependecies on res_*)
-
-2007-11-20 00:17 +0000 [r89442] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: Get rid of some debug messages in pbx.c
-
-2007-11-19 23:24 +0000 [r89441] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c, CHANGES: Changed the "busy-level" option in
- sip.conf to "busylevel" to be more parallel with the SIPPEER()
- argument of the same name. The deprecation procedure is not being
- used here since this is a trunk-only option. (closes issue
- #11307, reported by pj, patched by me)
-
-2007-11-19 23:03 +0000 [r89439-89440] Russell Bryant <russell@digium.com>
-
- * include/asterisk/module.h: Be a bit more pedantic about the type
- for holding the md5 sum for the build options. Also, doxygenify
- the comment.
-
- * funcs/func_sysinfo.c: Make the SYSINFO documentation reflect
- which options were compiled in
-
-2007-11-19 22:55 +0000 [r89438] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: These changes were made in response to
- niklas@tese.se's letter of 11-17-2007, where he had 20 and 201 in
- two different contexts, included in the same context. In that
- particular case, we were behaving the same as 1.4, but after
- experimenting, I quickly found that if 20 and 201 were in the
- same extension, 1.4 would return 201, and this code returns 20.
- These changes now enable the current code to replicate the
- behavior of 1.4 in respect to MATCHMORE in cases like this.
-
-2007-11-19 21:18 +0000 [r89430-89433] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_vpb.cc, channels/misdn_config.c, main/dsp.c:
- another few errno.h removals
-
- * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c,
- apps/app_meetme.c, pbx/pbx_ael.c, pbx/pbx_lua.c,
- pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_externalivr.c,
- apps/app_directory.c, apps/app_system.c, pbx/pbx_config.c,
- apps/app_milliwatt.c: more errno.h removal
-
- * funcs/func_sysinfo.c: remove unnecessary headers
-
- * funcs/func_base64.c, funcs/func_volume.c: remove some unnecessary
- includes.
-
-2007-11-19 20:13 +0000 [r89429] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Change delimiter of SIPPEER to be comma
- (instead of pipe) and further deprecate the old ':' delimiter
- Reported by: pj Patch by: tilghman Closes issue #11305
-
-2007-11-19 19:51 +0000 [r89424-89428] Luigi Rizzo <rizzo@icir.org>
-
- * codecs/codec_lpc10.c, codecs/codec_a_mu.c, codecs/codec_g722.c,
- codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c,
- codecs/codec_g726.c, codecs/codec_gsm.c, codecs/codec_ulaw.c,
- codecs/codec_ilbc.c, codecs/codec_zap.c: remove some useless
- includes from codecs
-
- * formats/format_ilbc.c, formats/format_sln.c,
- formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c,
- formats/format_ogg_vorbis.c, formats/format_g723.c,
- formats/format_h263.c, formats/format_h264.c,
- formats/format_wav_gsm.c, formats/format_g726.c,
- formats/format_jpeg.c, formats/format_gsm.c,
- formats/format_g729.c: format handlers don't need network, lock,
- channel and scheduler headers
-
- * include/asterisk.h, include/asterisk/compat.h,
- include/asterisk/lock.h, utils/extconf.c,
- include/asterisk/abstract_jb.h: move the declaration of struct
- ast_channel ast_frame and ast_module to compat.h so it is always
- available - hopefully this will let us reduce the number of
- inclusions of channel.h and frame.h
-
- * main/udptl.c, main/autoservice.c, funcs/func_rand.c,
- cdr/cdr_sqlite3_custom.c, main/frame.c, funcs/func_module.c,
- main/threadstorage.c, main/say.c, funcs/func_env.c,
- funcs/func_strings.c, main/devicestate.c,
- cdr/cdr_adaptive_odbc.c, main/indications.c, main/config.c,
- main/loader.c, main/term.c, main/cli.c, funcs/func_shell.c,
- main/http.c, cdr/cdr_odbc.c, main/db.c, cdr/cdr_manager.c,
- main/sched.c, main/pbx.c, funcs/func_timeout.c,
- funcs/func_math.c, funcs/func_cut.c, main/chanvars.c,
- main/netsock.c, funcs/func_curl.c, main/srv.c, main/privacy.c,
- funcs/func_cdr.c, funcs/func_channel.c, main/audiohook.c,
- funcs/func_iconv.c, main/alaw.c, main/asterisk.c,
- funcs/func_base64.c, funcs/func_md5.c, funcs/func_sysinfo.c,
- main/utils.c, funcs/func_sha1.c, cdr/cdr_pgsql.c,
- funcs/func_logic.c, cdr/cdr_radius.c, main/enum.c,
- funcs/func_uri.c, main/io.c, cdr/cdr_csv.c, main/ulaw.c,
- main/channel.c, main/cdr.c, funcs/func_enum.c, main/dial.c,
- funcs/func_groupcount.c, main/manager.c, main/tdd.c,
- funcs/func_odbc.c, cdr/cdr_sqlite.c, main/logger.c, main/app.c,
- main/image.c, main/dns.c, cdr/cdr_custom.c, funcs/func_version.c,
- funcs/func_db.c, main/dnsmgr.c, main/translate.c,
- main/slinfactory.c, funcs/func_lock.c, main/acl.c, main/rtp.c,
- cdr/cdr_tds.c, funcs/func_realtime.c, main/hashtab.c,
- funcs/func_blacklist.c, main/abstract_jb.c, main/cryptostub.c,
- main/adsistub.c, main/file.c, main/callerid.c, main/astmm.c,
- funcs/func_callerid.c, main/dsp.c: another bunch of include
- removals (errno.h and asterisk/logger.h)
-
- * channels/chan_local.c, apps/app_record.c,
- apps/app_alarmreceiver.c, apps/app_chanisavail.c,
- apps/app_ices.c, apps/app_exec.c, channels/chan_iax2.c,
- channels/chan_skinny.c, formats/format_pcm.c,
- apps/app_dumpchan.c, apps/app_zapras.c, formats/format_h263.c,
- codecs/codec_g722.c, formats/format_wav.c, apps/app_softhangup.c,
- codecs/codec_g726.c, formats/format_ogg_vorbis.c,
- apps/app_morsecode.c, apps/app_talkdetect.c, apps/app_db.c,
- apps/app_speech_utils.c, apps/app_sendtext.c,
- formats/format_g726.c, apps/app_mixmonitor.c, res/res_odbc.c,
- apps/app_voicemail.c, channels/chan_vpb.cc, formats/format_sln.c,
- res/res_snmp.c, apps/app_dictate.c, apps/app_authenticate.c,
- apps/app_readexten.c, codecs/codec_gsm.c, apps/app_userevent.c,
- channels/chan_gtalk.c, res/res_jabber.c, apps/app_setcallerid.c,
- res/res_config_odbc.c, apps/app_osplookup.c, apps/app_mp3.c,
- apps/app_minivm.c, res/res_realtime.c, formats/format_h264.c,
- apps/app_directory.c, apps/app_rpt.c, channels/chan_mgcp.c,
- apps/app_adsiprog.c, codecs/codec_lpc10.c,
- res/res_config_pgsql.c, apps/app_read.c, channels/chan_sip.c,
- codecs/codec_alaw.c, res/res_adsi.c, res/res_crypto.c,
- channels/chan_jingle.c, apps/app_channelredirect.c,
- apps/app_forkcdr.c, formats/format_vox.c, apps/app_sms.c,
- formats/format_g723.c, apps/app_verbose.c, apps/app_stack.c,
- apps/app_readfile.c, res/res_features.c, codecs/codec_adpcm.c,
- apps/app_sayunixtime.c, apps/app_test.c, apps/app_image.c,
- formats/format_wav_gsm.c, res/res_smdi.c,
- include/asterisk/compat.h, apps/app_skel.c, apps/app_zapscan.c,
- channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c,
- formats/format_jpeg.c, formats/format_gsm.c,
- apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c,
- apps/app_privacy.c, codecs/codec_speex.c, apps/app_echo.c,
- channels/chan_agent.c, apps/app_disa.c,
- channels/iax2-provision.c, res/res_ael_share.c,
- apps/app_transfer.c, res/res_monitor.c, apps/app_playback.c,
- channels/chan_misdn.c, apps/app_waitforring.c,
- apps/app_zapbarge.c, channels/chan_features.c, apps/app_macro.c,
- apps/app_zapateller.c, res/res_indications.c,
- codecs/codec_ilbc.c, apps/app_chanspy.c, channels/chan_unistim.c,
- apps/app_meetme.c, res/res_musiconhold.c, apps/app_followme.c,
- codecs/codec_zap.c, res/res_config_sqlite.c,
- channels/misdn_config.c, apps/app_controlplayback.c,
- formats/format_ilbc.c, channels/chan_phone.c, res/res_agi.c,
- main/logger.c, apps/app_ivrdemo.c, apps/app_parkandannounce.c,
- res/res_clioriginate.c, apps/app_while.c, include/asterisk.h,
- apps/app_nbscat.c, channels/chan_zap.c, codecs/codec_a_mu.c,
- res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
- res/res_convert.c, apps/app_getcpeid.c, apps/app_system.c,
- apps/app_queue.c, channels/chan_oss.c, channels/chan_usbradio.c,
- apps/app_flash.c, apps/app_directed_pickup.c,
- channels/chan_h323.c, codecs/codec_ulaw.c, channels/chan_nbs.c,
- apps/app_senddtmf.c, formats/format_g729.c: include "logger.h"
- and errno.h from asterisk.h - usage shows that they were included
- almost everywhere. Remove some of the instances.
-
-2007-11-19 17:18 +0000 [r89422] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: a correction to code involved in an extension removal
-
-2007-11-19 16:29 +0000 [r89421] Mark Michelson <mmichelson@digium.com>
-
- * funcs/func_sysinfo.c (added), CHANGES: Adding SYSINFO() dialplan
- function for retrieval of system information
-
-2007-11-19 15:55 +0000 [r89417-89420] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_features.c: Merged revisions 89419 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89419 | file | 2007-11-19 11:53:32 -0400 (Mon, 19 Nov 2007) | 6
- lines Print out the correct filename (features.conf) in the log
- message when parkpos options are incorrect. (closes issue #11295)
- Reported by: Laureano Patches: res_features.c.patch uploaded by
- Laureano (license 265) ........
-
- * /, doc/tex/localchannel.tex: Merged revisions 89416 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov
- 2007) | 4 lines Clarify documentation a bit, include that a frame
- has to pass through the core in order for the Local channel
- optimization to happen. (closes issue #11246) Reported by: jon
- ........
-
-2007-11-19 14:36 +0000 [r89412] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/logger.h: revert inclusion of options.h
-
-2007-11-19 14:03 +0000 [r89410] Joshua Colp <jcolp@digium.com>
-
- * apps/app_playback.c: Change warning messages (which are really
- debug messages) into debug messages. (closes issue #11288)
- Reported by: IgorG Patches: saydebug-89394-1-trunk.patch uploaded
- by IgorG (license 20)
-
-2007-11-19 09:16 +0000 [r89404-89407] Olle Johansson <oej@edvina.net>
-
- * CHANGES: Update CHANGES
-
- * channels/chan_sip.c: Adding busy-level to the SIP_PEER() dialplan
- function. With this, you can control the peer in the dialplan, so
- you avoid placing outbound calls when the device has reached
- busy-level. Reported by pj. Closes bug #11180
-
- * main/acl.c: Add some debugging to the routines that finds our
- local IP address. Related to bug #9225
-
- * channels/chan_sip.c: Make some notes about a problem I found with
- the OPTIONs handler while working with the bug tracker. Since we
- don't authenticate devices (peers/users) on OPTIONS we don't have
- the proper context set for the user/peer. However, we might not
- want to process an authentication for every OPTIONS, so we could
- have a config option for this, "optionsforceok" to always answer
- 200 OK on the request and not check device or destination, nor
- add a SDP. If Asterisk sends the OPTIONs request, it doesn't care
- about the reply. Some devices use OPTIONs to discover
- capabilities, since we should answer like an INVITE from the
- device and we need to support that properly too, which we don't
- today. So much to do :-)
-
-2007-11-18 21:50 +0000 [r89394-89399] Joshua Colp <jcolp@digium.com>
-
- * build_tools/make_buildopts_h: Add OSX into the logic that uses
- md5 instead of md5sum.
-
- * include/asterisk/compat.h: Use the easy way that rizzo mentioned,
- only include malloc.h on the Windows platform.
-
- * include/asterisk/compat.h: Revert last commit, apparently
- buildbot lied to me.
-
- * include/asterisk/compat.h: Change how we handle alloca to conform
- with how it is suggested in the autoconf manual for
- AC_FUNC_ALLOCA. FreeBSD 6 now builds again and no other platforms
- should be broken by this.
-
- * configure, configure.ac: Change autoconf logic a bit so it says
- what it is looking for in two instances where it didn't.
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/lock.h, include/asterisk/network.h: Use autoconf
- logic to determine the presence of
- PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP and
- PTHREAD_MUTEX_RECURSIVE_NP. Enclose error message from network.h
- in "
-
-2007-11-17 21:47 +0000 [r89393] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add SS7 Generic address support (#11156)
-
-2007-11-17 19:29 +0000 [r89389-89392] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/compat.h: if alloca.h is not present, try
- malloc.h
-
- * agi/Makefile: temporarily disable this target in mingw
-
- * Makefile: will i ever get precedences for windows right ? in the
- meantime, use a variable to ease enabling/disabling print
- subdirectories.
-
- * Makefile: reformulate dependencies in a more correct way
-
-2007-11-17 17:46 +0000 [r89388] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, pbx/pbx_dundi.c: a quick fix to pbx_dundi.c to make
- it so it will compile. Hope I did the right thing. And some
- additions to removal of extens to take care of hashtab pointers
- in all cases.
-
-2007-11-17 17:27 +0000 [r89363-89387] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile.moddir_rules, Makefile.rules: as discussed some time ago
- on the -dev list, create embedde object with a .eo suffix even if
- they are coming from .cc sources. This simplifies the handling in
- the build scripts.
-
- * include/asterisk/network.h: prefer socket.h over other variants
- (winsock etc.)
-
- * channels/chan_local.c, main/translate.c,
- channels/chan_features.c, main/http.c, main/config.c: trim more
- redundant headers
-
- * main/acl.c: remove unnecessary includes
-
- * main/udptl.c, main/dnsmgr.c, channels/chan_sip.c, main/acl.c,
- main/dns.c, main/rtp.c, main/netsock.c: fix breakage induced by
- previous mistake
-
- * Makefile: wrong variable, wrong order -> broken build.
-
- * include/asterisk/acl.h, include/asterisk/utils.h,
- include/asterisk/autoconfig.h.in, include/asterisk/rtp.h,
- configure.ac, main/acl.c, include/asterisk/netsock.h,
- main/utils.c, include/asterisk/manager.h, main/netsock.c,
- main/manager.c, res/res_agi.c, pbx/pbx_dundi.c,
- include/asterisk/udptl.h, include/asterisk/dnsmgr.h,
- main/asterisk.c: start using asterisk/network.h for network
- related headers. Also remove some unnecessary includes.
-
- * include/asterisk/network.h (added): wrapper for all generic
- network headers that have different names and locations on the
- various systems.
-
- * main/cygload.c: main is called main not amain!
-
- * main/Makefile: conditional targets for building the windows
- version
-
- * Makefile: support cygwin targets
-
- * Makefile.moddir_rules: and this is the last one to have asterisk
- compile (not run yet) natively under cygwin.
-
- * apps/app_sms.c: another cygwin compatibility fix. This one must
- be handled in a better way in configure, also for other
- architectures
-
- * utils/Makefile, main/Makefile, utils/extconf.c: more
- cygwin/mingw32 compatibility fixes
-
- * include/asterisk/channel.h: use autoconf results to conditionally
- compile timersub
-
- * include/asterisk/lock.h: compatibility fixes for cygwin
-
- * include/asterisk/compat.h: some version of flex produce code that
- wants __STDC_VERSION__ defined, but the compiler does not always
- define it.
-
- * Makefile: these linker flags apply to both cygwin and mingw32
-
- * utils/hashtest2.c: add a return NULL to a function that is
- expected to return a value so compilers that don't understand
- that this code is NOTREACHED will not complain (the fault is not
- much on the compiler but on the declaration of pthread_exit on
- certain platforms) s/certain platform/cygwin/ if you are really
- curious
-
- * main/loader.c: define RTLD_LOCAL for platforms that don't have
- it. This is only to complete the build, clearly the linker
- behaviour will be completely different and likely to cause
- trouble in those cases.
-
- * channels/Makefile: filter out modules that do not compile under
- windows (this should be handled with the dependencies generated
- by configure and menuselect, but will be fixed later)
-
- * main/utils.c: netdb.h is used for gethostbyname, and it was not
- included in some platforms.
-
- * main/cygload.c (added): Loader for cygwin where asterisk is
- really a big dll (something like this is already in 1.2)
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- timersub is a macro not a function, so write the check in a way
- that detects both formats.
-
-2007-11-17 06:34 +0000 [r89359-89362] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_lua.c: fix the build of pbx_lua
-
- * configure, include/asterisk/autoconfig.h.in,
- include/asterisk/compat.h, configure.ac, include/asterisk/io.h,
- include/asterisk/channel.h: Update the configure script check for
- sys/poll.h to also provide the result in
- include/asterisk/autoconfig.h. Also, move the conditional include
- of sys/poll.h or asterisk/poll-compat.h into asterisk/config.h
- instead of the two headers it existed in before.
-
- * build_tools/make_buildopts_h: actually let this compile, oops :(
-
- * build_tools/make_buildopts_h: Use the fix suggested by Tilghman
- on the -dev to make cutting up the BUILDSUM friendly to non-bash
- shells. I think this should work for BSD/mingw as well, but did
- not yet remove the switch statement.
-
-2007-11-17 04:19 +0000 [r89348-89358] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: linker flags for mingw32
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: add
- detection for timersub() and winsock.h/winsock2.h
-
- * include/asterisk/endian.h: provide definitions for
- __LITTLE_ENDIAN and __BIG_ENDIAN if not present.
-
- * main/Makefile, include/asterisk/io.h, include/asterisk/channel.h:
- use poll as detected by configure
-
- * configure, configure.ac, makeopts.in: use autoconf to check for
- the existence of sys/poll.h
-
- * build_tools/make_buildopts_h: this script is run on the build
- system, not on the host.
-
- * Makefile.moddir_rules: compatibility fix for mingw32
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4, makeopts.in: acinclude.m4: add a function to help
- checking sdl-config, gtk-config and the like (this could be used
- for gtk and gtk2 as well) Other files: add tests for sdl,
- sdl_image and avcodec and regenerate configure and
- autoconfig.h.in
-
- * include/asterisk/autoconfig.h.in, configure.ac: add check for the
- presence of glob
-
- * channels/chan_jingle.c, channels/chan_unistim.c,
- funcs/func_enum.c, channels/chan_local.c, channels/chan_misdn.c,
- channels/chan_skinny.c, funcs/func_odbc.c, channels/chan_h323.c,
- utils/ael_main.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c,
- apps/app_db.c, channels/chan_mgcp.c: more removal of duplicate
- #include lines
-
- * main/udptl.c, funcs/func_module.c, res/res_features.c,
- funcs/func_lock.c, res/res_adsi.c, funcs/func_strings.c,
- channels/chan_agent.c, pbx/dundi-parser.c, main/rtp.c,
- pbx/pbx_loopback.c, funcs/func_blacklist.c,
- channels/chan_features.c, apps/app_dumpchan.c, res/res_agi.c,
- main/logger.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
- apps/app_rpt.c, main/asterisk.c, apps/app_parkandannounce.c:
- remove a bunch of duplicate includes Reproduce with grep -r
- #include . | grep -v .svn | grep -v Binary | sort | uniq -c |
- sort -nr
-
-2007-11-16 23:44 +0000 [r89347] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: Fix broken parking dial-back
-
-2007-11-16 23:33 +0000 [r89346] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: My goodness, haven't handled an extension deletion.
- Add code to ast_context_remove_extension2() to remove an
- extension from the trie. Done by marking it deleted. The
- scoreboard won't update for it any more. Also, a couple of calls
- to insert hashtab had a spurious ->exten, which was removed.
-
-2007-11-16 23:28 +0000 [r89341-89345] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/paths.h, include/asterisk.h: paths are already
- in include/asterisk/paths.h so don't duplicate them in
- include/asterisk.h
-
- * include/asterisk/utils.h, include/asterisk/lock.h: whitespace
- only change - adjust indentation and add some comments on the
- content of these two files. utils.h (which is included in over
- 150 files) contains a lot of unrelated functions which require
- the inclusion of a large number of other headers. At some point
- we should partition its content in a better way.
-
-2007-11-16 21:23 +0000 [r89333-89338] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/logger.h: logger.h does not need options.h
-
- * include/asterisk/utils.h, channels/chan_sip.c,
- include/asterisk/astobj.h, include/asterisk/compat.h,
- include/asterisk/channel.h, include/asterisk/strings.h,
- utils/extconf.c, include/asterisk/frame.h,
- include/asterisk/stringfields.h, include/asterisk/endian.h:
- remove redundant #include "asterisk/compat.h", but make sure that
- asterisk/compiler.h is included everywhere
-
- * main/acl.c, main/asterisk.c: remove duplicate headers. Properly
- check for netdb.h (there is actually tens of places to fix)
-
- * Makefile.rules: put back default optimization to -O6 (previously
- changed by mistake)
-
- * main/frame.c, main/threadstorage.c, apps/app_alarmreceiver.c,
- apps/app_ices.c, channels/chan_iax2.c, apps/app_exec.c,
- channels/chan_skinny.c, main/strcompat.c, pbx/pbx_ael.c,
- apps/app_zapras.c, formats/format_h263.c, cdr/cdr_odbc.c,
- include/asterisk/sha1.h, main/db.c, cdr/cdr_manager.c,
- main/pbx.c, funcs/func_timeout.c, formats/format_wav.c,
- apps/app_softhangup.c, codecs/codec_g726.c, funcs/func_cut.c,
- apps/app_talkdetect.c, apps/app_db.c, funcs/func_channel.c,
- main/privacy.c, funcs/func_iconv.c, pbx/pbx_config.c,
- main/asterisk.c, res/res_odbc.c, include/asterisk/stringfields.h,
- apps/app_voicemail.c, formats/format_sln.c,
- apps/app_authenticate.c, apps/app_readexten.c,
- apps/app_userevent.c, codecs/codec_gsm.c, Makefile.rules,
- apps/app_setcallerid.c, include/asterisk/astmm.h,
- res/res_config_odbc.c, apps/app_osplookup.c, funcs/func_odbc.c,
- apps/app_mp3.c, formats/format_h264.c, apps/app_directory.c,
- main/md5.c, res/res_config_pgsql.c, main/dnsmgr.c,
- funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c,
- res/res_crypto.c, include/asterisk/cli.h, channels/chan_jingle.c,
- apps/app_forkcdr.c, funcs/func_blacklist.c, main/abstract_jb.c,
- main/file.c, apps/app_sms.c, formats/format_g723.c, main/astmm.c,
- apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c,
- main/autoservice.c, funcs/func_module.c, codecs/codec_adpcm.c,
- cdr/cdr_adaptive_odbc.c, main/devicestate.c, apps/app_image.c,
- formats/format_wav_gsm.c, main/indications.c, pbx/pbx_loopback.c,
- funcs/func_shell.c, include/asterisk/compat.h, apps/app_skel.c,
- main/plc.c, channels/chan_alsa.c, apps/app_externalivr.c,
- formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c,
- main/sched.c, apps/app_dial.c, apps/app_page.c, apps/app_disa.c,
- channels/iax2-provision.c, res/res_monitor.c, main/netsock.c,
- apps/app_waitforring.c, main/fixedjitterbuf.c,
- include/asterisk/lock.h, apps/app_chanspy.c, apps/app_cdr.c,
- channels/chan_unistim.c, funcs/func_base64.c, funcs/func_md5.c,
- apps/app_meetme.c, main/sha1.c, funcs/func_vmcount.c,
- res/res_musiconhold.c, cdr/cdr_radius.c, apps/app_followme.c,
- res/res_config_sqlite.c, main/fskmodem.c,
- channels/misdn_config.c, apps/app_controlplayback.c,
- cdr/cdr_csv.c, formats/format_ilbc.c, main/cdr.c,
- channels/chan_phone.c, funcs/func_enum.c, main/dial.c,
- main/manager.c, funcs/func_groupcount.c, cdr/cdr_sqlite.c,
- main/logger.c, main/image.c, apps/app_ivrdemo.c,
- res/res_clioriginate.c, apps/app_nbscat.c, codecs/codec_a_mu.c,
- channels/chan_zap.c, main/slinfactory.c, res/res_convert.c,
- pbx/pbx_lua.c, apps/app_queue.c, apps/app_system.c,
- channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c,
- channels/chan_usbradio.c, main/hashtab.c, apps/app_flash.c,
- include/asterisk/strings.h, apps/app_senddtmf.c,
- funcs/func_callerid.c, include/asterisk/time.h,
- channels/chan_local.c, funcs/func_dialgroup.c, funcs/func_env.c,
- apps/app_record.c, funcs/func_strings.c, apps/app_chanisavail.c,
- pbx/pbx_spool.c, apps/app_dumpchan.c, formats/format_pcm.c,
- main/http.c, main/stdtime/localtime.c, codecs/codec_g722.c,
- apps/app_morsecode.c, formats/format_ogg_vorbis.c,
- channels/iax2-parser.c, apps/app_speech_utils.c,
- include/asterisk/logger.h, main/srv.c, apps/app_sendtext.c,
- funcs/func_cdr.c, include/asterisk/md5.h, utils/hashtest2.c,
- utils/ael_main.c, main/audiohook.c, apps/app_mixmonitor.c,
- formats/format_g726.c, channels/chan_vpb.cc, apps/app_dictate.c,
- channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c,
- res/res_jabber.c, funcs/func_uri.c, main/io.c,
- include/asterisk/abstract_jb.h, main/channel.c,
- apps/app_minivm.c, res/res_realtime.c, main/dns.c,
- apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
- codecs/codec_lpc10.c, apps/app_read.c, codecs/codec_alaw.c,
- res/res_adsi.c, include/asterisk/plc.h,
- apps/app_channelredirect.c, formats/format_vox.c,
- main/cryptostub.c, main/callerid.c, pbx/pbx_dundi.c,
- funcs/func_devstate.c, funcs/func_rand.c, apps/app_readfile.c,
- cdr/cdr_sqlite3_custom.c, main/say.c, res/res_features.c,
- apps/app_sayunixtime.c, apps/app_test.c, main/config.c,
- main/loader.c, main/term.c, main/cli.c, res/res_smdi.c,
- include/asterisk/astobj.h, apps/app_zapscan.c, apps/app_amd.c,
- pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c,
- include/asterisk/utils.h, apps/app_privacy.c,
- codecs/codec_speex.c, apps/app_echo.c, channels/chan_agent.c,
- funcs/func_math.c, res/res_ael_share.c, pbx/dundi-parser.c,
- apps/app_transfer.c, include/asterisk/manager.h,
- apps/app_playback.c, main/chanvars.c, apps/app_zapbarge.c,
- channels/chan_misdn.c, funcs/func_curl.c,
- channels/chan_features.c, apps/app_macro.c, codecs/codec_ilbc.c,
- res/res_indications.c, apps/app_zapateller.c, main/dlfcn.c,
- include/asterisk/slinfactory.h, utils/hashtest.c, main/utils.c,
- funcs/func_sha1.c, codecs/codec_zap.c, main/enum.c,
- include/asterisk/file.h, main/tdd.c, funcs/func_volume.c,
- res/res_agi.c, main/app.c, apps/app_parkandannounce.c,
- cdr/cdr_custom.c, apps/app_while.c, funcs/func_db.c,
- res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
- main/translate.c, include/asterisk/config.h, main/jitterbuf.c,
- main/acl.c, apps/app_getcpeid.c, funcs/func_global.c, main/rtp.c,
- funcs/func_extstate.c, apps/app_directed_pickup.c,
- main/adsistub.c, channels/chan_h323.c, codecs/codec_ulaw.c,
- main/event.c, channels/chan_nbs.c, pbx/pbx_gtkconsole.c,
- formats/format_g729.c: Start untangling header inclusion in a way
- that does not affect build times - tested, there is no
- measureable difference before and after this commit. In this
- change: use asterisk/compat.h to include a small set of system
- headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h,
- stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the
- inclusion is conditional on HAVE_FOO_H as determined by autoconf.
- Normally, source files should not include any of the above system
- headers, and instead use either "asterisk.h" or
- "asterisk/compat.h" which does it better. For the time being I
- have left alone second-level directories (main/db1-ast, etc.).
-
-2007-11-16 19:51 +0000 [r89331-89332] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c: Fixing a problem pointed out by Qwell
-
- * main/manager.c: Added some locks that should have been around
- astman_send_error, at least according to the comments. (closes
- issue #11258, reported and patched by eliel)
-
-2007-11-16 19:26 +0000 [r89329-89330] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: This corrects a hashtab removal, given a bad argument
-
- * main/pbx.c, res/res_features.c: This fixes a problem with pattern
- ranges; and corrects a situation in res_features, where an
- extension would be created with the name Zap/51, as an example.
- THe / is bad because it would tend to mean that the 51 is to be
- cid matched.
-
-2007-11-16 18:48 +0000 [r89328] Luigi Rizzo <rizzo@icir.org>
-
- * build_tools/make_buildopts_h: both md5sum and variable
- substitutions such as ${BUILDSUM:0:8} are not available in
- FreeBSD. For the time being, put in a workaround so we can build
- the system, and wait for the result of the discussion on whether
- we can store the md5 as a string rather than 4 ints (if so, we
- won't need more complex tricks with awk or sed for splitting the
- md5). 1.4 will be fixed when we decide the issue.
-
-2007-11-16 17:11 +0000 [r89327] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Adding confirmation playback when
- forwarding voicemail messages. This will attempt to play the
- name(s) of the person(s) to whom you are forwarding the message
- prior to prompting for prepending. If no name is found, the
- extension is read back verbatim. (closes issue #9046, reported
- and patched by jaroth)
-
-2007-11-16 16:56 +0000 [r89326] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, include/asterisk/module.h, build_tools/make_buildopts_h,
- main/loader.c: Merged revisions 89325 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89325 | kpfleming | 2007-11-16 10:47:46 -0600 (Fri, 16 Nov 2007)
- | 4 lines To help combat problems where people build external
- modules (asterisk-addons or others) and then change the build
- options of the Asterisk build in a way that makes the
- incompatible without warning, this commit introduces an MD5
- signature of the important build-time options and includes that
- signature into modules when they are built. When the loader loads
- one of these modules and notices the problem, it will emit a
- warning to console and refuse to initialize the module, as doing
- so could cause the system to be unstable or even crash. If you
- upgrade to this version of Asterisk, you must rebuild *all* of
- your modules that came from other sources before trying to run
- this version. If you are using Digium's G.729 binary codec
- module, you will need v33 or newer. ........
-
-2007-11-16 15:44 +0000 [r89324] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 89323 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89323 | mmichelson | 2007-11-16 09:28:22 -0600 (Fri, 16 Nov
- 2007) | 5 lines Make realtime queues accessible from the
- QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
- patched by atis, with small modifications from me) ........
-
-2007-11-16 10:07 +0000 [r89322] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/config.h, main/config.c: add a small new
- function to retrieve variables from a config once we have a
- pointer to the category.
-
-2007-11-16 10:06 +0000 [r89321] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed #10631, about one way audio. thanks
- IgorG again.
-
-2007-11-16 09:51 +0000 [r89320] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_oss.c: move the inner part of config file parsing
- to a separate function, so it can be reused in the implementation
- of cli commands when they have a similar syntax.
-
-2007-11-16 08:54 +0000 [r89319] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed compilation of chan_misdn, #11269,
- thanks IgorG.
-
-2007-11-15 23:50 +0000 [r89299-89312] Tilghman Lesher <tlesher@digium.com>
-
- * main/utils.c, include/asterisk/stringfields.h: If we're going to
- be passing a negative value for the size of a stringfield, in
- order to indicate something, then using an UNSIGNED parameter is
- bad, mmmmmkay?
-
- * Makefile, /: Merged revisions 89302 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89302 | tilghman | 2007-11-15 12:37:38 -0600 (Thu, 15 Nov 2007)
- | 2 lines Start Asterisk in Debian at a more reasonable time
- (since zaptel is at level 20) ........
-
- * /, channels/misdn/isdn_lib.c: Merged revisions 89301 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15
- Nov 2007) | 2 lines Fix an uninitialized memory read found by
- valgrind ........
-
- * apps/app_zapscan.c: Fix trunk breakage due to chan->lock being
- renamed.
-
- * /, channels/chan_iax2.c: Merged revisions 89298 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007)
- | 5 lines Yet another memory corruption issue. Reported by: atis
- Patch by: tilghman Fixes issue #10923 ........
-
-2007-11-15 17:27 +0000 [r89297] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 89296 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) |
- 8 lines Update the SLAStation application to account for the case
- where the SLA thread has a call out to the station, but the user
- has pressed a line button to answer the call instead of picking
- up the handset. If they do, the phone sends out a new INVITE. So,
- the SLAStation app must check to see if it is picking up a
- ringing trunk, and ensure that the other stations stop ringing.
- (reported internally, patched by me, tested by mogorman) ........
-
-2007-11-15 16:50 +0000 [r89294-89295] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: Get rid of a previously missed ast_log call for
- debug, no longer nec.
-
- * main/pbx.c: Perhaps I went overboard on initializing things. I
- can remove unnecc. stuff later. A few bug fixes. Killing small
- bugs on the way to killing bigger ones. Removed locking on
- hashtabs; there's plenty of locks already being taken. A small
- bug in the root_tree hashtab compare func.
-
-2007-11-15 16:20 +0000 [r89293] Luigi Rizzo <rizzo@icir.org>
-
- * main/channel.c, apps/app_channelredirect.c, main/manager.c,
- res/res_features.c, apps/app_softhangup.c,
- include/asterisk/channel.h, include/asterisk/lock.h,
- apps/app_senddtmf.c: access channel locks through
- ast_channel_lock/unlock/trylock and not through ast_mutex
- primitives. To detect all occurrences, I have renamed the lock
- field in struct ast_channel so it is clear that it shouldn't be
- used directly. There are some uses in res/res_features.c (see
- details of the diff) that are error prone as they try and lock
- two channels without caring about the order (or without
- explaining why it is safe).
-
-2007-11-15 15:39 +0000 [r89290-89291] Joshua Colp <jcolp@digium.com>
-
- * UPGRADE.txt: Fix typo in UPGRADE.txt. 'increase' should have been
- used, not 'increasing'.
-
- * channels/chan_sip.c, channels/chan_h323.c,
- channels/misdn_config.c: And file said... let trunk build again!
- Accomplished by some more constification, and marking a function
- in chan_sip as purposely unused until it is fixed up.
-
-2007-11-15 14:58 +0000 [r89287-89289] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 89288 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89288 | mmichelson | 2007-11-15 08:57:28 -0600 (Thu, 15 Nov
- 2007) | 3 lines Undoing previous commit since I realize it was
- wrong ........
-
- * main/manager.c, /: Merged revisions 89286 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89286 | mmichelson | 2007-11-15 08:54:10 -0600 (Thu, 15 Nov
- 2007) | 4 lines Adding a missing mutex unlock. (closes issue
- 11256, reported and patched by ys) ........
-
-2007-11-15 12:21 +0000 [r89278-89285] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Always relying on the responses when
- crossing NAT's are not a good solution, it breaks communication.
- Rizzo - you need to implement a configuration option for this
- code. It's good, but maybe should be off by default.
-
- * /, channels/chan_sip.c: Merged revisions 89281 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6
- lines Don't send re-invites during pending INVITE transactions.
- Patch by one47 - thanks! Closes issue #9305 ........
-
- * /, channels/chan_sip.c: Merged revisions 89280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5
- lines Improve support for multipart messages. Code by gasparz,
- changes by me (mostly formatting). Thanks, gasparz! Closes issue
- #10947 ........
-
- * channels/chan_sip.c: Exit early instead of deciding to exit after
- processing the message.
-
- * channels/chan_sip.c, configs/sip.conf.sample: Add support for
- application/dtmf SIP INFO dtmf handling. Yep, another way of
- handling DTMF in SIP. Totally undocumented, but implemented in
- enough devices so we have to support it. Code by sergee, small
- changes by oej. Closes issue #11049
-
-2007-11-15 01:42 +0000 [r89277] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: Had trouble playing with parking; spent a long time
- trying to reason out MATCHMORE mode. made these updates and xfers
- on zaptel lines seem to work ok now
-
-2007-11-15 00:01 +0000 [r89273-89276] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/app.c: Merged revisions 89275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89275 | tilghman | 2007-11-14 17:23:58 -0600 (Wed, 14 Nov 2007)
- | 5 lines When a recording ends with '#', we are improperly
- trimming an extra 200ms from the recording. Reported by: sim
- Patch by: tilghman Closes issue #11247 ........
-
- * main/channel.c: Typo
-
- * main/channel.c: Add callerid to the Hangup manager event.
- Reported by: outtolunc Patch by: outtolunc Closes issue #11248
-
-2007-11-14 18:05 +0000 [r89271-89272] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: Rescaled the weights of the patterns to give
- something more independent of pattern length; and make . less
- likely to win. Question: which should win for 14102241145--
- _1xxxxxxx. or _XXXXXXXXXXX -- right now, the pure X pattern will
- win.
-
- * main/pbx.c: A further problem highlighted by 11233 has been
- resolved; a certain combination of patterns in a certain order,
- led to a malformed trie, due to a ptr not being initialized in
- the loop. Also, some tree printing prettifications.
-
-2007-11-14 15:13 +0000 [r89269-89270] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_phone.c, channels/chan_zap.c, res/res_jabber.c,
- res/res_config_sqlite.c, main/config.c, res/res_odbc.c: One more
- typo in config.c; and missed conversions due to the constifying
- of ast_variable_new parameters
-
- * main/config.c: Typo
-
-2007-11-14 13:18 +0000 [r89268] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/acl.h, channels/chan_sip.c,
- include/asterisk/config.h, channels/chan_agent.c, res/res_adsi.c,
- main/acl.c, pbx/dundi-parser.c, apps/app_queue.c,
- channels/chan_iax2.c, main/enum.c, channels/chan_oss.c,
- apps/app_playback.c, main/config.c, pbx/dundi-parser.h,
- include/asterisk/abstract_jb.h, main/manager.c,
- channels/chan_skinny.c, apps/app_minivm.c, main/abstract_jb.c,
- main/logger.c, pbx/pbx_dundi.c, apps/app_directory.c,
- apps/app_voicemail.c: make the 'name' and 'value' fields in
- ast_variable const char * This prevents modifying the strings in
- the stored variables, and catched a few instances where this was
- actually done. Given the differences between trunk and 1.4 (and
- the fact that this is effectively an API change) it is better to
- fix 1.4 independently. These are chan_sip.c::sip_register()
- chan_skinny.c:: near line 2847 config.c:: near line 1774
- logger.c::make_components() res_adsi.c:: near line 1049 I may
- have missed some instances for modules that do not build here.
-
-2007-11-14 03:22 +0000 [r89263-89266] Russell Bryant <russell@digium.com>
-
- * main/hashtab.c, include/asterisk/hashtab.h: Fix up various coding
- guidelines issues ... - handle memory allocation failures - add
- an ast_ prefix to a publicly exported function - put curly braces
- in the right places - add a bunch of spaces where they should be
- be used
-
- * res/res_clioriginate.c: - Use the ARRAY_LEN macro in a couple
- places - return errors from load_module / unload_module
-
- * apps/app_dial.c: Use BEGIN_OPTIONS / END_OPTIONS to make the
- syntax highlighting in my editor happy
-
- * apps/app_queue.c: Instead of reserving 800 bytes for periodic
- announcements, use an array of ast_str pointers and only alloate
- space for the strings as needed.
-
-2007-11-14 01:16 +0000 [r89262] Joshua Colp <jcolp@digium.com>
-
- * main/srv.c, /: Merged revisions 89260 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89260 | file | 2007-11-13 21:15:12 -0400 (Tue, 13 Nov 2007) | 4
- lines Return the proper value when the srv_callback function
- executes properly. (closes issue #11240) Reported by: jtodd
- ........
-
-2007-11-14 01:15 +0000 [r89261] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Convert most of the strings in the call_queue
- struct to use stringfields.
-
-2007-11-14 00:54 +0000 [r89259] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/channel.c, main/pbx.c: use simpler technique for removing
- known entries from lists
-
-2007-11-14 00:33 +0000 [r89258] Russell Bryant <russell@digium.com>
-
- * main/image.c: - Simplify removing an item from a list - move a
- verbose message to after the item is added to the list - make use
- of the ARRAY_LEN macro in one spot
-
-2007-11-13 23:43 +0000 [r89256-89257] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: This hopefully will fix the re-opened 11233. Hadn't
- covered the case of a context with no patterns. (blush)
-
- * main/pbx.c: closes issue #11233 -- where some fine points in the
- algorithm to build the tree needed to be corrected. Many thanks
- for the test case, jtodd
-
-2007-11-13 21:01 +0000 [r89250-89253] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: This fixes a build error on my mac. It
- also works on my linux box. Let me know if it breaks any other
- platform ...
-
- * res/res_features.c: Fix a typo pointed out by outtolunc, thanks
- :)
-
- * channels/chan_sip.c: - Convert initialization of a struct to C99
- style instead of GNU style - Fix a minor spelling error in a
- comment
-
- * res/res_features.c, CHANGES: Update the ParkedCall application to
- grab the first available parked call if no parked extension is
- provided as an argument. (closes issue #10803) Reported by:
- outtolunc Patches: res_features-parkedcall-any.diff4 uploaded by
- outtolunc (license 237) - modified by me to work a bit
- differently ...
-
-2007-11-13 19:48 +0000 [r89249] Jason Parker <jparker@digium.com>
-
- * /, res/res_features.c: Merged revisions 89248 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11237) ........ r89248 | qwell | 2007-11-13 13:47:45 -0600
- (Tue, 13 Nov 2007) | 7 lines Revert change from revision 67064.
- It is documented behavior that if a parking extension already
- exists while using PARKINGEXTEN, dialplan execution will
- continue. If blind transferring to a Park with PARKINGEXTEN, you
- must keep this in mind, and handle the failure yourself. Issue
- 11237, reported by jon. ........
-
-2007-11-13 17:41 +0000 [r89247] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 89246 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007)
- | 2 lines If we set a value for qualify, we should actually pay
- attention to it, instead of overriding the value ........
-
-2007-11-13 16:03 +0000 [r89242] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_mixmonitor.c: Merged revisions 89241 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13
- Nov 2007) | 5 lines Reverting commit made in revision 89205 since
- it is unnecessary. Thanks to Kevin for pointing this out ........
-
-2007-11-13 14:03 +0000 [r89240] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/utils.c: Merged revisions 89239 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89239 | tilghman | 2007-11-13 07:51:53 -0600 (Tue, 13 Nov 2007)
- | 4 lines Debugging is running into the 16-lock limit. Increase
- to avoid. (This define is only effective when debugging is turned
- on, so there's no effect for most installations.) ........
-
-2007-11-13 01:19 +0000 [r89206-89207] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_mixmonitor.c: There is the potential to copy
- uninitialized memory into the mixmonitor->post_process string.
- This fix prevents that.
-
- * /, apps/app_mixmonitor.c: Merged revisions 89205 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12
- Nov 2007) | 5 lines Some sanity checking for MixMonitor. If only
- 1 argument is given, then the args.options and args.post_process
- strings are uninitialized and could contain garbage. This change
- handles this situation properly by only using arguments that we
- have parsed. ........
-
-2007-11-13 00:19 +0000 [r89202-89203] Jason Parker <jparker@digium.com>
-
- * Makefile: oops, somebody left out the directory here...
-
- * channels/chan_unistim.c, res/res_features.c, main/ast_expr2f.c,
- include/asterisk/config.h, res/res_convert.c, res/res_crypto.c,
- pbx/pbx_lua.c, include/asterisk/cli.h, include/asterisk/pbx.h,
- res/res_config_sqlite.c, res/res_monitor.c,
- include/asterisk/stringfields.h, res/res_clioriginate.c: Doxygen
- fixes. Also fix a common typo I kept seeing (arguement) in
- various files. Closes issue #11222, patch by snuffy (with
- arguement > argument by me).
-
-2007-11-12 23:33 +0000 [r89196-89201] Steve Murphy <murf@digium.com>
-
- * utils/hashtest.c: Don't forget the ASTERISK_VERSION for the sake
- of the mtx_prof stuff.
-
- * include/asterisk/hashtab.h: Thanks to snuffy for this doxygen
- update to hashtab.h; closes issue #11223
-
- * main/hashtab.c, include/asterisk/hashtab.h: Thanks to snuff-work,
- who brought up that these fixes might need to be made.
-
-2007-11-12 20:48 +0000 [r89195] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, /: Merged revisions 89194 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89194 | qwell | 2007-11-12 14:46:52 -0600 (Mon, 12 Nov 2007) | 1
- line Fix a typo pointed out by De_Mon on #asterisk-dev ........
-
-2007-11-12 20:16 +0000 [r89190] Kevin P. Fleming <kpfleming@digium.com>
-
- * utils/Makefile, utils/hashtest.c: (closes issue #11221) Reported
- by: eliel Patches: utils.Makefile.patch uploaded by eliel
- (modified by me) (license 64)
-
-2007-11-12 18:44 +0000 [r89186] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
- funcs/func_logic.c, apps/app_exec.c, apps/app_queue.c,
- apps/app_mixmonitor.c, cdr/cdr_manager.c: Based on a note in
- asterisk-dev by Brian Capouch, I determined I too agressive in
- not initializing arrays passed to pbx_substitute_variables_xxxx;
- I reviewed the code (again) and hopefully found every possible
- spot where substitute_variables is called conditionally, and made
- sure the char array involved was set to a null string.
-
-2007-11-12 17:44 +0000 [r89185] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /, channels/chan_sip.c: Merged revisions 89184
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007)
- | 5 lines Fix two cases of memory corruption caused by background
- threads. Reported by: atis Patch by: tilghman Fixes issue #10923
- ........
-
-2007-11-12 13:36 +0000 [r89178-89179] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged
- revisions 89173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) |
- 1 line if we're NT and no number was dialed and overlapdial is
- set, we wait for the ISDN timeout instead of starting our own
- timer. added a comment for the misdn.conf.sample for the
- overlapdial config option. ........
-
- * channels/misdn/isdn_lib_intern.h, channels/chan_misdn.c, /,
- channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
- Merged revisions 89172 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) |
- 1 line added restart all interfaces Restart_Indicator, to
- automatically send a RESTART after the L2 of a PTP Port comes up.
- Also fixed some places where we have send a RELEASE without need
- for it. ........
-
-2007-11-12 13:26 +0000 [r89177] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_unistim.c, utils/hashtest.c: Fix building on
- FreeBSD by including/not including some headers. (closes issue
- #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys
- (license 281)
-
-2007-11-12 13:22 +0000 [r89174-89176] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
- revisions 89171 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) |
- 1 line fixed a state/event issue with overlapdial=yes when no
- extension matched. removed the general sending of a
- RELEASE_COMPLETE when we receive a RELEASE, this is done by
- mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with
- mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to
- upgrade to at least mISDNuser-1.1.6 (when using the NT mode at
- all) ........
-
- * /, channels/misdn/isdn_lib.c: Merged revisions 89170 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12
- Nov 2007) | 1 line fixed the support for CW and therefore for the
- reject_cause option. ........
-
- * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample,
- channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged
- revisions 89169 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) |
- 1 line aded ntkeepcalls option, to avoid droÃpping calls when the
- L2 goes down on a PTP link. There are some pbx which do turn off
- the L1 for a very short while and restart it immediately.
- normally T310 should be started and after 10 seconds or so the
- calls should be dropped, this is a simple fix wihtout this timer.
- ........
-
-2007-11-09 18:57 +0000 [r89130-89132] Jason Parker <jparker@digium.com>
-
- * configs/usbradio.conf.sample (added): Add usbradio.conf.sample
- from branches/1.4/configs - r84162. It was mistakenly deleted in
- 1.4 without ever being merged to trunk. Reported by eliel on
- #asterisk-dev.
-
- * cdr/cdr_sqlite3_custom.c, configs/cdr_sqlite3_custom.conf
- (removed), configs/cdr_sqlite3_custom.conf.sample (added): Fix a
- few potential deadlocks in cdr_sqlite3_custom. (also rename
- sample config to .sample) Closes issue #11208, patch by Laureano.
-
-2007-11-09 16:00 +0000 [r89129] Steve Murphy <murf@digium.com>
-
- * res/ael/pval.c, utils/Makefile, main/pbx.c, main/hashtab.c
- (added), main/Makefile, utils/hashtest.c (added), pbx/pbx_ael.c,
- include/asterisk/hashtab.h (added), main/config.c: This is the
- perhaps the biggest, boldest, most daring change I've ever
- committed to trunk. Forgive me in advance any disruption this may
- cause, and please, report any problems via the bugtracker. The
- upside is that this can speed up large dialplans by 20 times (or
- more). Context, extension, and priority matching are all fairly
- constant-time searches. I introduce here my hashtables
- (hashtabs), and a regression for them. I would have used the
- ast_obj2 tables, but mine are resizeable, and don't need the
- object destruction capability. The hashtab stuff is well tested
- and stable. I introduce a data structure, a trie, for extension
- pattern matching, in which knowledge of all patterns is
- accumulated, and all matches can be found via a single traversal
- of the tree. This is per-context. The trie is formed on the first
- lookup attempt, and stored in the context for future lookups.
- Destruction routines are in place for hashtabs and the pattern
- match trie. You can see the contents of the pattern match trie by
- using the 'dialplan show' cli command when 'core set debug' has
- been done to put it in debug mode. The pattern tree traversal
- only traverses those parts of the tree that are interesting. It
- uses a scoreboard sort of approach to find the best match. The
- speed of the traversal is more a function of the length of the
- pattern than the number of patterns in the tree. The tree also
- contains the CID matching patterns. See the source code comments
- for details on how everything works. I believe the approach
- general enough that any issues that might come up involving fine
- points in the pattern matching algorithm, can be solved by just
- tweaking things. We shall see. The current pattern matcher is
- fairly involved, and replicating every nuance of it is difficult.
- If you find and report problems, I will try to resolve than as
- quickly as I can. The trie and hashtabs are added to the existing
- context and exten structs, and none of the old machinery has been
- removed for the sake of the multitude of functions that use them.
- In the future, we can (maybe) weed out the linked lists and save
- some space.
-
-2007-11-08 23:53 +0000 [r89124-89126] Jason Parker <jparker@digium.com>
-
- * /, main/say.c: Merged revisions 89125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11203) ........ r89125 | qwell | 2007-11-08 17:52:35 -0600
- (Thu, 08 Nov 2007) | 4 lines Properly say the seconds here..
- Issue 11203, fix described by vma. ........
-
- * pbx/pbx_lua.c: Add check_hangup() method to pbx_lua, which can be
- used to check whether it is time to hangup a channel. Closes
- issue #11202, patch by mnicholson
-
-2007-11-08 22:33 +0000 [r89122-89123] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: app_voicemail failed to build when
- compiling with IMAP_STORAGE Now it does not.
-
- * main/threadstorage.c: AST_LIST_REMOVE_CURRENT takes only one
- argument. Thanks to snuffy for pointing this out on IRC
-
-2007-11-08 21:27 +0000 [r89121] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_env.c: Make func_env build again.
-
-2007-11-08 21:01 +0000 [r89120] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 89119 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov
- 2007) | 7 lines Rework of the commit I made yesterday to use the
- already built-in ast_uri_decode function as opposed to my
- home-rolled one. Also added comments. Thanks to oej for pointing
- me in the right direction ........
-
-2007-11-08 20:39 +0000 [r89118] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_features.c: convert this code to a more efficient
- idiom
-
-2007-11-08 18:49 +0000 [r89116-89117] Jason Parker <jparker@digium.com>
-
- * res/res_smdi.c: Change a warning to a notice. Issue #11195, patch
- by eliel
-
- * /, configs/cdr_adaptive_odbc.conf.sample,
- configs/res_odbc.conf.sample: Merged revisions 89115 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11195) ........ r89115 | qwell | 2007-11-08 12:45:15 -0600
- (Thu, 08 Nov 2007) | 4 lines Avoid warnings on load when using
- sample configuration files. Issue 11195, patch by eliel. ........
-
-2007-11-08 17:32 +0000 [r89113-89114] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_readfile.c, funcs/func_env.c: Add the FILE() dialplan
- function and deprecate ReadFile.
-
- * channels/chan_features.c: Fix missed conversion to linkedlists
- macro change
-
-2007-11-08 16:51 +0000 [r89112] Mark Michelson <mmichelson@digium.com>
-
- * /: Blocking changes from previous 1.4 commit
-
-2007-11-08 09:21 +0000 [r89108-89110] Luigi Rizzo <rizzo@icir.org>
-
- * apps/app_voicemail.c: use %f instead of %lf (the 'l' is ignored
- anyways).
-
- * main/audiohook.c: use %d and cast to int instead of %zd for
- size_t object, this helps portability.
-
- * channels/chan_unistim.c: initialize a variable to silence
- compiler. The type of warnings emitted depends on the
- optimization level, at the lower levels the compiler doesn't
- always understand what the programmer has in mind. In this case I
- could not understand it either.
-
-2007-11-08 05:36 +0000 [r89106-89107] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/srv.c, /: Merged revisions 89105 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89105 | kpfleming | 2007-11-08 00:26:47 -0500 (Thu, 08 Nov 2007)
- | 2 lines fix a glaring bug in the new SRV record handling that
- would cause incorrect weight sorting ........
-
- * main/autoservice.c, main/frame.c, apps/app_meetme.c,
- res/res_features.c, funcs/func_strings.c, main/devicestate.c,
- res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c,
- codecs/codec_zap.c, res/res_jabber.c, main/indications.c,
- main/astobj2.c, main/config.c, main/loader.c, main/cli.c,
- main/cdr.c, main/channel.c, main/manager.c, res/res_agi.c,
- main/logger.c, main/app.c, main/image.c, res/res_speech.c,
- main/sched.c, main/pbx.c, main/translate.c, res/res_crypto.c,
- channels/chan_agent.c, utils/astman.c, apps/app_queue.c,
- channels/iax2-parser.c, main/srv.c,
- include/asterisk/linkedlists.h, main/file.c, pbx/pbx_dundi.c,
- main/event.c, main/audiohook.c, res/res_odbc.c, main/asterisk.c,
- apps/app_voicemail.c: improve linked-list macros in two ways: -
- the *_CURRENT macros no longer need the list head pointer
- argument - add AST_LIST_MOVE_CURRENT to encapsulate the
- remove/add operation when moving entries between lists
-
-2007-11-08 05:00 +0000 [r89104] Tilghman Lesher <tlesher@digium.com>
-
- * /, doc/valgrind.txt: Merged revisions 89103 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007)
- | 2 lines Typo ........
-
-2007-11-08 02:28 +0000 [r89096-89102] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 89101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4
- lines Do not add a sip: to the beginning of the To URI unless
- needed. (closes issue #10756) Reported by: goestelecom ........
-
- * /, channels/chan_sip.c: Merged revisions 89099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6
- lines Improve the devicestate logic for multiple devices. If any
- are available then the extension is considered available. (closes
- issue #10164) Reported by: nic_bellamy Patches:
- sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299)
- ........
-
- * /, channels/chan_sip.c: Merged revisions 89097 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8
- lines Add support for allowing one outgoing transaction. This
- means if a response comes back out of order chan_sip will still
- handle it. I dream of a chan_sip with real transaction support.
- (closes issue #10946) Reported by: flefoll (closes issue #10915)
- Reported by: ramonpeek (closes issue #9567) Reported by:
- atca_pres ........
-
- * /, channels/chan_sip.c: Merged revisions 89095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4
- lines If callerid is configured in sip.conf use that for checking
- the presence of an extension in the dialplan. (closes issue
- #11185) Reported by: spditner ........
-
-2007-11-07 23:47 +0000 [r89094] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 89093 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007)
- | 7 lines The member refcount must be incremented, to avoid using
- it after deallocation. A huge thanks go to lvl- for patiently
- providing the necessary valgrind output that was necessary to
- finding this problem of memory corruption. Reported by: lvl-
- Patch by: tilghman Closes issue #11174 ........
-
-2007-11-07 23:18 +0000 [r89091-89092] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: If imapfolder has been specified in
- voicemail.conf, we should not connect to INBOX... ever. It may
- not exist. (closes issue #11151, reported by selsky, patched by
- me)
-
- * /, channels/chan_sip.c: Merged revisions 89090 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov
- 2007) | 6 lines This patch makes it possible for SIP phones to
- dial extensions defined with '#' characters in extensions.conf
- AND maintain their escaped characters when forming URI's (closes
- issue #10681, reported by cahen, patched by me, code review by
- file) ........
-
-2007-11-07 22:09 +0000 [r89089] Steve Murphy <murf@digium.com>
-
- * /, res/res_jabber.c, cdr/cdr_tds.c: Merged revisions 89088 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1
- line In response to 10578, I just ran 1.4 thru valgrind; some of
- the config leakage I've already fixed, but it doesn't hurt to
- double check. I found and fixed leaks in res_jabber, cdr_tds,
- pbx_ael. Nothing major, tho. ........
-
-2007-11-07 17:45 +0000 [r89086] Joshua Colp <jcolp@digium.com>
-
- * channels/h323/ast_h323.cxx: Minor change so chan_h323 builds
- again.
-
-2007-11-07 13:12 +0000 [r89082-89084] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: remove enter/exit comments when handling subdirectory.
- If we really want them we can remove the --no-print-directory
-
- * main/loader.c: remove a debugging message which i forgot in.
-
- * Makefile: match changes in menuselect's Makefile
-
-2007-11-07 04:21 +0000 [r89077-89081] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_playback.c: Suppress erroneous warnings on load.
- Reported by: eliel Patch by: eliel Closes issue #11177
-
- * /, configs/extensions.ael.sample: Merged revisions 89079 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89079 | tilghman | 2007-11-06 22:07:49 -0600 (Tue, 06 Nov 2007)
- | 5 lines Suppress AEL warnings on load. Reported by: eliel Patch
- by: eliel Closes issue #11178 ........
-
- * channels/chan_zap.c, configs/zapata.conf.sample: Provide the
- ability to directly manipulate the TON/NPI bits in the
- dialstring. Reported by: thetatag Patch by:
- thetatag/stevens/tilghman Closes issue #5331
-
- * contrib/utils/eagi_proxy.c (added): Add contributed EAGI proxy,
- which provides FastAGI functionality for EAGI, while also
- buffering the audio stream. Reported by: devil_slayer Patch by:
- devil_slayer Closes issue #8921
-
-2007-11-07 00:16 +0000 [r89076] Russell Bryant <russell@digium.com>
-
- * main/astmm.c: Fix another CLI command so it doesn't run the real
- code when called for initialization.
-
-2007-11-07 00:04 +0000 [r89075] Mark Michelson <mmichelson@digium.com>
-
- * doc/tex/imapstorage.tex: Adding documentation regarding
- imapfolder, imapgreetings, and greetingsfolder options in
- voicemail.conf (closes issue #11133, reported by selsky, patched
- by blitzrage)
-
-2007-11-07 00:00 +0000 [r89073-89074] Russell Bryant <russell@digium.com>
-
- * include/asterisk/agi.h, res/res_agi.c, CHANGES: Print out the
- channel name as a prefix to the "agi debug" output. This makes
- AGI debugging on busy systems much easier. (closes issue #10730)
- Reported by: junky Patches: agi_debug_chan.diff uploaded by junky
- (license 177) 20070923_10730.diff uploaded by mvanbaak (license
- 7)
-
- * apps/app_meetme.c, CHANGES: Added the ability to do "meetme
- concise" with the "meetme" CLI command. This extends the concise
- capabilities of this CLI command to include listing all
- conferences, instead of an addition to the other sub commands for
- the "meetme" command. (closes issue #11078) Reported by: jthomas
- Patches: meetme-concise.patch uploaded by jthomas (license 293)
-
-2007-11-06 23:08 +0000 [r89072] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: Fix up some PBX logic that became broken. The code
- would exit prematurely when it should have been collecting more
- digits. (closes issue #11175) Reported by: pj
-
-2007-11-06 22:51 +0000 [r89071] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_jingle.c, channels/chan_phone.c,
- codecs/codec_g722.c, main/frame.c, channels/chan_sip.c,
- channels/chan_skinny.c, main/translate.c, channels/chan_h323.c,
- main/file.c, channels/chan_gtalk.c, include/asterisk/frame.h,
- main/rtp.c, channels/chan_mgcp.c, include/asterisk/translate.h:
- Commit some cleanups to the format type code. - Remove the
- AST_FORMAT_MAX_* types, as these are consuming 3 out of our
- available 32 bits. - Add a native slin16 type, so that 16kHz
- codecs can translate without losing resolution. (This doesn't
- affect anything immediately, until another codec has wb support.)
-
-2007-11-06 22:36 +0000 [r89070] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the
- queue strategy wrandom (closes issue #10942, reported and patched
- by julianjm, documentation changes by me)
-
-2007-11-06 22:15 +0000 [r89069] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c, doc/tex/channelvariables.tex, CHANGES: Added
- the S() and L() options to the MeetMe application. These are
- pretty much identical to the S() and L() options to Dial(). They
- let you set timeouts for the conference, as well as have warning
- sounds played to let the caller know how much time is left, and
- when it is running out. (closes issue #8030) Reported by: areski
- Patches: meetme_timeout_timelimit_v2.patch uploaded by areski
- (license 29)
-
-2007-11-06 22:05 +0000 [r89068] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Added CLI and manager commands for changing a
- queue member's penalty (closes issue #9374, reported and
- initially patched by wuwu, intermediate patch by eliel, and final
- patch by me)
-
-2007-11-06 22:01 +0000 [r89067] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c: Add some more locking as well as API update
- for libss7 for new transport types
-
-2007-11-06 21:08 +0000 [r89062] Steve Murphy <murf@digium.com>
-
- * /, main/config.c: Merged revisions 89036 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1
- line closes issue #8786 - where the [catname](!) and
- [catname](othercat1,othercat2,...) notation gets dropped across a
- ConfigUpdate (or any other thing that would cause a config file
- to be written). While I was at it, I also cleaned up some of the
- destroy routines to free up comments, which was not being done.
- Made sure the new struct I introduced is also cleaned up properly
- at destruction time. My code handles multiple template
- inclusions. Many thanks to ssokol for his patch, which, while not
- literally used in the final merge, served as a foundation for the
- fix. ........
-
-2007-11-06 20:55 +0000 [r89057] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Remove native bridging check for DTMF based
- transfers. Thanks to the last batch of RTP changes it is no
- longer required for the media stream to go through Asterisk if
- DTMF is going over signalling. It will simply reinvite back as
- needed. (closes issue #11172) Reported by: ibc
-
-2007-11-06 20:32 +0000 [r89055] Mark Michelson <mmichelson@digium.com>
-
- * res/res_features.c: Instead of trying to callback a local channel
- on a failed attended transfer, call the device that made the
- transfer instead. This makes for much smoother calling back when
- queues are involved. (closes issue #11155, reported by IPetrov)
- Tremendous thanks to Russell for pulling me out of my block I was
- having on this one
-
-2007-11-06 20:22 +0000 [r89052-89054] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 89053 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06
- Nov 2007) | 3 lines Fix init_classes() so that classes that
- actually do have files loaded aren't treated as empty, and
- immediately destroyed ... ........
-
- * main/astmm.c: Fix the memory show allocations CLI command so that
- it doesn't spew out all of the current memory allocations when
- you start Asterisk, when the command's handler gets called for
- initialization.
-
-2007-11-06 19:40 +0000 [r89051] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2f.c, main/ast_expr2.fl: Hoping to avoid a crash in
- OSX for a problem blitzrage found
-
-2007-11-06 19:23 +0000 [r89050] Olle Johansson <oej@edvina.net>
-
- * main/fskmodem.c: Formatting. Illegaly using some spare spaces
- from Russell's space-bucket.
-
-2007-11-06 19:16 +0000 [r89049] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 89045 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06
- Nov 2007) | 2 lines We went to the trouble of creating a method
- of tracking failed trylocks, then never turned it on (oops).
- ........
-
-2007-11-06 19:10 +0000 [r89048] Olle Johansson <oej@edvina.net>
-
- * main/tdd.c, include/asterisk/tdd.h: Additional TDD changes
- (preparing for SIP changes - adding TDD support to SIP)
-
-2007-11-06 19:10 +0000 [r89047] Jason Parker <jparker@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 89046 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4
- lines Correctly set the total number of channels from a zaptel
- transcoder board. SPD-49, patch by Matthew Nicholson. ........
-
-2007-11-06 19:04 +0000 [r89044] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_readfile.c, res/res_features.c, apps/app_sayunixtime.c,
- apps/app_test.c, apps/app_chanisavail.c, res/res_musiconhold.c,
- apps/app_exec.c, apps/app_followme.c, apps/app_minivm.c,
- apps/app_mp3.c, apps/app_amd.c, apps/app_while.c, main/pbx.c,
- apps/app_nbscat.c, channels/chan_sip.c, apps/app_festival.c,
- apps/app_softhangup.c, apps/app_waitforsilence.c,
- channels/chan_agent.c, apps/app_morsecode.c, apps/app_getcpeid.c,
- apps/app_playback.c, res/res_monitor.c, apps/app_speech_utils.c,
- apps/app_forkcdr.c, apps/app_waitforring.c,
- apps/app_directed_pickup.c, apps/app_macro.c, apps/app_sms.c,
- res/res_indications.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
- apps/app_stack.c: "show application <foo>" changes for clarity.
- (closes issue #11171, reported and patched by blitzrage) Many
- thanks!
-
-2007-11-06 19:04 +0000 [r89043] Olle Johansson <oej@edvina.net>
-
- * /, main/tdd.c: Merged revisions 89042 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2
- lines Bug fixes to tdd support in zaptel. ........ (Small changes
- for trunk)
-
-2007-11-06 18:44 +0000 [r89041] Jason Parker <jparker@digium.com>
-
- * channels/chan_jingle.c, include/asterisk/jabber.h,
- channels/chan_gtalk.c, res/res_jabber.c: Allow gtalk and jingle
- to use TLS connections again. Closes issue #9972
-
-2007-11-06 18:23 +0000 [r89038] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 89037 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06
- Nov 2007) | 11 lines If someone were to delete the files used by
- an existing MOH class, and then issue a reload, further use of
- that class could result in a crash due to dividing by zero. This
- set of changes fixes up some places to prevent this from
- happening. (closes issue #10948) Reported by: jcomellas Patches:
- res_musiconhold_division_by_zero.patch uploaded by jcomellas
- (license 282) Additional changes added by me. ........
-
-2007-11-06 17:10 +0000 [r89034] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 89032 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4
- lines Make it so that if a peer is determined to be unreachable
- using qualify their devicestate will report back unavailable.
- (closes issue #11006) Reported by: pj ........
-
-2007-11-06 17:05 +0000 [r89031] Luigi Rizzo <rizzo@icir.org>
-
- * main/loader.c: Fix embedding of modules on FreeBSD: the
- constructor for the list of modules was run after the
- constructors for the embedded modules (which appended entries to
- the list). As a result, the list appeared empty when it was time
- to use it. On linux the order of execution of constructor was
- evidently different (it may depend on the ordering of modules in
- the ELF file). This is only a workaround - there may be other
- situations where the execution of constructors causes problems,
- so if we manage to find a more general solution this workaround
- can go away.
-
-2007-11-06 16:29 +0000 [r88974-88995] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, /, configs/zapata.conf.sample: Merged
- revisions 88994 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6
- lines Fix improbable but possible memory leaks in chan_zap.
- (closes issue #11166) Reported by: eliel Patches:
- chan_zap.c.patch uploaded by eliel (license 64) ........
-
- * channels/chan_agent.c: Update chan_agent documentation. Change a
- | to , as that is now the required way. (closes issue #11167)
- Reported by: eliel Patches: chan_agent.c.patch uploaded by eliel
- (license 64)
-
-2007-11-06 15:01 +0000 [r88973] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_unistim.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Set up detection
- of IP_PKTINFO in autoconf for chan_unistim
-
-2007-11-06 14:17 +0000 [r88932-88937] Russell Bryant <russell@digium.com>
-
- * channels/chan_unistim.c: convert uses of LOG_DEBUG to use
- ast_debug()
-
- * channels/chan_unistim.c, configs/unistim.conf.sample: Add
- jitterbuffer support to chan_unistim. (closes issue #11168)
- Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by
- IgorG (license 20)
-
- * main/pbx.c, /, channels/busy.h, channels/ringtone.h,
- include/asterisk/pbx.h: Merged revisions 88805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) |
- 12 lines After seeing crashes related to channel variables, I
- went looking around at the ways that channel variables are
- handled. In general, they were not handled in a thread-safe way.
- The channel _must_ be locked when reading or writing from/to the
- channel variable list. What I have done to improve this situation
- is to make pbx_builtin_setvar_helper() and friends lock the
- channel when doing their thing. Asterisk API calls almost all
- lock the channel for you as necessary, but this family of
- functions did not. (closes issue #10923, reported by atis)
- (closes issue #11159, reported by 850t) ........
-
- * /, include/asterisk/lock.h: Merged revisions 88931 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06
- Nov 2007) | 8 lines Remove some checks to see if locks are
- initialized from the non-DEBUG_THREADS versions of the lock
- routines. These are incorrect for a number of reasons: - It
- breaks the build on mac. - If there is a problem with locks not
- getting initialized, then the proper fix is to find that place
- and fix the code so that it does get initialized. - If additional
- debug code is needed to help find the problem areas, then this
- type of things should _only_ be put in the DEBUG_THREADS
- wrappers. ........
-
-2007-11-06 08:17 +0000 [r88898-88913] Luigi Rizzo <rizzo@icir.org>
-
- * channels/Makefile: explain that the host environment must be used
- to build gentone; Remove unset variables, they would be
- misleading.
-
- * Makefile: don't export variables that can be retrieved from
- makeopts in child subdirs
-
-2007-11-06 02:53 +0000 [r88863] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, include/asterisk/srv.h: Merged revisions 88862 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r88862 | kpfleming | 2007-11-05 20:52:05 -0600 (Mon, 05
- Nov 2007) | 2 lines update comment to match the state of the code
- ........
-
-2007-11-05 23:31 +0000 [r88827] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 88826 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov
- 2007) | 6 lines Reworked deadlock avoidance in __ast_read.
- Restored audio to callback agents. (closes issue #11071, reported
- by callguy, patched by me, tested by callguy and Ted Brown)
- ........
-
-2007-11-05 21:36 +0000 [r88770] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile, utils/Makefile: Move AUDIO_LIBS outside the top level
- Makefile. This too is used only in one place.
-
-2007-11-05 21:35 +0000 [r88769] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 88768 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) |
- 8 lines When traversing the list of channel variables here in
- transmit_invite(), the asterisk channel must be locked, as this
- data may change at any time. (I have seen numerous reports of
- crashes related to the handling of channel variables. There are a
- couple of issues on the bug tracker related to it, but it has
- also been noted on IRC and mailing lists. So, I am finding and
- fixing some places where channel variables are handled
- improperly.) ........
-
-2007-11-05 21:27 +0000 [r88767] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile, main/Makefile: Move the last instance of AST_LIBS to
- the only place it is used, namely main/Makefile . I am unclear
- where decisions on the build environment (CFLAGS, LDFLAGS, LIBS
- and so on) should be made - right now they are split here and
- there. As a first step in cleaning up this situation, i am trying
- to at least collect all instances of each variable in one place.
-
-2007-11-05 21:23 +0000 [r88766] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 88765 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) |
- 2 lines Fix up some indentation. ........
-
-2007-11-05 20:50 +0000 [r88764] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile.moddir_rules: comment out an unused variable. Remove it
- in a few days if no problems arise.
-
-2007-11-05 20:44 +0000 [r88710-88740] Russell Bryant <russell@digium.com>
-
- * main/srv.c, /, include/asterisk/srv.h: Merged revisions 88719 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) |
- 7 lines Merge changes from
- asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
- record support in Asterisk was broken. There was no guarantee on
- what record Asterisk would choose to actually use. This set of
- changes improves the situation by ensuring that Asterisk will
- choose the highest priority record. ........
-
- * main/channel.c, /: Merged revisions 88709 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) |
- 20 lines Merge the last bit of changes from
- asterisk/team/russell/readq-1.4 The issue here is that the
- channel frame readq handling got broken when the code was
- converted to use the linked list macros. It caused corruption of
- the list head and tail pointers. So, I fixed up the usage of the
- linked list macros and in passing, simplified the code. I also
- documented what the code is doing, as it was a bit difficult to
- figure out at first. This bug showed itself with crashes showing
- messed up head/tail pointers for the readq. However, there are a
- couple of crashes that aren't quite as obvious, but I think may
- be related. So, if your bug gets closed by this commit, but you
- still have a problem, please reopen or create a new bug report.
- (closes issue #10936) (closes issue #10595) (closes issue #10368)
- (closes issue #11084) (closes issue #10040) (closes issue #10840)
- ........
-
-2007-11-05 19:22 +0000 [r88675] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: Cleanup the installation of samples, avoiding
- repetitions. I am preserving the behaviour on *.adsi files, i.e.
- overwrite anything there without making a backup. However I am
- not sure that this is the intended behaviour.
-
-2007-11-05 18:52 +0000 [r88673] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 88671 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7
- lines If a SIP channel is put on hold multiple times do not keep
- incrementing the onHold value. (closes issue #11085) Reported by:
- francesco_r Tested by: blitzrage (closes issue #10474) Reported
- by: acennami ........
-
-2007-11-05 18:22 +0000 [r88653] Tilghman Lesher <tlesher@digium.com>
-
- * CHANGES: Change wording to that suggested by MasterYoda
-
-2007-11-05 18:00 +0000 [r88652] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: simplify (hopefully) the printing of $(MAKE) in aligned
- output.
-
-2007-11-05 17:52 +0000 [r88651] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 88624 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) |
- 5 lines Fix up datastore handling in ast_do_masquerade(). The
- code is intended to move any channel datastores from the old
- channel to the new one. However, it did not use the linked list
- macros properly to accomplish the task. The existing code would
- only work if there was only a single datastore on the old
- channel. ........
-
-2007-11-05 17:44 +0000 [r88587-88615] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile: print messages when entering/leaving a directory so we
- know where we are (sometimes it is obvious, sometimes it is not).
-
- * Makefile.moddir_rules: merge two rules with the same right hand;
- document a bit what is done here.
-
-2007-11-05 17:21 +0000 [r88586] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 88585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11163) ........ r88585 | qwell | 2007-11-05 11:19:41 -0600
- (Mon, 05 Nov 2007) | 4 lines Make sure we destroy the config
- structure on configuration failure. Issue 11163, patch by eliel.
- ........
-
-2007-11-05 17:00 +0000 [r88584] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile.rules: use a variable name that actually indicates what
- it is for
-
-2007-11-05 16:41 +0000 [r88553] Luigi Rizzo <rizzo@icir.org>
-
- * Makefile.rules: Put extra compiler flags into a variable so they
- are not repeated too many times. On passing, add some comments
- and fix indentation a bit. On passing, i suspect that the
- following pattern is wrong %.eoo: %.o but in case it will be
- fixed in a later commit.
-
-2007-11-05 16:30 +0000 [r88540] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_odbc.c: Merged revisions 88539 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88539 | tilghman | 2007-11-05 10:20:13 -0600 (Mon, 05 Nov 2007)
- | 4 lines Don't check used pooled connections for connection
- status, as it will cause issues for prepared queries. Reported
- by: Nick Gorham (via -dev list) Patch by: tilghman ........
-
-2007-11-05 15:15 +0000 [r88525] Luigi Rizzo <rizzo@icir.org>
-
- * main/db.c: remove a cygwin-specific function remap that does not
- work.
-
-2007-11-05 13:11 +0000 [r88510] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_unistim.c: Fix memory leaks and deadlocks in
- chan_unistim. (closes issue #11158) Reported by: eliel Patches:
- chan_unistim.c.patch uploaded by eliel (license 64)
-
-2007-11-04 22:42 +0000 [r88454-88490] Luigi Rizzo <rizzo@icir.org>
-
- * /: block merging of not-applicable patch
-
- * main/channel.c, main/pbx.c, apps/app_meetme.c,
- channels/chan_sip.c, res/res_features.c, main/utils.c,
- channels/chan_iax2.c, include/asterisk/stringfields.h: Simplify
- the implementation and the API for stringfields; details and
- examples are in include/asterisk/stringfields.h. Not applicable
- to older branches except for 1.4 which will receive a fix for the
- routines that free memory pools.
-
-2007-11-03 14:19 +0000 [r88437] Tilghman Lesher <tlesher@digium.com>
-
- * main/term.c: Revert commit #86119. Some users intentionally do
- not want colorized terminals, so this was a misfeature.
-
-2007-11-03 04:55 +0000 [r88422] James Golovich <james@gnuinter.net>
-
- * main/db.c: Set CLI command to the correct name. Rev 85460
- introduced two 'database show' commands when this one should have
- been 'database showkey'
-
-2007-11-02 22:36 +0000 [r88368-88409] Russell Bryant <russell@digium.com>
-
- * channels/chan_unistim.c: fix some issues with crashing on unload,
- when it didn't completely load cleanly
-
- * channels/chan_unistim.c: Convert the CLI commands to the new
- format
-
- * pbx/pbx_lua.c: propagate the DECLINE return value back to the
- loader
-
- * pbx/pbx_lua.c: Don't kill asterisk if extensions.lua is not
- present.
-
- * main/cli.c: Show the channel unique ID in the "show channel
- concise" output (closes issue #11148, requested by falves11,
- patched by me)
-
- * channels/chan_unistim.c (added), CREDITS,
- configs/unistim.conf.sample (added), CHANGES, doc/unistim.txt
- (added): Merge the code from asterisk/team/group/chan_unistim:
- This introduces a new channel driver, chan_unistim, that supports
- the Unistim VoIP protocol for Nortel phones. The following models
- have been confirmed to work: i2002, i2004 and i2050. (closes
- issue #8864) Reported by: c_hans Patches: chan_unistim.patch
- uploaded by c (license 304) ustm_no_conf.diff uploaded by junky
- (license 177) Tested by: c_hans, dbowerman, math, junky, loloski
-
-2007-11-02 20:51 +0000 [r88329-88367] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 88366 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4
- lines Make subscribecontext behave as advertised. It will now
- look for the presence of a hint in the given context (be it
- subscribecontext or context). (closes issue #10702) Reported by:
- slavon ........
-
- * /, channels/chan_sip.c: Merged revisions 88328 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6
- lines If an INFO request within a dialog is received with a
- content length of 0 simply send back a 200 OK. It is valid to do
- this and the remote side is probably using it to make sure the
- signalling is still alive. (closes issue #5747) Reported by:
- chandi Patches: infofix-81430-1.patch uploaded by IgorG (license
- 20) ........
-
-2007-11-02 20:13 +0000 [r88327] Russell Bryant <russell@digium.com>
-
- * doc/tex/Makefile: Fix replacing the version number when it has a
- '/' in it, like SVN-group-chan_unistim-r88326M-/trunk
-
-2007-11-02 17:34 +0000 [r88287] Tilghman Lesher <tlesher@digium.com>
-
- * pbx/pbx_lua.c: Oops, some dev-mode changes for ISO C90
-
-2007-11-02 16:54 +0000 [r88284] Jason Parker <jparker@digium.com>
-
- * /, main/say.c: Merged revisions 88283 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11147) ........ r88283 | qwell | 2007-11-02 11:51:08 -0500
- (Fri, 02 Nov 2007) | 4 lines We need to make sure to specify a
- language to ast_fileexists, otherwise it may fail for anything
- besides en Issue 11147, fix discovered by both citats and myself
- (independently), with input from Corydon76 ........
-
-2007-11-02 16:26 +0000 [r88209-88267] Tilghman Lesher <tlesher@digium.com>
-
- * CHANGES: Add a few bytes on LUA
-
- * main/pbx.c, utils/build-extensions-conf.lua (added),
- build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c
- (added), configs/extensions.lua.sample (added),
- include/asterisk/pbx.h, makeopts.in: Add pbx_lua as a method of
- doing extensions Reported by: mnicholson Patch by: mnicholson
- Closes issue #11140
-
- * main/config.c: Don't re-cache the filename, but check to see if
- it already exists Reported by: jamesgolovich Patch by:
- jamesgolovich Closes issue #11144
-
- * /, include/asterisk/lock.h: Merged revisions 88210 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r88210 | tilghman | 2007-11-02 08:03:03 -0500 (Fri, 02
- Nov 2007) | 5 lines Fix build on Solaris Reported by: snuffy
- Patch by: ys Closes issue #11143 ........
-
- * main/pbx.c: 'h' extension doesn't execute past first priority
- Reported by: dimas Patch by: dimas Closes bug #11146
-
-2007-11-02 03:09 +0000 [r88197] Joshua Colp <jcolp@digium.com>
-
- * cdr/cdr_odbc.c: Restore building under 64-bit platforms.
-
-2007-11-01 23:26 +0000 [r88184] Jason Parker <jparker@digium.com>
-
- * channels/chan_jingle.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/jabber.h, channels/chan_gtalk.c, makeopts.in:
- Remove traces of gnutls, since we no longer use/need it.
-
-2007-11-01 23:26 +0000 [r88182-88183] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: Modify WaitExten to include an optional dialtone
- Closes issue #10783
-
- * UPGRADE.txt, cdr/cdr_odbc.c: Convert cdr_odbc to use res_odbc
- managed connections Closes issue #10614
-
-2007-11-01 22:26 +0000 [r88166] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c,
- funcs/func_strings.c, funcs/func_cut.c, funcs/func_logic.c,
- apps/app_exec.c, apps/app_queue.c, apps/app_playback.c,
- res/ael/pval.c, pbx/pbx_loopback.c, funcs/func_odbc.c,
- apps/app_minivm.c, res/res_agi.c, main/logger.c,
- pbx/pbx_realtime.c, apps/app_macro.c, pbx/pbx_dundi.c,
- utils/extconf.c, include/asterisk/pbx.h, pbx/pbx_config.c,
- apps/app_mixmonitor.c, apps/app_rpt.c, cdr/cdr_custom.c,
- cdr/cdr_manager.c: This commits the performance mods that give
- the priority processing engine in the pbx, a 25-30% speed boost.
- The two updates used, are, first, to merge the
- ast_exists_extension() and the ast_spawn_extension() where they
- are called sequentially in a loop in the code, into a slightly
- upgraded version of ast_spawn_extension(), with a few extra args;
- and, second, I modified the substitute_variables_helper_full, so
- it zeroes out the byte after the evaluated string instead of
- demanding you pre-zero the buffer; I also went thru the code and
- removed the code that zeroed this buffer before every call to the
- substitute_variables_helper_full. The first fix provides about a
- 9% speedup, and the second the rest. These figures come from the
- 'PIPS' benchmark I describe in blogs, conf. reports, etc.
-
-2007-11-01 22:19 +0000 [r88164-88165] Jason Parker <jparker@digium.com>
-
- * /: Crap, accidentally copied the props. Thanks for pointing this
- out mvanbaak. The odds are quite high that this will break
- automerge on every team branch.
-
- * /, include/asterisk/jabber.h, res/res_jabber.c: Switch res_jabber
- to use openssl rather than gnutls. Closes issue #9972, patch by
- phsultan. Copied from branch at
- http://svn.digium.com/svn/asterisk/team/phsultan/res_jabber-openssl/
-
-2007-11-01 17:25 +0000 [r88117] Tilghman Lesher <tlesher@digium.com>
-
- * /, doc/valgrind.txt (added): Merged revisions 88116 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r88116 | tilghman | 2007-11-01 12:17:56 -0500 (Thu, 01
- Nov 2007) | 2 lines Add some notes on using valgrind ........
-
-2007-11-01 16:22 +0000 [r88079] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 88078 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4
- lines Make sure we set the poll fds to NULL after free()ing it.
- Part of issue 11017, patch by tzafrir. ........
-
-2007-11-01 15:56 +0000 [r88062-88077] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c, pbx/pbx_dundi.c: Change some uses of free()
- to ast_free(). (No functional differences.) (closes issue #11138)
- Reported by: eliel Patches: pbx_dundi.c.patch uploaded by eliel
- (license 64) chan_sip.c.patch uploaded by eliel (license 64)
-
- * utils/Makefile: Remove another copied source file on "make
- clean". (closes issue #11137) Reported by: IgorG Patches:
- addonclean-87971-1.patch uploaded by IgorG (license 20)
-
-2007-11-01 13:30 +0000 [r88027] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 88026 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r88026 | file | 2007-11-01 10:27:37 -0300 (Thu, 01 Nov 2007) | 2
- lines Fix up commit for my Zap channel with spies in Meetme fix.
- (thanks Tony Mountifield!) ........
-
-2007-11-01 06:12 +0000 [r88007-88010] Tilghman Lesher <tlesher@digium.com>
-
- * main/utils.c: Conditionally free lock_info->thread_name to avoid
- a useless warning Reported by: snuffy Patch by: snuffy Closes
- issue #11125
-
- * apps/app_meetme.c, channels/chan_iax2.c: Janitor: use ast_free to
- pair calls of ast_malloc and ast_calloc Reported by: eliel Patch
- by: eliel Closes issue #11135
-
- * cdr/cdr_adaptive_odbc.c: Fix memory leak Reported by: eliel Fixed
- by: tilghman Closes issue #11136
-
-2007-11-01 01:55 +0000 [r87953-87971] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 87970 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4
- lines If a Zap channel contains a spy or a spy is added take it
- out of the conference in kernel space and make it go through
- Asterisk so the spy gets audio from both sides. (closes issue
- #10060) Reported by: mparker ........
-
- * main/pbx.c: Drop any more references to type in the Exception
- dialplan function. (closes issue #11134) Reported by: blitzrage
- Patches: exception_patch.txt uploaded by blitzrage (license 10)
-
-2007-10-31 21:23 +0000 [r87889-87909] Jason Parker <jparker@digium.com>
-
- * /, res/res_jabber.c: Merged revisions 87908 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11131) ........ r87908 | qwell | 2007-10-31 16:23:11 -0500
- (Wed, 31 Oct 2007) | 4 lines Make sure we free some allocated
- memory before returning. Issue 11131, patch by eliel. ........
-
- * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged
- revisions 87906 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
- issue #11130) (closes issue #11132) ........ r87906 | qwell |
- 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try
- to allocate memory that we're just going to re-allocate later
- anyways. Issues 11130 and 11132, patch by eliel. ........
-
- * formats/format_sln.c, codecs/codec_adpcm.c, codecs/codec_gsm.c,
- formats/format_wav_gsm.c, res/res_musiconhold.c,
- codecs/codec_zap.c, formats/format_ilbc.c, res/res_smdi.c,
- formats/format_pcm.c, formats/format_h263.c,
- formats/format_h264.c, formats/format_jpeg.c,
- formats/format_gsm.c, res/res_speech.c, res/res_clioriginate.c,
- codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c,
- formats/format_wav.c, codecs/codec_speex.c, codecs/codec_alaw.c,
- res/res_adsi.c, res/res_convert.c, codecs/codec_g726.c,
- formats/format_ogg_vorbis.c, res/res_ael_share.c,
- formats/format_vox.c, codecs/codec_ulaw.c, formats/format_g723.c,
- res/res_indications.c, codecs/codec_ilbc.c,
- formats/format_g726.c, formats/format_g729.c: More changes to
- change return values from load_module functions. (issue #11096)
- Patches: codec_adpcm.c.patch uploaded by moy (license 222)
- codec_alaw.c.patch uploaded by moy (license 222)
- codec_a_mu.c.patch uploaded by moy (license 222)
- codec_g722.c.patch uploaded by moy (license 222)
- codec_g726.c.diff uploaded by moy (license 222) codec_gsm.c.patch
- uploaded by moy (license 222) codec_ilbc.c.patch uploaded by moy
- (license 222) codec_lpc10.c.patch uploaded by moy (license 222)
- codec_speex.c.patch uploaded by moy (license 222)
- codec_ulaw.c.patch uploaded by moy (license 222)
- codec_zap.c.patch uploaded by moy (license 222)
- format_g723.c.patch uploaded by moy (license 222)
- format_g726.c.patch uploaded by moy (license 222)
- format_g729.c.patch uploaded by moy (license 222)
- format_gsm.c.patch uploaded by moy (license 222)
- format_h263.c.patch uploaded by moy (license 222)
- format_h264.c.patch uploaded by moy (license 222)
- format_ilbc.c.patch uploaded by moy (license 222)
- format_jpeg.c.patch uploaded by moy (license 222)
- format_ogg_vorbis.c.patch uploaded by moy (license 222)
- format_pcm.c.patch uploaded by moy (license 222)
- format_sln.c.patch uploaded by moy (license 222)
- format_vox.c.patch uploaded by moy (license 222)
- format_wav.c.patch uploaded by moy (license 222)
- format_wav_gsm.c.patch uploaded by moy (license 222)
- res_adsi.c.patch uploaded by eliel (license 64)
- res_ael_share.c.patch uploaded by eliel (license 64)
- res_clioriginate.c.patch uploaded by eliel (license 64)
- res_convert.c.patch uploaded by eliel (license 64)
- res_indications.c.patch uploaded by eliel (license 64)
- res_musiconhold.c.patch uploaded by eliel (license 64)
- res_smdi.c.patch uploaded by eliel (license 64)
- res_speech.c.patch uploaded by eliel (license 64)
-
-2007-10-31 18:53 +0000 [r87888] Steve Murphy <murf@digium.com>
-
- * /: Merged revisions 87849 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87849 | murf | 2007-10-31 11:49:39 -0600 (Wed, 31 Oct 2007) | 1
- line closes issue #11108 -- where the 'dialplan save' cli command
- saves a file where the semicolon is not escaped. Fixed this; User
- also wanted comments to be preserved across dialplan save, but
- this is impossible at this point in time, because comments are
- not stored in the dialplan. They are 'compiled' out of
- extensions.conf. The only way to preserve those comments is to
- use the config file reader/writer that the GUI uses to allow
- online user edits. extensions.conf is first and foremost, a
- config file, and is read in by the normal config-file reading
- routines. Then, it is processed into a dialplan (context/exten
- structs). (in the case of trunk, tho, no mods needed to be made
- -- works OK there -- just make sure you use ',' to sep app args!)
- ........
-
-2007-10-31 18:09 +0000 [r87854] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile, /: Merged revisions 87852 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87852 | tilghman | 2007-10-31 13:03:53 -0500 (Wed, 31 Oct 2007)
- | 2 lines Create samples for ALL of the available options in
- asterisk.conf ........
-
-2007-10-31 18:03 +0000 [r87833-87851] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c: Add volume adjustment in.
-
- * apps/app_mixmonitor.c: Restore operation of the option that only
- writes when the channel is bridged.
-
- * apps/app_chanspy.c: Add volume adjustment to spy audiohook in
- app_chanspy.
-
-2007-10-31 16:13 +0000 [r87817] Tilghman Lesher <tlesher@digium.com>
-
- * CREDITS: Formatting cleanups, remove obsolete contributions
- (modules no longer in Asterisk), and obfuscate email addresses
- enough to stop most spam harvesters.
-
-2007-10-31 16:07 +0000 [r87815] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/channel.h: Remove old whisper remnants from
- channel.h
-
-2007-10-31 15:46 +0000 [r87811] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: Optimize pbx_substitute_variables
-
-2007-10-31 04:20 +0000 [r87776] Steve Murphy <murf@digium.com>
-
- * res/ael/pval.c, /: Merged revisions 87775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1
- line Included some verbage in the check_includes func, to inform
- the user that included contexts that have no match in the AEL,
- might be OK, as AEL cannot check in the extensions.conf or the
- in-memory contexts, as they may not be there at the time of the
- check. ........
-
-2007-10-30 23:08 +0000 [r87724-87740] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 87739 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r87739 | tilghman | 2007-10-30 18:02:22 -0500 (Tue, 30
- Oct 2007) | 5 lines Fix for uninitialized mutexes on *BSD
- Reported by: ys Fixed by: ys Closes issue #11116 ........
-
- * apps/app_exec.c: If no '?' is found in the arguments