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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-07-28 14:34:46 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-07-28 14:34:46 +0000
commit04a70b44676380476271983ce5ac2d04ca8863c6 (patch)
treee3655bc8a8f52042898dc9613d322b72ae3e1361
parent065432d7e7f1a667933c52ba260695381b5e804f (diff)
Importing files for 1.6.1.3-rc1 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.1.3-rc1@209435 f38db490-d61c-443f-a65b-d21fe96a405b
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+1.6.1.3-rc1
diff --git a/ChangeLog b/ChangeLog
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+2009-07-28 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.3-rc1
+
+2009-07-28 00:19 +0000 [r209327] Tilghman Lesher <tlesher@digium.com>
+
+ * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009)
+ | 9 lines Merged revisions 209315 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
+ | 2 lines Publish French extra sounds ........ ................
+
+2009-07-27 21:44 +0000 [r209262-209281] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 |
+ kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7
+ lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE
+ messages about T.38 negotiation in debug level 1 messages, clean
+ up some looping logic, and correct an improper use of ast_free()
+ for freeing an ast_frame. ........
+
+ * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 |
+ kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10
+ lines Make T.38 switchover in ReceiveFAX synchronous. In receive
+ mode, if the channel that ReceiveFAX is running on supports T.38,
+ we should *always* attempt to switch T.38, rather than listening
+ for an incoming CNG tone and only triggering on that. The channel
+ may be using a low-bitrate codec that distorts the CNG tone, the
+ sending FAX endpoint may not send CNG at all, or there could be a
+ variety of other reasons that we don't detect it, but in all
+ those cases if T.38 is available we certainly want to use it.
+ ........
+
+2009-07-27 20:57 +0000 [r209237] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /: Merged revisions 209235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 |
+ mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5
+ lines Gracefully handle malformed RTP text packets. AST-2009-004
+ ........
+
+2009-07-27 20:28 +0000 [r209233] David Brooks <dbrooks@digium.com>
+
+ * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c,
+ channels/chan_vpb.cc, res/res_smdi.c, /,
+ include/asterisk/module.h, main/features.c, res/res_agi.c: Merged
+ revisions 209098 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 |
+ dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
+ Fixing typos. Replaces "recieved" with "received" and "initilize"
+ with "initialize" (closes issue #15571) Reported by: alecdavis
+ ........
+
+2009-07-27 20:17 +0000 [r209134-209199] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
+ 2009) | 9 lines Honor channel's music class when using realtime
+ music on hold. (closes issue #15051) Reported by: alexh Patches:
+ 15051.patch uploaded by mmichelson (license 60) Tested by: alexh
+ ........
+
+ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
+ 209132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
+ 2009) | 24 lines Merged revisions 209131 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
+ 2009) | 18 lines Allow for UDPTL to use only even-numbered ports
+ if desired. There are some VoIP providers out there that will not
+ accept SDP offers with odd numbered UDPTL ports. While it is my
+ personal opinion that these VoIP providers are misinterpreting
+ RFC 2327, it really is not a big deal to play along with their
+ silly little games. Of course, since restricting UDPTL ports to
+ only even numbers reduces the range of available ports by half,
+ so the option to use only even port numbers is off by default. A
+ user can enable the behavior by setting use_even_ports=yes in
+ udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
+ 15182.patch uploaded by mmichelson (license 60) Tested by:
+ CGMChris ........ ................
+
+2009-07-27 15:40 +0000 [r209058] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 209056 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 |
+ kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10
+ lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
+ underscore-variants to sub-makes. During the recent Makefile
+ improvements I made, it seemed the 'make' was automatically
+ carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
+ I removed the explict export of them. However, there are some
+ circumstances where make does this, and some where it does not,
+ so I've brought them back to ensure they are always exported. I
+ also removed an extraneous double setting of _ASTLDFLAGS on *BSD
+ platforms. ........
+
+2009-07-27 01:22 +0000 [r208926] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_iax2.c, /, main/translate.c: Merged revisions
+ 208924 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009)
+ | 9 lines Merged revisions 208923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
+ | 2 lines Fix logic errors from 208746 ........ ................
+
+2009-07-26 14:04 +0000 [r208888] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208886 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26
+ Jul 2009) | 2 lines add OpenBSD to the install_prereq script
+ ........
+
+2009-07-25 06:25 +0000 [r208754] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_iax2.c, /, channels/chan_skinny.c,
+ main/translate.c: Merged revisions 208749 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009)
+ | 13 lines Merged revisions 208746 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
+ | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
+ trivial changes, but I did not know of any other way to fix the
+ "dereferencing type-punned pointer will break strict-aliasing
+ rules" error without creating a tmp variable in chan_skinny.
+ ........ ................
+
+2009-07-24 18:52 +0000 [r208595] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009)
+ | 14 lines Merged revisions 208592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
+ | 7 lines Do not log an ERROR if autoservice_stop() returns -1.
+ This does not indicate an error. A return of -1 just means that
+ the channel has been hung up. (reported in #asterisk-dev)
+ ........ ................
+
+2009-07-24 18:32 +0000 [r208590] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul
+ 2009) | 16 lines Merged revisions 208587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
+ 2009) | 10 lines Only send a BYE when hanging up a channel that
+ is up. For cases where Asterisk sends an INVITE and receives a
+ non 2XX final response, Asterisk would follow the INVITE
+ transaction by immediately sending a BYE, which was unnecessary.
+ (closes issue #14575) Reported by: chris-mac ........
+ ................
+
+2009-07-24 15:05 +0000 [r208550] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 208548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 |
+ kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8
+ lines Resolve a T.38 negotiation issue left over from the
+ udptl-updates merge. The udptl-updates branch that was merged
+ yesterday failed to properly send back T.38 SDP responses with
+ the correct error correction mode, if the incoming SDP from the
+ other end caused us to change error correction modes. This patch
+ corrects that situation. ........
+
+2009-07-24 14:38 +0000 [r208544] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208542 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24
+ Jul 2009) | 13 lines use aptitude for debian based systems The
+ function to check wether we need to install packages was using
+ dpkg-query which was gives wrong output on Debian 5 Also, the
+ apt-get has been replaced with aptitude because aptitude is now
+ the preferred way to handle packages on Debian (closes issue
+ #15570) Reported by: mvanbaak Patches:
+ 2009072400_installprereq-aptitude.diff uploaded by mvanbaak
+ (license 7) ........
+
+2009-07-23 22:32 +0000 [r208484-208503] Kevin P. Fleming <kpfleming@digium.com>
+
+ * UPGRADE.txt: Use correct formatting for T.38 change note in
+ UPGRADE.txt
+
+ * include/asterisk/frame.h, main/rtp.c, main/channel.c,
+ main/udptl.c, main/frame.c, /, channels/chan_sip.c,
+ apps/app_fax.c, UPGRADE.txt, include/asterisk/udptl.h: Merged
+ revisions 208464 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 |
+ kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46
+ lines Rework of T.38 negotiation and UDPTL API to address
+ interoperability problems Over the past couple of months, a
+ number of issues with Asterisk negotiating (and successfully
+ completing) T.38 sessions with various endpoints have been found.
+ This patch attempts to address many of them, primarily focused
+ around ensuring that the endpoints' MaxDatagram size is honored,
+ and in addition by ensuring that T.38 session parameter
+ negotiation is performed correctly according to the ITU T.38
+ Recommendation. The major changes here are: 1) T.38 applications
+ in Asterisk (app_fax) only generate/receive IFP packets, they do
+ not ever work with UDPTL packets. As a result of this, they
+ cannot be allowed to generate packets that would overflow the
+ other endpoints' MaxDatagram size after the UDPTL stack adds any
+ error correction information. With this patch, the application is
+ told the maximum *IFP* size it can generate, based on a
+ calculation using the far end MaxDatagram size and the active
+ error correction mode on the T.38 session. The same is true for
+ sending *our* MaxDatagram size to the remote endpoint; it is
+ computed from the value that the application says it can accept
+ (for a single IFP packet) combined with the active error
+ correction mode. 2) All treatment of T.38 session parameters as
+ 'capabilities' in chan_sip has been removed; these parameters are
+ not at all like audio/video stream capabilities. There are strict
+ rules to follow for computing an answer to a T.38 offer, and
+ chan_sip now follows those rules, using the desired parameters
+ from the application (or channel) that wants to accept the T.38
+ negotiation. 3) chan_sip now stores and forwards
+ ast_control_t38_parameters structures for tracking 'our' and
+ 'their' T.38 session parameters; this greatly simplifies
+ negotiation, especially for pass-through calls. 4) Since T.38
+ negotiation without specifying parameters or receiving the final
+ negotiated parameters is not very worthwhile, the AST_CONTROL_T38
+ control frame has been removed. A note has been added to
+ UPGRADE.txt about this removal, since any out-of-tree
+ applications that use it will no longer function properly until
+ they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
+ https://reviewboard.asterisk.org/r/310/ ........
+
+2009-07-23 20:45 +0000 [r208459] David Brooks <dbrooks@digium.com>
+
+ * apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing
+ typos "recieved" with "received". (closes issue #15360) Reported
+ by: okrief
+
+2009-07-23 19:35 +0000 [r208390] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul
+ 2009) | 24 lines Merged revisions 208386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
+ 2009) | 17 lines Fix a problem where a 491 response could be sent
+ out of dialog. This generalizes the fix for issue 13849. The
+ initial fix corrected the problem that Asterisk would reply with
+ a 491 if a reinvite were received from an endpoint and we had not
+ yet received an ACK from that endpoint for the initial INVITE it
+ had sent us. This expansion also allows Asterisk to appropriately
+ handle an INVITE with authorization credentials if Asterisk had
+ not received an ACK from the previous transaction in which
+ Asterisk had responded to an unauthorized INVITE with a 407.
+ (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
+ uploaded by mmichelson (license 60) Tested by: klaus3000 ........
+ ................
+
+2009-07-23 19:24 +0000 [r208385] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500
+ (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009)
+ | 6 lines Only set the priindication setting when not performing
+ a reload (closes issue #14696) Reported by: fdecher ........
+ ................
+
+2009-07-23 16:30 +0000 [r208265-208318] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul
+ 2009) | 9 lines Merged revisions 208312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
+ 2009) | 3 lines Remove inaccurate XXX comment. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul
+ 2009) | 15 lines Merged revisions 208262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
+ 2009) | 8 lines Properly handle 183 responses which do not
+ contain an SDP. (closes issue #15442) Reported by: ffloimair
+ Patches: 15442.patch uploaded by mmichelson (license 60) Tested
+ by: tkarl, ffloimair ........ ................
+
+2009-07-22 21:45 +0000 [r208115] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 208113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 |
+ qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines
+ Restore an int declaration on PPC platforms. This x is one crafty
+ little bugger... It was used for 2 different things (one of which
+ was only done on PPC) in 1.4. One of the uses were removed in
+ trunk, and with it went the declaration. (closes issue #14038)
+ Reported by: ffloimair ........
+
+2009-07-21 22:48 +0000 [r207948] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500
+ (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009)
+ | 8 lines Force an error if a blank is passed to QUOTE (because
+ the documentation states the argument is not optional). This
+ change makes URIENCODE and QUOTE behave similarly, since the
+ documentation states that the argument is not optional, for both.
+ (closes issue #15439) Reported by: pkempgen Patches:
+ 20090706__issue15439.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2009-07-21 20:29 +0000 [r207784-207861] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500
+ (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009)
+ | 9 lines Wait for wink before dialing when using E&M wink
+ signaling There was already code for other signaling types in
+ dahdi_handle_event to handle dialing if a dial operation dial
+ string was present. Simply add SIG_EMWINK to the list. (closes
+ issue #14434) Reported by: araasch ........ ................
+
+ * channels/chan_dahdi.c: Revert r207637, this approach could
+ potentially block for an unacceptable amount of time.
+
+2009-07-21 14:31 +0000 [r207726] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Merged revisions 207723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul
+ 2009) | 11 lines Merged revisions 207714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul
+ 2009) | 5 lines Document default timeout for AMI originations.
+ AST-224 ........ ................
+
+2009-07-21 13:48 +0000 [r207684] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/Makefile, doc/video_console.txt, Makefile, agi/Makefile,
+ codecs/Makefile, utils/Makefile, funcs/Makefile,
+ codecs/lpc10/Makefile, main/db1-ast/Makefile, /, main/Makefile,
+ codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules,
+ pbx/Makefile, res/Makefile: Merged revisions 207680 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500
+ (Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul
+ 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are
+ honored. This commit changes the build system so that
+ user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to
+ the compiler/linker *after* all flags provided by the build
+ system itself, so that the user can effectively override the
+ build system's flags if desired. In addition, ASTCFLAGS and
+ ASTLDFLAGS can now be provided *either* in the environment before
+ running 'make', or as variable assignments on the 'make' command
+ line. As a result, the use of COPTS and LDOPTS is no longer
+ necessary, so they are no longer documented, but are still
+ supported so as not to break existing build systems that supply
+ them when building Asterisk. ........ ................
+
+2009-07-21 04:45 +0000 [r207637] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Wait for wink before dialing when using
+ E&M wink signaling This patch adds a new dahdi_wait function to
+ specifically wait for the wink event. If the wink is not
+ eventually received the channel is hung up. (closes issue #14434)
+ Reported by: araasch Patches: emwinkmod uploaded by araasch
+ (license 693)
+
+2009-07-20 20:02 +0000 [r207426] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul
+ 2009) | 39 lines Merged revisions 207423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul
+ 2009) | 33 lines Answer video SDP offers properly when
+ videosupport is not enabled. Copied from Review board: In issue
+ 12434, the reporter describes a situation in which audio and
+ video is offered on the call, but because videosupport is
+ disabled in sip.conf, Asterisk gives no response at all to the
+ video offer. According to RFC 3264, all media offers should have
+ a corresponding answer. For offers we do not intend to actually
+ reply to with meaningful values, we should still reply with the
+ port for the media stream set to 0. In this patch, we take note
+ of what types of media have been offered and save the information
+ on the sip_pvt. The SDP in the response will take into account
+ whether media was offered. If we are not otherwise going to
+ answer a media offer, we will insert an appropriate m= line with
+ the port set to 0. It is important to note that this patch is
+ pretty much a bandage being applied to a broken bone. The patch
+ *only* helps for situations where video is offered but
+ videosupport is disabled and when udptl_pt is disabled but T.38
+ is offered. Asterisk is not guaranteed to respond to every media
+ offer. Notable cases are when multiple streams of the same type
+ are offered. The 2 media stream limit is still present with this
+ patch, too. In trunk and the 1.6.X branches, things will be a bit
+ different since Asterisk also supports text in SDPs as well.
+ (closes issue #12434) Reported by: mnnojd Review:
+ https://reviewboard.asterisk.org/r/311 Review:
+ https://reviewboard.asterisk.org/r/313 ........ ................
+
+2009-07-20 16:40 +0000 [r207363] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 207361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009)
+ | 16 lines Merged revisions 207360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009)
+ | 9 lines Only do the chan->fdno check in ast_read() in a
+ developer build. I changed this check to only happen in a
+ dev-mode build. I also added a comment explaining what is going
+ on. I also made it so that detection of this situation does not
+ affect ast_read() operation. (closes issue #14723) Reported by:
+ seadweller ........ ................
+
+2009-07-18 04:17 +0000 [r207321] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Recorded merge of revisions 207317 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17
+ Jul 2009) | 3 lines Flag field in wrong position. Reported by
+ "Hoggins!" on asterisk-dev list. ........
+
+2009-07-18 02:09 +0000 [r207287] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn_config.c,
+ channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
+ doc/tex/misdn.tex, channels/chan_misdn.c, main/callerid.c,
+ configs/misdn.conf.sample: Merged revisions 145293,158010 from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 to make
+ merging easier. These changes are already on trunk.
+ ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500
+ (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c
+ channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk
+ to make merging easier later. ........ r145200 | rmudgett |
+ 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines *
+ Miscellaneous formatting changes to make v1.4 and trunk more
+ merge compatible in the mISDN area. channels/chan_misdn.c *
+ Eliminated redundant code in cb_events() EVENT_SETUP ........
+ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008)
+ | 9 lines improved helptext of misdn_set_opt. ........ r142181 |
+ rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
+ Cleaned up comment ........ r138738 | rmudgett | 2008-08-18
+ 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
+ channels/chan_misdn.c * Made bearer2str() use
+ allowed_bearers_array[] * Made use the causes.h defines instead
+ of hardcoded numbers. * Made use Asterisk presentation indicator
+ values if either of the mISDN presentation or screen options are
+ negative. * Updated the misdn_set_opt application option
+ descriptions. * Renamed the awkward Caller ID presentation
+ misdn_set_opt application option value not_screened to
+ restricted. Deprecated the not_screened option value.
+ channels/misdn/isdn_lib.c * Made use the causes.h defines instead
+ of hardcoded numbers. * Fixed some spelling errors and typos. *
+ Added all defined facility code strings to fac2str().
+ channels/misdn/isdn_lib.h * Added doxygen comments to struct
+ misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
+ comments to struct misdn_stack. channels/misdn_config.c
+ configs/misdn.conf.sample * Updated the mISDN presentation and
+ screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex)
+ * Updated the misdn_set_opt application option descriptions. *
+ Fixed some spelling errors and typos. ................ r158010 |
+ rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
+ Merged revision 157977 from
+ https://origsvn.digium.com/svn/asterisk/team/group/issue8824
+ ........ Fixes JIRA ABE-1726 The dial extension could be empty if
+ you are using MISDN_KEYPAD to control ISDN provider features.
+ ................
+
+2009-07-17 22:30 +0000 [r207227-207256] Tilghman Lesher <tlesher@digium.com>
+
+ * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 207255 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17
+ Jul 2009) | 2 lines Add flag here, too (as requested by jsmith)
+ ........
+
+ * /, doc/tex/odbcstorage.tex, UPGRADE.txt: Recorded merge of
+ revisions 207224 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r207224 |
+ tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17 Jul 2009) | 2 lines
+ Document the "flag" field in the voicemessages table. ........
+
+2009-07-17 19:39 +0000 [r207101-207158] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500
+ (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009)
+ | 7 lines Fix format specifier to print out an unsigned long
+ long. Yep, it's even ifdefed out code. But it made it to the RR
+ list... (closes issue #14726) Reported by: lmadsen ........
+ ................
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17
+ Jul 2009) | 2 lines Update some missing allowed options for
+ overlapdial ........
+
+2009-07-17 17:53 +0000 [r206870-207031] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 |
+ dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines
+ sip option flags handled incorrectly (closes issue #15376)
+ Reported by: Takehiko Ooshima Tested by: dvossel,
+ Takehiko_Ooshima ........
+
+ * /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009)
+ | 20 lines Merged revisions 206938 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009)
+ | 14 lines SIP incorrect From: header information when callpres
+ is prohib Some ITSP make use of the "Anonymous" display name to
+ detect a requirement to withhold caller id across the PSTN. This
+ does not work if the display name is "Unknown". (closes issue
+ #14465) Reported by: Nick_Lewis Patches:
+ chan_sip.c-callerpres.patch uploaded by Nick (license 657)
+ chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license
+ 671) Tested by: Nick_Lewis, dvossel ........ ................
+
+ * /, funcs/func_timeout.c: Merged revisions 206877 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009)
+ | 6 lines TIMEOUT(absolute) returned negative value. (closes
+ issue #15513) Reported by: ys ........
+
+ * configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500
+ (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009)
+ | 6 lines error in iax.conf related IP-based access control
+ (closes issue #15518) Reported by: pkempgen ........
+ ................
+
+ * /, main/callerid.c: Merged revisions 206868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009)
+ | 14 lines Merged revisions 206867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009)
+ | 8 lines avoid segfault caused by user error If the CALLERPRES()
+ dialplan function is set to nothing, a segfault occurs. This is
+ user error to begin with, but I'd rather see a cli warning
+ message than have Asterisk crash on me. ........ ................
+
+2009-07-16 16:53 +0000 [r206810] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500
+ (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009)
+ | 6 lines Fix a memory leak. (closes issue #15517) Reported by:
+ adomjan Patches: func_realtime.c-ast_variable_destroy.diff
+ uploaded by adomjan (license 487) ........ ................
+
+2009-07-15 22:06 +0000 [r206774] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 |
+ dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines
+ Session timer were not activated if Supported header field in
+ INVITE had both "timer" and other options. (closes issue #15403)
+ Reported by: makoto Patches: sip-session-timer.patch uploaded by
+ makoto (license ........
+
+2009-07-15 21:40 +0000 [r206764] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
+ Merged revisions 206707 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009)
+ | 33 lines Merged revisions 206706 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500
+ (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... Fixed chan_misdn crash because mISDNuser library is
+ not thread safe. With Asterisk the mISDNuser library is driven by
+ two threads concurrently: 1.
+ channels/misdn/isdn_lib.c::manager_event_handler() 2.
+ channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls
+ into the library are done concurrently and recursively from
+ isdn_lib.c. Both threads can fiddle with the master/child
+ layer3_proc_t lists. One thread may traverse the list when the
+ other interrupts it and then removes the list element which the
+ first thread was currently handling. This is exactly what caused
+ the crash. About 60 calls were needed to a Gigaset CX475 before
+ it occurred once. This patch adds locking when calling into the
+ mISDNuser library. This also fixes some cb_log calls with wrong
+ port parameter. JIRA ABE-1913 Patches: misdn-locking.patch
+ (Modified with mostly cosmetic changes) ..........
+ ................ ................
+
+2009-07-15 20:21 +0000 [r206704] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 |
+ dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines
+ callerid(num) is wrong when username is missing A domain only sip
+ uri <sip:123.123.123.123> would return 123.123.123.123 as callid
+ num. Now, if the username is missing from a uri, the callerid num
+ field is left empty. (closes issue #15476) Reported by: viraptor
+ ........
+
+2009-07-15 16:03 +0000 [r206638] Sean Bright <sean@malleable.com>
+
+ * /, codecs/codec_dahdi.c: Merged revisions 206636 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400
+ (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed,
+ 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we
+ are asking for it. ........ ................
+
+2009-07-14 20:25 +0000 [r206596] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/scripts/meetme.sql: Recorded merge of revisions 206567
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14
+ Jul 2009) | 6 lines Document all meetme realtime fields, and in
+ the process, make some field lengths more consistent. (closes
+ issue #15493) Reported by: lasko Patches: meetme.diff uploaded by
+ lasko (license 833) ........
+
+2009-07-14 18:32 +0000 [r206558] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500
+ (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009)
+ | 28 lines Fixes several call transfer issues with chan_misdn. *
+ issue #14355 - Crash if attempt to transfer a call to an
+ application. Masquerade the other pair of the four asterisk
+ channels involved in the two calls. The held call already must be
+ a bridged call (not an applicaton) or it would have been
+ rejected. * issue #14692 - Held calls are not automatically
+ cleared after transfer. Allow the core to initate disconnect of
+ held calls to the ISDN port. This also fixes a similar case where
+ the party on hold hangs up before being transferred or taken off
+ hold. * JIRA ABE-1903 - Orphaned held calls left in
+ music-on-hold. Do not simply block passing the hangup event on
+ held calls to asterisk core. * Fixed to allow held calls to be
+ transferred to ringing calls. Previously, held calls could only
+ be transferred to connected calls. * Eliminated unused call
+ states to simplify hangup code. * Eliminated most uses of
+ "holded" because it is not a word. (closes issue #14355) (closes
+ issue #14692) Reported by: sodom Patches:
+ misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
+ Tested by: rmudgett ........ ................
+
+2009-07-14 14:56 +0000 [r206388] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 206386 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r206386 | russell | 2009-07-14 09:51:44 -0500
+ (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r206385 | russell | 2009-07-14 09:48:00 -0500
+ (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009)
+ | 6 lines Ensure apathetic replies are sent out on the proper
+ socket. chan_iax2 supports multiple address bindings. The
+ send_apathetic_reply() function did not attempt to send its
+ response on the same socket that the incoming message came in on.
+ ........ ................ ................
+
+2009-07-14 01:35 +0000 [r206372] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+ revisions 206341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009)
+ | 11 lines Merged revisions 206284 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009)
+ | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911
+ ........ ................
+
+2009-07-13 23:33 +0000 [r206282] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 206280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 |
+ dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines
+ dns lookup of peername rather than peer's host in
+ transmit_register() (closes issue #15052) Reported by: fsantulli
+ Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by
+ fsantulli (license 818) Tested by: fsantulli ........
+
+2009-07-13 16:24 +0000 [r206186] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 206185 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009)
+ | 2 lines Remove reference to non-existent help file ........
+
+2009-07-10 21:52 +0000 [r205987] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 |
+ dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines
+ SIP register not using peer's outbound proxy If callbackextension
+ is defined for a peer it successfully causes a registration to
+ occur, but the registration ignores the outboundproxy settings
+ for the peer. This patch allows the peer to be passed to
+ obproxy_get() in transmit_register(). (closes issue #14344)
+ Reported by: Nick_Lewis Patches:
+ callbackextension_peer_trunk.diff uploaded by dvossel (license
+ 671) Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/294/ ........
+
+2009-07-10 18:45 +0000 [r205941] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /: Merged revisions 205939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 |
+ kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line
+ Update comments about the level of T.38 support in Asterisk.
+ ........
+
+2009-07-10 17:50 +0000 [r205881] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul
+ 2009) | 30 lines Merged revisions 205877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500
+ (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500
+ (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
+ 2009) | 10 lines Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc ........ ................ ................
+ ................
+
+2009-07-10 16:48 +0000 [r205842] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009)
+ | 37 lines Merged revisions 205804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009)
+ | 31 lines SIP registration auth loop caused by stale nonce If an
+ endpoint sends two registration requests in a very short period
+ of time with the same nonce, both receive 401 responses from
+ Asterisk, each with a different nonce (the second 401 containing
+ the current nonce and the first one being stale). If the endpoint
+ responds to the first 401, it does not match the current nonce so
+ Asterisk sends a third 401 with a newly generated nonce (which
+ updates the current nonce)... Now if the endpoint responds to the
+ second 401, it does not match the current nonce either and
+ Asterisk sends a fourth 401 with a newly generated nonce... This
+ loop goes on and on. There appears to be a simple fix for this.
+ If the nonce from the request does not match our nonce, but is a
+ good response to a previous nonce, instead of sending a 401 with
+ a newly generated nonce, use the current one instead. This breaks
+ the loop as the nonce is not updated until a response is
+ received. Additional logic has been added to make sure no nonce
+ can be responded to twice though. (closes issue #15102) Reported
+ by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license
+ 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by:
+ Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........
+ ................
+
+2009-07-10 15:57 +0000 [r205778] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul
+ 2009) | 16 lines Merged revisions 205775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
+ 2009) | 10 lines Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc ........ ................
+
+2009-07-10 15:36 +0000 [r205772] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 205770 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 |
+ kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12
+ lines Fix some remaining T.38 negotiation problems in app_fax.
+ Revision 205696 did not quite fix all the issues with the T.38
+ negotiation changes and app_fax; this patch corrects them, along
+ with a couple of other minor issues. (closes issue #15480)
+ Reported by: dimas Patches: test2-15480.patch uploaded by dimas
+ (license 88) ........
+
+2009-07-09 23:51 +0000 [r205730] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 205728 via svn merge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009)
+ | 21 lines No audio on calls from Asterisk to various ISDN
+ devices until DTMF sent by caller. Add missing clearing of the
+ dialing flag when the ISDN call is CONNECTED. (i.e. When libpri
+ generates the event PRI_EVENT_ANSWER.) (closes issue #15420)
+ Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt
+ uploaded by alecdavis (license 585) Tested by: scottbmilne,
+ alecdavis (closes issue #15416) Reported by: avinoash (closes
+ issue #15389) Reported by: alecdavis This patch should also fix
+ the following issue: (issue #15205) Reported by: vinsik ........
+
+2009-07-09 21:27 +0000 [r205698] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, /, channels/chan_sip.c, apps/app_fax.c:
+ Merged revisions 205696 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 |
+ kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16
+ lines Repair ability of SendFAX/ReceiveFAX to respond to T.38
+ switchover. Recent changes in T.38 negotiation in Asterisk caused
+ these applications to not respond when the other endpoint
+ initiated a switchover to T.38; this resulted in the T.38
+ switchover failing, and the FAX attempt to be made using an audio
+ connection, instead of T.38 (which would usually cause the FAX to
+ fail completely). This patch corrects this problem, and the
+ applications will now correctly respond to the T.38 switchover
+ request. In addition, the response will include the appopriate
+ T.38 session parameters based on what the other end offered and
+ what our end is capable of. (closes issue #14849) Reported by:
+ afosorio ........
+
+2009-07-09 16:20 +0000 [r205596-205605] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/time.h, /: Merged revisions 205600 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500
+ (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09
+ Jul 2009) | 2 lines Changing ast_samp2tv to not use floating
+ point. ........ ................
+
+ * channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c, /:
+ Merged revisions 205479 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009)
+ | 16 lines Merged revisions 205471 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009)
+ | 10 lines Fixes 8khz assumptions Many calculations assume 8khz
+ is the codec rate. This is not always the case. This patch only
+ addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there
+ are other areas that make this assumption as well. Review:
+ https://reviewboard.asterisk.org/r/306/ ........ ................
+
+2009-07-09 08:33 +0000 [r205534] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, main/ssl.c: Merged revisions 205532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 |
+ mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines
+ pthread_self returns a pthread_t which is not an unsigned int on
+ all pthread implementations. Casting it to an unsigned int fixes
+ compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit
+ ........
+
+2009-07-08 22:16 +0000 [r205414] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c, include/asterisk/pbx.h: Merged revisions
+ 205412 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009)
+ | 12 lines Merged revisions 205409 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009)
+ | 6 lines moving ast_devstate_to_extenstate to pbx.c from
+ devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This
+ change fixes a compile time error with chan_vpb as well. ........
+ ................
+
+2009-07-08 19:27 +0000 [r205352] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 205350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul
+ 2009) | 20 lines Merged revisions 205349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul
+ 2009) | 14 lines Prevent phantom calls to queue members. If a
+ caller were to hang up while a periodic announcement or position
+ were being said, the return value for those functions would
+ incorrectly indicate that the caller was still in the queue. With
+ these changes, the problem does not occur. (closes issue #14631)
+ Reported by: latinsud Patches: queue_announce_ghost_call2.diff
+ uploaded by latinsud (license 745) (with small modification from
+ me) ........ ................
+
+2009-07-08 18:21 +0000 [r205299] Jason Parker <jparker@digium.com>
+
+ * config.guess, config.sub, /: Merged revisions 205291 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205291 | qwell | 2009-07-08 13:19:46 -0500
+ (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul
+ 2009) | 1 line Update config.guess and config.sub from the
+ savannah.gnu.org git repo. ........ ................
+
+2009-07-08 18:07 +0000 [r205279] David Brooks <dbrooks@digium.com>
+
+ * /, main/features.c: Merged revisions 205254 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 |
+ dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines
+ Fixes Park() argument handling Park() was not respecting the
+ arguments passed to it. Any extension/context/priority given to
+ it was being ignored. This patch remedies this. (closes issue
+ #15380) Reported by: DLNoah ........
+
+2009-07-08 16:59 +0000 [r205222] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: oops, fixing build
+
+2009-07-08 16:56 +0000 [r205218] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/time.h, /: Merged revisions 205216 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500
+ (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009)
+ | 10 lines ast_samp2tv needs floating point for 16khz audio In
+ ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The
+ .5 is currently stripped off because we don't calculate using
+ floating points. This causes madness with 16khz audio. (issue
+ ABE-1899) Review: https://reviewboard.asterisk.org/r/305/
+ ........ ................
+
+2009-07-08 16:29 +0000 [r205203] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c: Merged revisions 205196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009)
+ | 9 lines Merged revisions 205188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009)
+ | 2 lines Add redirection warnings for the invalid language codes
+ previously removed. ........ ................
+
+2009-07-08 15:57 +0000 [r205147-205153] Russell Bryant <russell@digium.com>
+
+ * /, main/ssl.c: Merged revisions 205151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 |
+ russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines
+ Use tabs instead of spaces for indentation. ........
+
+ * res/res_jabber.c, main/asterisk.c, /, main/Makefile,
+ res/res_crypto.c, main/ssl.c (added),
+ include/asterisk/_private.h: Merged revisions 205120 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009)
+ | 16 lines Move OpenSSL initialization to a single place, make
+ library usage thread-safe. While doing some reading about
+ OpenSSL, I noticed a couple of things that needed to be improved
+ with our usage of OpenSSL. 1) We had initialization of the
+ library done in multiple modules. This has now been moved to a
+ core function that gets executed during Asterisk startup. We
+ already link OpenSSL into the core for TCP/TLS functionality, so
+ this was the most logical place to do it. 2) OpenSSL is not
+ thread-safe by default. However, making it thread safe is very
+ easy. We just have to provide a couple of callbacks. One callback
+ returns a thread ID. The other handles locking. For more
+ information, start with the "Is OpenSSL thread-safe?" question on
+ the FAQ page of openssl.org. ........
+
+2009-07-06 14:24 +0000 [r204976] Ryan Brindley <rbrindley@digium.com>
+
+ * main/config.c, /: Merged revisions 202753 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 |
+ rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9
+ lines If we delete the info, lets also delete the lines (closes
+ issue 0014509) Reported by: timeshell Patches:
+ 20090504__bug14509.diff.txt uploaded by tilghman (license 14)
+ Tested by: awk, timeshell ........
+
+2009-07-06 13:40 +0000 [r204950] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, /: Merged revisions 204948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 |
+ kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7
+ lines Improve handling of AST_CONTROL_T38 and
+ AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This
+ change allows applications that request T.38 negotiation on a
+ channel that does not support it to get the proper indication
+ that it is not supported, rather than thinking that negotiation
+ was started when it was not. ........
+
+2009-07-02 22:05 +0000 [r204837] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500
+ (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009)
+ | 10 lines Removed confusing warning message "Got Busy in
+ Connected State" If an incoming mISDN call is answered with the
+ Answer application and a subsequent Dial gets a busy endpoint
+ then it is valid for that already connected channel to get the
+ busy indication. Asterisk will play the busy tones until the
+ dialplan plays something else or hangs up the call. (closes issue
+ #11974) Reported by: fvdb ........ ................
+
+2009-07-02 16:28 +0000 [r204736] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c: Merged revisions 204710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009)
+ | 21 lines Merged revisions 204681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009)
+ | 14 lines Improved mapping of extension states from combined
+ device states. This fixes a few issues with incorrect extension
+ states and adds a cli command, core show device2extenstate, to
+ display all possible state mappings. (closes issue #15413)
+ Reported by: legart Patches: exten_helper.diff uploaded by
+ dvossel (license 671) Tested by: dvossel, legart, amilcar Review:
+ https://reviewboard.asterisk.org/r/301/ ........ ................
+
+2009-06-30 21:30 +0000 [r204612] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500
+ (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009)
+ | 6 lines More incorrect language codes, plus ensuring that
+ regionalizations use the specified language, and not English for
+ grammar. (closes issue #15022) Reported by: greenfieldtech
+ Patches: 20090519__issue15022.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+2009-06-30 18:52 +0000 [r204477] Jason Parker <jparker@digium.com>
+
+ * /, main/say.c: Merged revisions 204475 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) |
+ 9 lines Merged revisions 204474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) |
+ 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a
+ comment typo in passing. ........ ................
+
+2009-06-30 18:44 +0000 [r204472] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge
+ of revisions 204470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009)
+ | 18 lines Recorded merge of revisions 204469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009)
+ | 11 lines "tw" is the language specification for Twi (from
+ Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier
+ Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__trunk.diff.txt uploaded by
+ tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt
+ uploaded by tilghman (license 14)
+ 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by
+ tilghman (license 14) Tested by: volivier ........
+ ................
+
+2009-06-29 22:53 +0000 [r204249-204303] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun
+ 2009) | 15 lines Merged revisions 204300 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun
+ 2009) | 9 lines Add error message so that it is clear why a SIP
+ peer was not processed when a DNS lookup fails on a host or
+ outboundproxy. (closes issue #13432) Reported by: p_lindheimer
+ Patches: outboundproxy.patch uploaded by p (license 558) ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun
+ 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun
+ 2009) | 22 lines Fix a problem where chan_sip would ignore "old"
+ but valid responses. chan_sip has had a problem for quite a long
+ time that would manifest when Asterisk would send multiple SIP
+ responses on the same dialog before receiving a response. The
+ problem occurred because chan_sip only kept track of the highest
+ outgoing sequence number used on the dialog. If Asterisk sent two
+ requests out, and a response arrived for the first request sent,
+ then Asterisk would ignore the response. The result was that
+ Asterisk would continue retransmitting the requests and ignoring
+ the responses until the maximum number of retransmissions had
+ been reached. The fix here is to rearrange the code a bit so that
+ instead of simply comparing the sequence number of the response
+ to our latest outgoing sequence number, we walk our list of
+ outstanding packets and determine if there is a match. If there
+ is, we continue. If not, then we ignore the response. In doing
+ this, I found a few completely useless variables that I have now
+ removed. (closes issue #11231) Reported by: flefoll Review:
+ https://reviewboard.asterisk.org/r/298 ........ r204246 |
+ mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3
+ lines Fix build oops. ........ ................
+
+2009-06-27 01:18 +0000 [r203918] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500
+ (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009)
+ | 16 lines The ISDN CPE side should not exclusively pick B
+ channels normally. Before this patch, Asterisk unconditionally
+ picked B channels exclusively on the CPE side and normally
+ allowed alternative B channels on the network side. Now Asterisk
+ does the opposite. Reasons for the CPE side to normally not pick
+ B channels exclusively: * For CPE point-to-multipoint mode (i.e.
+ phone side), the CPE side does not have enough information to
+ exclusively pick B channels. (There may be other devices on the
+ line.) * Q.931 gives preference to the network side picking B
+ channels. * Some telcos require the CPE side to not pick B
+ channels exclusively. (closes issue #14383) Reported by:
+ mbrancaleoni ........ ................
+
+2009-06-26 22:13 +0000 [r203856] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500
+ (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009)
+ | 5 lines Make sure to recreate the dahdi pseudo channel after
+ dahdi restart (closes issue #14477) Reported by: timking ........
+ ................
+
+2009-06-26 21:26 +0000 [r203781-203823] Russell Bryant <russell@digium.com>
+
+ * /, main/file.c: Merged revisions 203802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009)
+ | 22 lines Merged revisions 203785 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009)
+ | 15 lines Don't fast forward past the end of a message. This is
+ nice change for users of the voicemail application. If someone
+ gets a little carried away with fast forwarding through a
+ message, they can easily get to the end and accidentally exit the
+ voicemail application by hitting the fast forward key during the
+ following prompt. This adds some safety by not allowing a fast
+ forward past the end of a message. (closes issue #14554) Reported
+ by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
+ 707) Tested by: lacoursj ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 |
+ russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines
+ Ensure the TCP read buffer is fully initialized before handling
+ each packet. (closes issue #14452) Reported by: umberto71
+ ........
+
+2009-06-26 20:18 +0000 [r203727] David Brooks <dbrooks@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009)
+ | 16 lines Fixing voicemail's error in checking max silence vs
+ min message length Max silence was represented in milliseconds,
+ yet vmminsecs (minmessage) was represented as seconds. Also, the
+ inequality was reversed. The warning, if triggered, was "Max
+ silence should be less than minmessage or you may get empty
+ messages", which should have been logged if max silence was
+ greater than minmessage, but the check was for less than. Also,
+ conforming if statement to coding guidelines. closes issue
+ #15331) Reported by: markd Review:
+ https://reviewboard.asterisk.org/r/293/ ........
+
+2009-06-26 19:56 +0000 [r203718] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: reverse whitespace change 203713 that was
+ based on looking at sig_analog (which has about a 1000 line
+ indentation change that is not worth doing here)
+
+2009-06-26 19:48 +0000 [r203714] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 203710 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009)
+ | 7 lines moving debug message from level 0 to 1. (closes issue
+ #15404) Reported by: leobrown Patches: iax_codec_debug.patch
+ uploaded by leobrown (license 541) ........
+
+2009-06-26 19:48 +0000 [r203713] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: whitespace fix
+
+2009-06-26 19:37 +0000 [r203704] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, /,
+ main/devicestate.c: Merged revisions 203702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 |
+ russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines
+ Make invalid hints report Unavailable instead of Idle. (closes
+ issue #14413) Reported by: pj ........
+
+2009-06-26 19:31 +0000 [r203703] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/frame.h, main/rtp.c, main/channel.c,
+ main/frame.c, /, channels/chan_sip.c, apps/app_fax.c,
+ configs/sip.conf.sample: Merged revisions 203699 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2
+ lines Improve T.38 negotiation by exchanging session parameters
+ between application and channel. ........
+
+2009-06-26 19:28 +0000 [r203700] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009)
+ | 16 lines Check if polarityonanswerdelay has elapsed before
+ setting a channel as answered after a polarity reversal.
+ Previously on a polarity switch event chan_dahdi would set the
+ channel immediately as answered. This would cause problems if a
+ polarity reversal occurred when the line was picked up as the
+ dial would not have yet occurred. Now if the polarity reversal
+ occurs before delay has elapsed after coming off hook or an
+ answer, it is ignored. Also, some refactoring was done in
+ _handle_event. (closes issue #13917) Reported by: alecdavis
+ Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by
+ alecdavis (license 585) Tested by: alecdavis ........
+
+2009-06-25 21:46 +0000 [r203446] David Vossel <dvossel@digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25
+ Jun 2009) | 4 lines fixes a few redundant conditions (issue
+ #15269) ........
+
+2009-06-25 21:19 +0000 [r203393] Terry Wilson <twilson@digium.com>
+
+ * main/cli.c, /: Merged revisions 203381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009)
+ | 11 lines Merged revisions 203380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009)
+ | 4 lines I didn't see that Mark already fixed the underlying
+ issue! Yay for removing useless code. ........ ................
+
+2009-06-25 21:07 +0000 [r203378] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 203376 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009)
+ | 16 lines Merged revisions 203375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009)
+ | 9 lines Fix a case where CDR answer time could be before the
+ start time involving parking. (closes issue #13794) Reported by:
+ davidw Patches: 13794.patch uploaded by murf (license 17)
+ 13794.patch.160 uploaded by murf (license 17) Tested by: murf,
+ dbrooks ........ ................
+
+2009-06-25 19:27 +0000 [r203274] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203258 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) |
+ 10 lines Unmute when we get a dtmfup (we muted on dtmfdown)
+ event. This would occasionally cause one-way audio when using
+ hardware DTMF detection. (closes issue #14761) Reported by:
+ tzafrir Patches: v1-14761.patch uploaded by dimas (license 88)
+ Tested by: tzafrir, dimas ........
+
+2009-06-25 16:07 +0000 [r203118] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009)
+ | 18 lines Merged revisions 203115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009)
+ | 11 lines Resolve a crash related to a T.38 reinvite race
+ condition. This change resolves a crash observed locally during
+ some T.38 testing. A call was set up using a call file, and when
+ the T.38 reinvite came in, the channel state was still
+ AST_STATE_DOWN. The reason is explained by a comment in the code
+ that previously lived in the handling of AST_STATE_RINGING. This
+ change modifies the logic to handle the same race condition for
+ any channel state that is not UP. (closes ABE-1895) ........
+ ................
+
+2009-06-24 21:22 +0000 [r203057] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500
+ (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009)
+ | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid
+ format is: pritimer=timer_name,timer_value * Fixed segfault if
+ the ',' is missing. * Completely check the range returned by
+ pri_timer2idx() to prevent possible access outside array bounds.
+ ........ ................
+
+2009-06-24 18:30 +0000 [r202969] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202967 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun
+ 2009) | 9 lines Merged revisions 202966 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun
+ 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding
+ the same thing in-line. ........ ................
+
+2009-06-24 18:10 +0000 [r202927] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 |
+ file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines
+ Ensure the default settings are applied for T.38 when we set it
+ up for a peer. ........
+
+2009-06-23 22:11 +0000 [r202764] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) |
+ 1 line I could have sworn I committed this patch ages ago, but...
+ bug fix with setting NAI properly on linksets in certain
+ situations. ........
+
+2009-06-23 16:34 +0000 [r202674] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009)
+ | 18 lines Merged revisions 202671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009)
+ | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to
+ non-standard port and transport (closes issue #14659) Reported
+ by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded
+ by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded
+ by dvossel (license 671) Tested by: dvossel, klaus3000 Review:
+ https://reviewboard.asterisk.org/r/288/ ........ ................
+
+2009-06-22 20:18 +0000 [r202503] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 202497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009)
+ | 11 lines Merged revisions 202496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009)
+ | 4 lines Report CallerID change during a masquerade. Reported
+ by: markster ........ ................
+
+2009-06-22 16:31 +0000 [r202472] Sean Bright <sean@malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun
+ 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid
+ potential crashes during reload. Pointed out by Russell while
+ working on the CEL branch. ........
+
+2009-06-22 16:14 +0000 [r202418] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009)
+ | 9 lines Merged revisions 202414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009)
+ | 2 lines Make Polycom subscription type override check more
+ explicit. ........ ................
+
+2009-06-22 15:41 +0000 [r202412] David Vossel <dvossel@digium.com>
+
+ * main/loader.c, /, include/asterisk/module.h: Merged revisions
+ 202410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 |
+ dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines
+ attempting to load running modules Modules placed in the priority
+ heap for loading were not properly removed from the linked list.
+ This resulted in some modules attempting to load twice. ........
+
+2009-06-22 15:10 +0000 [r202339-202345] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun
+ 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun
+ 2009) | 26 lines Fix a situation in which Asterisk would not stop
+ retransmitting 487s. If a CANCEL were received by Asterisk, we
+ would send a 487 in response to the original INVITE and a 200 OK
+ for the CANCEL. If there were a network hiccup which caused the
+ 200 OK and the 487 to be lost, then the UA communicating with
+ Asterisk may try to retransmit its CANCEL. Asterisk's response to
+ this used to be to try sending another 487 to the canceled INVITE
+ and another 200 OK to the CANCEL. The problem here is that the
+ originally-sent 487 was sent "reliably" meaning that it will be
+ retransmitted until it is received properly. So when we receive
+ the second CANCEL it is likely that the first batch of 487s we
+ sent is still going strong and reaches the UA. The result was
+ that the second set of 487s would be retransmitted constantly
+ until the maximum number of retries had been reached. The fix for
+ this is that if we receive a second CANCEL for an INVITE, then we
+ cancel the retransmission of the first set of 487s and start a
+ second set. This causes the dialog to be terminated reasonably.
+ (closes issue #14584) Reported by: klaus3000 Patches:
+ 14584_v2.patch uploaded by mmichelson (license 60) Tested by:
+ klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58
+ -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line
+ left from previous commit. ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun
+ 2009) | 31 lines Merged revisions 202336 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun
+ 2009) | 25 lines Fix a possible infinite loop in SDP parsing
+ during glare situation. There was a while loop in
+ get_ip_and_port_from_sdp which was controlled by a call to
+ get_sdp_iterate. The loop would exit either if what we were
+ searching for was found or if the return was NULL. The problem is
+ that get_sdp_iterate never returns NULL. This means that if what
+ we were searching for was not present, the loop would run
+ infinitely. This modification of the loop fixes the problem.
+ (closes issue #15213) Reported by: schmidts (closes issue #15349)
+ Reported by: samy (closes issue #14464) Reported by: pj (closes
+ issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
+ uploaded by mmichelson (license 60) Tested by: aragon ........
+ ................
+
+2009-06-21 16:15 +0000 [r202260-202264] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 |
+ russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines
+ Fix possibility of crashiness during reload in custom fields
+ handling. ........
+
+ * cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 |
+ russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines
+ Standardize return values of load_config() so reload() doesn't
+ report an error on success. ........
+
+2009-06-20 19:14 +0000 [r202185] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_fax.c: Merged revisions 202183 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 |
+ seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5
+ lines Fix version detection for API changes in spandsp. (closes
+ issue #15355) Reported by: deuffy ........
+
+2009-06-19 21:08 +0000 [r202008] Matthew Nicholson <mnicholson@digium.com>
+
+ * channels/chan_sip.c: Added deadlock protection to
+ try_suggested_sip_codec in chan_sip.c. Review:
+ https://reviewboard.asterisk.org/r/287/
+
+2009-06-19 20:26 +0000 [r201996] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 201994 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500
+ (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009)
+ | 8 lines timestamp was being converted to host order as a short
+ rather than a long (closes issue #15361) Reported by: ffloimair
+ Patches: ts_issue.diff uploaded by dvossel (license 671) ........
+ ................
+
+2009-06-19 15:48 +0000 [r201784-201905] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c, /: Merged revisions 201904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r201904 | tilghman | 2009-06-19 10:47:55 -0500 (Fri, 19 Jun 2009)
+ | 4 lines Fix 2 typos and add support for wide character types.
+ Reported by Benny Amorsen via the asterisk-users mailing list.
+ http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html
+ ........
+
+ * main/features.c: If the "h" extension fails, give it another
+ chance in main/pbx.c. If the "h" extension fails, give it another
+ chance in main/pbx.c, when it returns from the bridge code. Fixes
+ an issue where the "h" extension may occasionally not fire, when
+ a Dial is executed from a Macro. Debugged in #asterisk with user
+ tompaw.
+
+ * /, apps/Makefile: Merged revisions 201783 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 |
+ tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines
+ One of the changes in 1.6.1 was to allow app_directory to use
+ functionality within app_voicemail for directory functions. It is
+ therefore no longer necessary for app_directory to be linked
+ against the ODBC libraries (and it never was necessary for
+ app_directory to be linked against IMAP, though it was). ........
+
+2009-06-18 16:51 +0000 [r201680] David Vossel <dvossel@digium.com>
+
+ * channels/misdn/isdn_lib.c, utils/conf2ael.c, main/ast_expr2.c,
+ utils/stereorize.c, main/ast_expr2f.c, res/ael/ael_lex.c,
+ utils/ael_main.c, utils/extconf.c, channels/xpmr/xpmr.c,
+ pbx/pbx_config.c, res/res_config_ldap.c, apps/app_rpt.c,
+ main/asterisk.c, codecs/gsm/src/gsm_destroy.c, /,
+ channels/h323/ast_h323.cxx: Merged revisions 201678 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009)
+ | 11 lines fixes some memory leaks and redundant conditions
+ (closes issue #15269) Reported by: contactmayankjain Patches:
+ patch.txt uploaded by contactmayankjain (license 740)
+ memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
+ Tested by: contactmayankjain, dvossel ........
+
+2009-06-18 15:36 +0000 [r201613] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 201610 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201610 | russell | 2009-06-18 10:27:10 -0500
+ (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009)
+ | 29 lines Fix memory corruption and leakage related reloads of
+ non files mode MoH classes. For Music on Hold classes that are
+ not files mode, meaning that we are executing an application that
+ will feed us audio data, we use a thread to monitor the external
+ application and read audio from it. This thread also makes use of
+ the MoH class object. In the MoH class destructor, we used
+ pthread_cancel() to ask the thread to exit. Unfortunately, the
+ code did not wait to ensure that the thread actually went away.
+ What needed to be done is a pthread_join() to ensure that the
+ thread fully cleans up before we proceed. By adding this one
+ line, we resolve two significant problems: 1) Since the thread
+ was never joined, it never fully goes away. So, on every reload
+ of non-files mode MoH, an unused thread was sticking around. 2)
+ There was a race condition here where the application monitoring
+ thread could still try to access the MoH class, even though the
+ thread executing the MoH reload has already destroyed it. (issue
+ #15109) Reported by: jvandal (issue #15123) Reported by:
+ axisinternet (issue #15195) Reported by: amorsen (issue AST-208)
+ ........ ................
+
+2009-06-18 15:24 +0000 [r201601] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201570 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 |
+ dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines
+ parsing extension correctly from sip register lines If a
+ transport type was specified, but no extension, parsing of the
+ extension would return whatever was after the transport rather
+ than defaulting to 's'. (closes issue #15111) Reported by: ffs
+ Patches: chan_sip.c_register-parser.patch uploaded by ffs
+ (license 730) Tested by: ffs, dvossel ........
+
+2009-06-17 21:32 +0000 [r201532] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 201531 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009)
+ | 7 lines Initialize additional variables, to prevent a possible
+ crash. (closes issue #15186) Reported by: ajohnson Patches:
+ 20090528__issue15186.diff.txt uploaded by tilghman (license 14)
+ Tested by: ajohnson ........
+
+2009-06-17 20:11 +0000 [r201460-201464] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 |
+ mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12
+ lines Fix problem with no audio due to ignoring the SDP. A recent
+ change to our SDP version comparison made audio not function on
+ some calls. This was because of a test wherein we were trying to
+ see if an unsigned value was less than 0. This is a dumb
+ comparison and arguably the compiler should have warned about it.
+ Alas, though, it slipped past. Now it's fixed by changing the
+ variable to be a signed type. Found by several developers. Tested
+ by mnicholson and dbrooks. ........
+
+ * main/channel.c, /: Merged revisions 201458 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun
+ 2009) | 15 lines Merged revisions 201450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun
+ 2009) | 9 lines Change the datastore traversal in
+ ast_do_masquerade to use a safe list traversal. It is possible
+ for datastore fixup functions to remove the datastore from the
+ list and free it. In particular, the queue_transfer_fixup in
+ app_queue does this. While I don't yet know of this causing any
+ crashes, it certainly could. Found while discussing a separate
+ issue with Brian Degenhardt. ........ ................
+
+2009-06-17 20:01 +0000 [r201448-201456] David Vossel <dvossel@digium.com>
+
+ * doc/datastores.txt, /: Merged revisions 201453 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 |
+ dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines
+ ast_channel_datastore_alloc is no longer used. updating
+ datastores.txt to reflect that. ........
+
+ * apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500
+ (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009)
+ | 19 lines StopMixMonitor race condition (not giving up file
+ immediately) StopMixMonitor only indicates to the MixMonitor
+ thread to stop writing to the file. It does not guarantee that
+ the recording's file handle is available to the dialplan
+ immediately after execution. This results in a race condition. To
+ resolve this, the filestream pointer is placed in a datastore on
+ the channel. When StopMixMonitor is called, the datastore is
+ retrieved from the channel and the filestream is closed
+ immediately before returning to the dialplan. Documentation
+ indicating the use of StopMixMonitor to free files has been
+ updated as well. (closes issue #15259) Reported by: travisghansen
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/283/ ........ ................
+
+2009-06-17 19:39 +0000 [r201444] David Brooks <dbrooks@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009)
+ | 16 lines Merged revisions 201380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009)
+ | 9 lines Checks for NULL sip_pvt pointer in
+ chan_sip.c->acf_channel_read() Zombie channels could be passed,
+ and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
+ checking for NULL pointer. (closes issue #15330) Reported by:
+ okrief Tested by: dbrooks ........ ................
+
+2009-06-17 15:32 +0000 [r201365] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 |
+ dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines
+ SIP registry ref count error During a sip reload, the list of
+ sip_registry objects are supposed to be traversed, unlinked, and
+ destroyed, but destruction never takes place due to a ref
+ counting error. This causes a memory leak when registry items are
+ removed from sip.conf and reloaded. While the registries are
+ removed from the global list, they are not removed from the
+ scheduler. Because of this, SIP register attempts continue to be
+ sent out for the item even though it may no longer be in the
+ .conf. (closes issue #15295) Reported by: amorsen Review:
+ https://reviewboard.asterisk.org/r/282/ ........
+
+2009-06-17 12:05 +0000 [r201264] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 201262 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500
+ (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun
+ 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list
+ to be appended is empty. When the list to be appended is empty,
+ and the list to be appended to is *not*, AST_LIST_APPEND_LIST
+ would actually cause the target list to become broken, and no
+ longer have a pointer to its last entry. This patch fixes the
+ problem. (reported by Stanislaw Pitucha on the asterisk-dev
+ mailing list) ........ ................
+
+2009-06-16 22:31 +0000 [r201225] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 |
+ dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines
+ fix issue with build_contact introduced by the "SIP trasnport
+ type issues" commit ........
+
+2009-06-16 19:42 +0000 [r200989-201096] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, apps/app_chanspy.c,
+ apps/app_mixmonitor.c, main/channel.c, main/autoservice.c,
+ main/frame.c, /, apps/app_meetme.c, main/slinfactory.c,
+ include/asterisk/linkedlists.h, main/file.c,
+ include/asterisk/channel.h: Merged revisions 201056 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500
+ (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun
+ 2009) | 11 lines Improve support for media paths that can
+ generate multiple frames at once. There are various media paths
+ in Asterisk (codec translators and UDPTL, primarily) that can
+ generate more than one frame to be generated when the application
+ calling them expects only a single frame. This patch addresses a
+ number of those cases, at least the primary ones to solve the
+ known problems. In addition it removes the broken TRACE_FRAMES
+ support, fixes a number of bugs in various frame-related API
+ functions, and cleans up various code paths affected by these
+ changes. https://reviewboard.asterisk.org/r/175/ ........
+ ................
+
+ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
+ revisions 201090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201090 |
+ kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5
+ lines Another minor fix to compiler attribute checking.
+ Defaulting to 'static' for the function scope was bad... so
+ remove it. ........
+
+ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
+ revisions 200985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 |
+ kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7
+ lines Fix problems with new compiler attribute checking in
+ configure script. The last changes to ast_gcc_attribute.m4 caused
+ some problems checking for various attributes, because the scope
+ of the symbol the attribute is applied to can be important; this
+ patch allows the scope to be specified for the check. ........
+
+2009-06-16 16:34 +0000 [r200987] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 |
+ dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines
+ SIP transport type issues What this patch addresses: 1.
+ ast_sip_ouraddrfor() by default binds to the UDP address/port
+ reguardless if the sip->pvt is of type UDP or not. Now when no
+ remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
+ transport type, attempting to set the address and port to the
+ correct TCP/TLS bindings if necessary. 2. It is not necessary to
+ send the port number in the Contact header unless the port is
+ non-standard for the transport type. This patch fixes this and
+ removes the todo note. 3. In sip_alloc(), the default dialog
+ built always uses transport type UDP. Now sip_alloc() looks at
+ the sip_request (if present) and determines what transport type
+ to use by default. 4. When changing the transport type of a
+ sip_socket, the file descriptor must be set to -1 and in some
+ cases the tcptls_session's ref count must be decremented and set
+ to NULL. I've encountered several issues associated with this
+ process and have created a function, set_socket_transport(), to
+ handle the setting of the socket type. (closes issue #13865)
+ Reported by: st Patches: dont_add_port_if_tls.patch uploaded by
+ Kristijan (license 753) 13865.patch uploaded by mmichelson
+ (license 60) tls_port_v5.patch uploaded by vrban (license 756)
+ transport_issues.diff uploaded by dvossel (license 671) Tested
+ by: mmichelson, Kristijan, vrban, jmacz, dvossel Review:
+ https://reviewboard.asterisk.org/r/278/ ........
+
+2009-06-16 16:04 +0000 [r200947] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009)
+ | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail
+ can only use one storage module at the moment. Because it's
+ unclear that selecting one of the storage modules in menuselect
+ will disable filesystem storage we now have a FILE_STORAGE option
+ that conflicts with the other modules. (closes issue #15333)
+ ........
+
+2009-06-16 01:32 +0000 [r200707-200766] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15
+ Jun 2009) | 11 lines Ensure that configure-script testing for
+ compiler attributes actually works. The configure script tests
+ for compiler attributes didn't actually enable enough warnings or
+ provide a proper test harness to determine whether the compiler
+ supports the attribute in question or not; this caused gcc 4.1 to
+ report that it supports 'weakref', but it doesn't actually
+ support it in the way that is needed for our optional API
+ mechanism. The new configure script test will properly
+ distinguish between full support and partial support for this
+ attribute, among others. ........
+
+ * CHANGES, /: Merged revisions 200726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 |
+ kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6
+ lines Document the new automatic 'ignoresdpversion' behavior.
+ Asterisk will now automatically ignore incorrect incoming SDP
+ version numbers when necessary to complete a T.38 re-INVITE
+ operation. ........
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 165180,200689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 |
+ mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14
+ lines This patch adds a new 'ignoresdpversion' option to
+ sip.conf. When this is enabled (either globally or for a specific
+ peer), chan_sip will treat any SDP data it receives as new data
+ and update the media stream accordingly. By default, Asterisk
+ will only modify the media stream if the SDP session version
+ received is different from the current SDP session version. This
+ option is required to interoperate with devices that have
+ non-standard SDP session version implementations (observed by toc
+ on the bug tracker with Microsoft OCS which always uses 0 as the
+ session version). http://reviewboard.digium.com/r/94/ (closes
+ issue #13958) Reported by: toc Tested by: toc ........ r200689 |
+ kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12
+ lines Accept T.38 re-INVITE responses with invalid SDP versions.
+ This commit changes the 'incoming SDP version' check logic a bit
+ more; when 'ignoresdpversion' is *not* set for a peer, if we
+ initiate a re-INVITE to switch to T.38, we'll always accept the
+ peer's SDP response, even if they don't properly increment the
+ SDP version number as they should. If this situation occurs, a
+ warning message will be generated suggesting that the peer's
+ configuration be changed to include the 'ignoresdpversion'
+ configuration option (although ideally they'd fix their SIP
+ implementation to be RFC compliant). AST-221 ........
+
+2009-06-15 15:23 +0000 [r200516] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun
+ 2009) | 11 lines Merged revisions 200513 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun
+ 2009) | 5 lines Add INFO to our allowed methods so that endpoints
+ know they may send it to us. AST-223 ........ ................
+
+2009-06-12 19:08 +0000 [r200363] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 200361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun
+ 2009) | 16 lines Merged revisions 200360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun
+ 2009) | 10 lines Suppress a warning message and give a better
+ return code when generating inband ringing after a call is
+ answered. (closes issue #15158) Reported by: madkins Patches:
+ 15158.patch uploaded by mmichelson (license 60) Tested by:
+ madkins ........ ................
+
+2009-06-11 22:44 +0000 [r200229] Sean Bright <sean@malleable.com>
+
+ * Makefile, /: Merged revisions 199781 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 |
+ seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2
+ lines Fix all of the parallel build warnings issued when running
+ make -j#. ........
+
+2009-06-11 21:25 +0000 [r200171] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null
+
+2009-06-11 21:18 +0000 [r200152] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 |
+ mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5
+ lines Fix a crash due to a potentially NULL p->options. Thanks to
+ mnicholson for pointing it out. ........
+
+2009-06-11 12:16 +0000 [r200041] Leif Madsen <lmadsen@digium.com>
+
+ * build_tools/make_version_h, /, build_tools/make_version_c: Merged
+ revisions 200039 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 |
+ lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
+ Fix path for .flavor and .version (issue #14737) Reported by:
+ davidw Patches: flavor.patch uploaded by davidw (license 780)
+ Tested by: davidw ........
+
+2009-06-10 20:35 +0000 [r199996] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Fixes the argument order in definition of
+ new_find_extension(). In the definition of new_find_extension(),
+ the arguments 'callerid' and 'label' were swapped. The prototype
+ declaration and all calls to the function are ordered 'callerid'
+ then 'label', but the function itself was ordered 'label' then
+ 'callerid'. (closes issue #15303) Reported by: JimDickenson
+
+2009-06-10 20:18 +0000 [r199963] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 |
+ mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6
+ lines Only try to use the invite_branch on outgoing INVITEs with
+ auth credentials. I have added a comment to the code to help ease
+ understanding of the logic here as well. ........
+
+2009-06-10 16:13 +0000 [r199859] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400
+ (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
+ 10 Jun 2009) | 2 lines __WORDSIZE is not available on all
+ platforms, so use sizeof(void *) instead. ........
+ ................
+
+2009-06-09 20:50 +0000 [r199745-199820] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 |
+ dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
+ CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command
+ only used UDP rather than copying the transport type from the
+ peer. (closes issue #15283) Reported by: jthurman Patches:
+ sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
+ Tested by: jthurman, dvossel ........
+
+ * main/loader.c, /, res/res_timing_pthread.c,
+ include/asterisk/module.h, res/res_timing_dahdi.c: Merged
+ revisions 199743 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199743 |
+ dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines
+ module load priority This patch adds the option to give a module
+ a load priority. The value represents the order in which a
+ module's load() function is initialized. The lower the value, the
+ higher the priority. The value is only checked if the
+ AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER
+ flag is not set, the value will never be read and the module will
+ be given the lowest possible priority on load. Since some modules
+ are reliant on a timing interface, the timing modules have been
+ given a high load priorty. (closes issue #15191) Reported by:
+ alecdavis Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/262/ ........
+
+2009-06-08 19:39 +0000 [r199633] Sean Bright <sean@malleable.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400
+ (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
+ 2009) | 21 lines Increase the size of our thread stack on 64 bit
+ processors. We were setting the stack size for each thread to
+ 240KB regardless of architecture, which meant that in some
+ scenarios we actually had less available stack space on 64 bit
+ processors (pointers use 8 bytes instead of 4). So now we
+ calculate the stack size we reserve based on the platform's
+ __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
+ bit -> 1008KB (that's right, we're ready for 128 bit processors)
+ Patch typed by me but written by several members of
+ #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
+ issue #14932) Reported by: jpiszcz Patches:
+ 06052009_issue14932.patch uploaded by seanbright (license 71)
+ Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
+ 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
+ stack size calculation just introduced. ........ ................
+
+2009-06-08 17:35 +0000 [r199590] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Recorded merge of revisions 199588 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon,
+ 08 Jun 2009) | 9 lines Fix a deadlock that could occur when
+ setting rtp stats on SIP calls. (closes issue #15143) Reported
+ by: cristiandimache Patches: 15143.patch uploaded by mmichelson
+ (license 60) Tested by: cristiandimache ........
+
+2009-06-05 21:32 +0000 [r199300] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/devicestate.h, /, main/devicestate.c: Merged
+ revisions 199298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009)
+ | 21 lines Merged revisions 199297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
+ | 14 lines Fixes issue with hints giving unexpected results.
+ Hints with two or more devices that include ONHOLD gave
+ unexpected results. (closes issue #15057) Reported by:
+ p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
+ (license 671) pbx.c.1.4.patch uploaded by p (license 558)
+ devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
+ p_lindheimer, dvossel Review:
+ https://reviewboard.asterisk.org/r/254/ ........ ................
+
+2009-06-05 13:51 +0000 [r199229] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun
+ 2009) | 14 lines Correct "dahdi show channels" output when
+ specifying a group. Since a DAHDI channel may belong to multiple
+ groups, we need to use a bitwise and instead of equivalence to
+ determine whether to display the channel information. (closes
+ issue #15248) Reported by: gentian Patches: 15248.patch uploaded
+ by mmichelson (license 60) Tested by: gentian ........
+
+2009-06-04 19:16 +0000 [r199141] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 199139 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500
+ (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
+ Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
+ ................
+
+2009-06-05 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.1 released
+
+2009-06-04 David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Additional updates for AST-2009-001
+
+2009-06-04 David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001
+
+2009-04-27 Leif Madsen <lmadsen@digium.com>
+
+ * Create Asterisk 1.6.1.0
+
+2009-04-20 Leif Madsen <lmadsen@digium.com>
+
+ * Create Asterisk 1.6.1.0-rc5
+
+2009-04-20 17:08 +0000 [r189352] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 |
+ file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
+ Fix a bug with non-UDP connections that caused dialogs to not get
+ freed. This issue crept up because of a reference count issue on
+ non-UDP based dialogs. The dialog reference count was increased
+ when transmitting a packet reliably but never decreased. This
+ caused the dialog structure to hang around despite being unlinked
+ from the dialogs container. (closes issue #14919) Reported by:
+ vrban ........
+
+2009-04-20 14:06 +0000 [r189280] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 189278 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr
+ 2009) | 18 lines Merged revisions 189277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
+ 2009) | 12 lines Move the check for chan->fdno == -1 to after the
+ zombie/hangup check. Many users were finding that their hung up
+ channels were staying up and causing 100% CPU usage. (issue
+ #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
+ uploaded by mmichelson (license 60) Tested by: falves11, bamby
+ ........ ................
+
+2009-04-18 01:38 +0000 [r189206] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 189204 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500
+ (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009)
+ | 12 lines Fixed autologoff in agents.conf not working when agent
+ logs in via AgentLogin app An agent logs in by calling an
+ extension that calls the AgentLogin app. In agents.conf
+ ackcall=always is set, so when they get a call they have the
+ choice to either acknowledge it or ignore it. autologoff=10 is
+ set as well, so if the agent ignores the call over 10sec one may
+ assume that the agent should be logged out (and in this case
+ hungup on as well), but this was not happening. (closes issue
+ #14091) Reported by: evandro Patches: autologoff.diff uploaded by
+ dvossel (license 671) Review:
+ http://reviewboard.digium.com/r/225/ ........ ................
+
+2009-04-17 21:55 +0000 [r189139] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+ revisions 189137 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009)
+ | 17 lines Merged revisions 188833,189134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
+ | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
+ Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
+ rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
+ Modifed/added some debug messages. JIRA ABE-1835 ........
+ ................
+
+2009-04-17 20:21 +0000 [r189103] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 |
+ mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13
+ lines Prevent a crash when SIP blonde transferring an unbridged
+ call. If one attempts to use the attended transfer button on a
+ SIP phone to transfer an unbridged call (such as a call to an
+ IVR) but hangs up while the target of the transfer is still
+ ringing, we need to not crash. The problem was that ast_hangup
+ was called from outside the channel thread. AST-211 ........
+
+2009-04-17 19:46 +0000 [r189080] Sean Bright <sean.bright@gmail.com>
+
+ * main/asterisk.c, /: Merged revisions 189077 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 |
+ seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1
+ line Fix copy/paste error with 'transmit silence' flag. ........
+
+2009-04-17 17:33 +0000 [r189069] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 189010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr
+ 2009) | 12 lines Merged revisions 189009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
+ 2009) | 5 lines Make Busy() application set the CDR disposition
+ to BUSY. (closes issue #14306) Reported by: cristiandimache
+ ........ ................
+
+2009-04-17 14:48 +0000 [r188940-188949] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) |
+ 22 lines Merged revisions 188946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
+ 15 lines Fix a bug where a value used to create the channel name
+ was bogus. This commit fixes the scenario where an incoming call
+ is authenticated using a peer entry. Previously the channel name
+ was created using either the username setting from the sip.conf
+ entry or the IP address that the call came from. Now the channel
+ name will be created using the peer name itself. This commit will
+ not change the way the channel name is generated for users or
+ friends. (closes issue #14256) Reported by: Nick_Lewis Patches:
+ chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by:
+ Nick_Lewis, file ........ ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri,
+ 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4
+ lines Fix a situation where the DAHDI channel private structure
+ lock was not unlocked when it should have been. (issue AST-210)
+ ........ ................
+
+2009-04-16 22:05 +0000 [r188776-188838] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009)
+ | 14 lines Merged revisions 188835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
+ | 7 lines Only update realtime, if global option rtupdate !=
+ false (closes issue #14885) Reported by: deepesh Patches:
+ 20090413__bug14885.diff.txt uploaded by tilghman (license 14)
+ Tested by: deepesh ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500
+ (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009)
+ | 4 lines Umask should not be exported into global namespace.
+ (closes issue #14912) Reported by: jcapp ........
+ ................
+
+2009-04-15 22:12 +0000 [r188649] David Vossel <dvossel@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500
+ (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009)
+ | 12 lines National prefix inserted even when caller ID not
+ available When the caller ID is restricted, the expected behavior
+ is for the caller id to be blank. In chan_dahdi, the national
+ prefix is placed onto the callers number even if its restricted
+ (empty) causing the caller id to be the national prefix rather
+ than blank. (closes issue #13207) Reported by: shawkris Patches:
+ national_prefix.diff uploaded by dvossel (license 671) Review:
+ http://reviewboard.digium.com/r/220/ ........ ................
+
+2009-04-15 20:20 +0000 [r188473-188596] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/file.c: Merged revisions 188585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr
+ 2009) | 13 lines Merged revisions 188582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
+ 2009) | 7 lines Update ast_readvideo_callback to match
+ ast_readaudio_callback. This fixes potential refcount errors that
+ may occur on ast_filestreams. AST-208 ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 188470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 |
+ mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3
+ lines Fix a couple of queue member reference leaks. ........
+
+2009-04-14 17:43 +0000 [r188254-188415] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 188413 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 |
+ file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix
+ an incorrect clock rate when sending T140 text. (closes issue
+ #14029) Reported by: epicac ........
+
+ * /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 |
+ file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix
+ a bug with the change I made yesterday to outbound proxy support.
+ Per discussion with oej on IRC we need the actual IP address, not
+ the outbound proxy IP address, in the sa field. Upon further
+ inspection this should make the behaviour of all other uses of
+ the outbound proxy in the code. ........
+
+2009-04-14 05:46 +0000 [r188208-188212] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 188210 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 |
+ tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines
+ As suggested by Russell, warn users when their dialplan arguments
+ contain pipes, but not commas. ........
+
+ * /, utils/smsq.c: Merged revisions 188206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 |
+ tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines
+ Application delimiter is ',', not '|'. (closes issue #14881)
+ Reported by: stegro Patches: smsq.patch uploaded by stegro
+ (license 752) ........
+
+2009-04-13 19:33 +0000 [r188104] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 188102 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr
+ 2009) | 5 lines Fix another crash related to cached realtime
+ music on hold. This was another off-by-one problem caused by
+ moh_register. ........
+
+2009-04-13 16:32 +0000 [r188069] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 |
+ file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
+ Fix a bug where using an outbound proxy would cause the local
+ address to be 127.0.0.1. Copy the outbound proxy IP address into
+ the SIP dialog structure as the IP address we will be sending to.
+ This has to be done because the logic that determines what local
+ IP address to use in the SIP messages is not aware of an outbound
+ proxy being in place. It only knows what IP address we are
+ sending to. (closes issue #12006) Reported by: mnicholson
+ ........
+
+2009-04-13 14:20 +0000 [r188038] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 188032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 |
+ mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6
+ lines Set all queue variables on both the caller and member
+ channels. This allows for the variables to be accessed if a
+ member macro is run. Thanks to Grigoriy Puzankin for bringing
+ this up on the -dev list. ........
+
+2009-04-10 20:28 +0000 [r187914] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/Makefile, /: Merged revisions 187906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 |
+ jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
+ Fix module embedding for chan_h323. Include libchanh323.a in the
+ modules.link file so that all the symbols can be resolved at link
+ time. (closes issue #11966) Reported by: dome Patches:
+ issue_11966.patch uploaded by kpfleming (license 421) Tested by:
+ jpeeler ........
+
+2009-04-10 17:30 +0000 [r187767] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/sip-friends.sql,
+ contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500
+ (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10
+ Apr 2009) | 2 lines Add lastms column to the contributed table
+ designs ........ ................
+
+2009-04-10 16:54 +0000 [r187723] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, build_tools/embed_modules.xml: Merged revisions 187721 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10
+ Apr 2009) | 5 lines clean up some patterns for files to remove
+ add embedding support for bridge and test modules ........
+
+2009-04-10 16:03 +0000 [r187678] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 |
+ tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines
+ Ensure pvt is not NULL before dereferencing it. (closes issue
+ #14784) Reported by: pj ........
+
+2009-04-10 16:00 +0000 [r187676] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c, /: Merged revisions 187675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187675 |
+ russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines
+ Disable test modules by default. ........
+
+2009-04-10 03:56 +0000 [r187600] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, main/pbx.c, main/manager.c, /,
+ include/asterisk/linkedlists.h, main/features.c, main/http.c,
+ main/app.c, include/asterisk/lock.h, main/audiohook.c: Merged
+ revisions 187599 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187599 |
+ tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) | 2 lines
+ Modify headers and macros, according to Russell's suggestions on
+ the -dev list ........
+
+2009-04-09 19:14 +0000 [r187495] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 187488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r187488 | mmichelson | 2009-04-09 13:58:41 -0500 (Thu, 09 Apr
+ 2009) | 24 lines Merged revisions 187484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr
+ 2009) | 18 lines Handle a SIP race condition (reinvite before an
+ ACK) properly. RFC 5047 explains the proper course of action to
+ take if a reINVITE is received before the ACK from a previous
+ invite transaction. What we are to do is to treat the reINVITE as
+ if it were both an ACK and a reINVITE and process it normally.
+ Later, when we receive the ACK we had been expecting, we will
+ ignore it since its CSeq is less than the current iseqno of the
+ sip_pvt representing this dialog. (closes issue #13849) Reported
+ by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson
+ (license 60) Tested by: mmichelson, klaus3000 ........
+ ................
+
+2009-04-09 18:54 +0000 [r187486] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /, include/asterisk/linkedlists.h,
+ include/asterisk/lock.h: Merged revisions 187483 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500
+ (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009)
+ | 8 lines Race condition between ast_cli_command() and 'module
+ unload' could cause a deadlock. Add lock timeouts to avoid this
+ potential deadlock. (closes issue #14705) Reported by: jamessan
+ Patches: 20090320__bug14705.diff.txt uploaded by tilghman
+ (license 14) Tested by: jamessan ........ ................
+
+2009-04-09 17:43 +0000 [r187427] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 187421,187424 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu,
+ 09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using
+ cached realtime moh. The moh_register function links an mohclass
+ and then immediately unrefs the class since the container now has
+ a reference. The problem with using realtime music on hold is
+ that the class is allocated, registered, and started in one fell
+ swoop. The refcounting logic resulted in the count being off by
+ one. The same problem did not happen when using a static config
+ because the allocation and registration of an mohclass is a
+ separate operation from starting moh. This also did not affect
+ non-cached realtime moh because the classes are not registered at
+ all. I also have modified res_musiconhold to use the _t_ variants
+ of the ao2_ functions so that more info can be gleaned when
+ attempting to trace the refcounts. I found this to be incredibly
+ helpful for debugging this issue and there's no good reason to
+ remove it. (closes issue #14661) Reported by: sum ........
+ r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr
+ 2009) | 3 lines Use safe macro practices even though they really
+ aren't necessary. ........
+
+2009-04-09 17:22 +0000 [r187305-187388] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 |
+ tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines
+ Allow '/' in username portion of register; this is a regression.
+ (closes issue #14668) Reported by: Netview ........
+
+ * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions
+ 187363 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009)
+ | 10 lines Merged revisions 187362 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009)
+ | 3 lines Permit zero-length text messages in SIP. (Related to an
+ issue posted to the -users list, subject "AEL2, BASE64_DECODE and
+ hexadecimal") ........ ................
+
+ * main/asterisk.c, agi/Makefile, build_tools/cflags.xml,
+ utils/Makefile, include/asterisk.h, /, main/Makefile,
+ main/file.c, main/astfd.c (added): Merged revisions 187302 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500
+ (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009)
+ | 3 lines Add debugging mode for diagnosing file descriptor
+ leaks. (Related to issue #14625) ........ r187301 | tilghman |
+ 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops,
+ missed this file in the last commit. ........ ................
+
+2009-04-08 16:53 +0000 [r186987-187048] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 187046 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500
+ (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr
+ 2009) | 10 lines Fix a small logical error when loading moh
+ classes. We were unconditionally incrementing the number of
+ mohclasses registered. However, we should actually only increment
+ if the call to moh_register was successful. While this probably
+ has never caused problems, I noticed it and decided to fix it
+ anyway. ........ ................
+
+ * main/channel.c, /: Merged revisions 186985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr
+ 2009) | 30 lines Merged revisions 186984 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr
+ 2009) | 24 lines Make a couple of changes with regards to a new
+ message printed in ast_read(). "ast_read() called with no
+ recorded file descriptor" is a new message added after a bug was
+ discovered. Unfortunately, it seems there are a bunch of places
+ that potentially make such calls to ast_read() and trigger this
+ error message to be displayed. This commit does two things to
+ help to make this message appear less. First, the message has
+ been downgraded to a debug level message if dev mode is not
+ enabled. The message means a lot more to developers than it does
+ to end users, and so developers should take an effort to be sure
+ to call ast_read only when a channel is ready to be read from.
+ However, since this doesn't actually cause an error in operation
+ and is not something a user can easily fix, we should not spam
+ their console with these messages. Second, the message has been
+ moved to after the check for any pending masquerades. ast_read()
+ being called with no recorded file descriptor should not
+ interfere with a masquerade taking place. This could be seen as a
+ simple way of resolving issue #14723. However, I still want to
+ try to clear out the existing ways of triggering this message,
+ since I feel that would be a better resolution for the issue.
+ ........ ................
+
+2009-04-08 05:07 +0000 [r186900] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 |
+ tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines
+ Add lastms to the require API call. ........
+
+2009-04-08 00:10 +0000 [r186835-186844] Mark Michelson <mmichelson@digium.com>
+
+ * /, formats/format_wav.c, formats/format_wav_gsm.c: Merged
+ revisions 186842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr
+ 2009) | 14 lines Merged revisions 186841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr
+ 2009) | 8 lines Fix a few typos of the word "frequency." (closes
+ issue #14842) Reported by: jvandal Patches: frequency-typo.diff
+ uploaded by jvandal (license 413) ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 |
+ mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7
+ lines Fix bad merge from fix for issue 13867. (closes issue
+ #14686) Reported by: davidw ........
+
+ * main/channel.c, /: Merged revisions 186833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr
+ 2009) | 15 lines Merged revisions 186832 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr
+ 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a
+ p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
+ warning sounds will not be properly played to either party of the
+ bridge. (closes issue #14845) Reported by: adomjan ........
+ ................
+
+2009-04-07 22:33 +0000 [r186806] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_macro.c: Merged revisions 186799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009)
+ | 10 lines Merged revisions 186775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009)
+ | 3 lines Fix Macro documentation to match current (and intended)
+ behavior. (See -dev mailing list) ........ ................
+
+2009-04-07 20:53 +0000 [r186722] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, /: Merged revisions 186720 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr
+ 2009) | 12 lines Merged revisions 186719 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr
+ 2009) | 6 lines Ensure that \r\n is printed after the ActionID in
+ an OriginateResponse. (closes issue #14847) Reported by: kobaz
+ ........ ................
+
+2009-04-03 20:21 +0000 [r186466] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500
+ (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr
+ 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not
+ properly switch formats when requested Don't offer
+ AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
+ provide a slight performance benefit, the translation core in
+ Asterisk has some flaws when a channel driver offers multiple raw
+ formats. this fix is much simpler than fixing the translation
+ core to solve that issue (although that will be done later).
+ ........ ................
+
+2009-04-03 20:04 +0000 [r186448] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
+ revisions 186444,186447 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009)
+ | 14 lines Merged revisions 186415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009)
+ | 7 lines Distinguish in a sent email between simple sends and
+ forwards. (closes issue #11678) Reported by: jamessan Patches:
+ 20090330__bug11678.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman, lmadsen ........ ................ r186447 |
+ tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines
+ Merged revisions 186445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009)
+ | 2 lines Found a conflict in the last commit, due to multiple
+ targets ........ ................
+
+2009-04-03 16:38 +0000 [r186381] David Vossel <dvossel@digium.com>
+
+ * /, main/audiohook.c: Merged revisions 186379 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 |
+ dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 4 lines
+ audio_audiohook_write_list() did not correctly update sample size
+ after ast_translate. audio_audiohook_write_list() did not take
+ into account that the sample size may change after translation
+ depending on if the original frame is is 8khz or 16khz. the
+ sample size is now updated after translating to reflect this
+ possibility. This caused the audio on the receiving end to sound
+ terrible. Thanks to jcolp and mmichelson for helping me work this
+ out. (issue AST-197) ........
+
+2009-04-03 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1.0-rc4 released.
+
+2009-04-03 15:54 +0000 [r186323] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/crypto.h, /: Merged revisions 186321 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri,
+ 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
+ lines Fix a problem with the crypto variable definitions not
+ actually being defined properly. (closes issue #14804) Reported
+ by: jvandal ........ ................
+
+2009-04-03 14:33 +0000 [r186288] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr
+ 2009) | 20 lines Fix the ability to retrieve voicemail messages
+ from IMAP. A recent change made interactive vm_states no longer
+ get added to the list of vm_states and instead get stored in
+ thread-local storage. In trunk and all the 1.6.X branches, the
+ problem is that when we search for messages in a voicemail box,
+ we would attempt to update the appropriate vm_state struct by
+ directly searching in the list of vm_states instead of using the
+ get_vm_state_by_imap_user function. This meant we could not find
+ the interactive vm_state that we wanted. (closes issue #14685)
+ Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
+ (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........
+
+2009-04-03 02:06 +0000 [r186232] Russell Bryant <russell@digium.com>
+
+ * cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009)
+ | 29 lines Merged revisions 186229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
+ | 21 lines Fix a memory leak in cdr_radius. I came across this
+ while doing some testing of my ast_channel_ao2 branch. After
+ running a test overnight that generated over 5 million calls,
+ Asterisk had taken up about 1 GB of my system memory. So, I
+ re-ran the test with MALLOC_DEBUG turned on. However, it showed
+ no leaks in Asterisk during the test, even though Asterisk was
+ still consuming it somehow. Instead, I turned to valgrind, which
+ when run with --leak-check=full, told me exactly where the leak
+ came from, which was from allocations inside the radiusclient-ng
+ library. This explains why MALLOC_DEBUG did not report it. After
+ a bit of analysis, I found that we were leaking a little bit of
+ memory every time a CDR record was passed to cdr_radius. I don't
+ actually have a radius server set up to receive CDR records.
+ However, I always have my development systems compile and install
+ all modules. In addition to making sure there are not build
+ errors across modules, always loading modules helps find bugs
+ like this, too, so it is strongly recommend for all developers.
+ ........ ................
+
+2009-04-02 21:59 +0000 [r186177] Mark Michelson <mmichelson@digium.com>
+
+ * configs/features.conf.sample, /: Merged revisions 186175 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500
+ (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
+ 2009) | 5 lines Fix instructions in one-step parking comment to
+ make more sense. Changed a capital K to a lowercase k. ........
+ ................
+
+2009-04-02 17:27 +0000 [r186108] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500
+ (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
+ Apr 2009) | 3 lines ensure that the buffer passed to
+ DAHDI_SET_BUFINFO is fully initialized ........ ................
+
+2009-04-02 17:14 +0000 [r186062] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 186060 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009)
+ | 16 lines Merged revisions 186059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
+ (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
+ Apr 2009) | 2 lines Fix for AST-2009-003 ........
+ ................ ................
+
+2009-04-02 13:53 +0000 [r185956] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500
+ (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr
+ 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
+ DAHDI_GET_PARAMS ioctls were recently corrected to show that they
+ do, in fact, read data from userspace as part of their work. due
+ to this fix, valgrind now reports a number of cases where
+ chan_dahdi passed an uninitialized (or partially) buffer to these
+ ioctls, which could lead to unexpected behavior. this patch
+ corrects chan_dahdi to ensure that buffers passed to these ioctls
+ are always fully initialized. ........ ................
+
+2009-04-01 19:06 +0000 [r185848] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009)
+ | 16 lines Merged revisions 185845 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
+ | 10 lines Fixes issue with dropped calles due to re-Invite glare
+ and re-Invites never executing after a 491 Acknowledgement for
+ 491 responses were never being processed because it didn't match
+ our pending invite's seqno. Since the ACK was never processed,
+ the 491 frame would continue to be retransmitted until eventually
+ the call was dropped due to max retries. Now during a pending
+ invite, if we receive another invite, we send an 491 and hold on
+ to that glare invite's seqno in the "glareinvite" variable for
+ that sip_pvt struct. When ACK's are received, we first check to
+ see if it is in response to our pending invite, if not we check
+ to see if it is in response to a glare invite. In this case, it
+ is in response to the glare invite and must be dealt with or the
+ call is dropped. I've changed the wait time for resending the
+ re-Invite after receving a 491 response to comply with RFC 3261.
+ Before this patch the scheduled re-Invite would only change a
+ flag indicating that the re-Invite should be sent out, now it
+ actually sends it out as well. (closes issue #12013) Reported by:
+ alx Review: http://reviewboard.digium.com/r/213/ ........
+ ................
+
+2009-04-01 13:50 +0000 [r185774] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 185772 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009)
+ | 14 lines Merged revisions 185771 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009)
+ | 6 lines Fix a case where DTMF could bypass audiohooks. This
+ change fixes a situation where an audiohook that wants DTMF would
+ not actually get it. This is in the code path where we end DTMF
+ digit length emulation while handling a NULL frame. ........
+ ................
+
+2009-03-31 22:38 +0000 [r185666] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils, /: Merged revisions 185664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 |
+ kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line
+ ignore copied (generated) file ........
+
+2009-03-31 22:05 +0000 [r185471-185602] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185600 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar
+ 2009) | 12 lines Merged revisions 185599 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar
+ 2009) | 6 lines Fix crash that would occur if an empty member was
+ specified in queues.conf. (closes issue #14796) Reported by: pida
+ ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500
+ (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar
+ 2009) | 8 lines Fix Russian voicemail intro to say the word
+ "messages" properly. (closes issue #14736) Reported by: chappell
+ Patches: voicemail_no_messages.diff uploaded by chappell (license
+ 8) ........ ................
+
+2009-03-31 17:48 +0000 [r185427] David Brooks <dbrooks@digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 185363 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500
+ (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009)
+ | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains
+ extra whitespaces To drill into the xmpp to find the capabilities
+ between channels, chan_gtalk calls iks_child() and iks_next().
+ iks_child() and iks_next() are functions in the iksemel xml
+ parsing library that traverse xml nodes. The bug here is that
+ both iks_child() and iks_next() will return the next iks_struct
+ node *regardless* of type. chan_gtalk expects the next node to be
+ of type IKS_TAG, which in most cases, it is, but in this case (a
+ call being made from the Empathy IM client), there exists
+ iks_struct nodes which are not IKS_TAG data (they are extraneous
+ whitespaces), and chan_gtalk doesn't handle that case, so
+ capabilities don't match, and a call cannot be made.
+ iks_first_tag() and iks_next_tag(), on the other hand, will not
+ return the very next iks_struct, but will check to see if the
+ next iks_struct is of type IKS_TAG. If it isn't, it will be
+ skipped, and the next struct of type IKS_TAG it finds will be
+ returned. This assures that chan_gtalk will find the iks_struct
+ it is looking for. This fix simply changes all calls to
+ iks_child() and iks_next() to become calls to iks_first_tag() and
+ iks_next_tag(), which resolves the capability matching. The
+ following is a payload listing from Empathy, which, due to the
+ extraneous whitespace, will not be parsed correctly by iksemel:
+ <iq from='dbrooksjab@235-22-24-10/Telepathy'
+ to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
+ <session xmlns='http://www.google.com/session'
+ initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
+ id='1837267342'> <description
+ xmlns='http://www.google.com/session/phone'> <payload-type
+ clockrate='16000' name='speex' id='96'/> <payload-type
+ clockrate='8000' name='PCMA' id='8'/> <payload-type
+ clockrate='8000' name='PCMU' id='0'/> <payload-type
+ clockrate='90000' name='MPA' id='97'/> <payload-type
+ clockrate='16000' name='SIREN' id='98'/> <payload-type
+ clockrate='8000' name='telephone-event' id='99'/> </description>
+ </session> </iq> Review: http://reviewboard.digium.com/r/181/
+ ........ ................
+
+2009-03-31 14:57 +0000 [r185263] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 |
+ russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines
+ Don't free() an astobj2 object. (closes issue #14672) Reported
+ by: makoto ........
+
+2009-03-31 14:10 +0000 [r185199] Joshua Colp <jcolp@digium.com>
+
+ * /, main/audiohook.c: Merged revisions 185197 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) |
+ 15 lines Merged revisions 185196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8
+ lines Fix crash when moving audiohooks between channels. Handle
+ the scenario where we are called to move audiohooks between
+ channels and the source channel does not actually have any on it.
+ (closes issue #14734) Reported by: corruptor ........
+ ................
+
+2009-03-30 20:50 +0000 [r185126-185127] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, /, configs/misdn.conf.sample: Merged
+ revisions 185123 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009)
+ | 9 lines Merged revisions 185121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009)
+ | 1 line Update the channel allocation method documentation.
+ ........ ................
+
+ * channels/misdn/isdn_lib.c, /: Merged revisions 185122 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500
+ (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009)
+ | 19 lines Make chan_misdn BRI TE side normally defer channel
+ selection to the NT side. Channel allocation collisions are not
+ handled by chan_misdn very well. This patch simply avoids the
+ problem for BRI only. For PRI, allocation collisions are still
+ possible but less likely since there are simply more channels
+ available and each end could use a different allocation strategy.
+ misdn.conf options available: te_choose_channel - Use to force
+ the TE side to allocate channels. method - Specify the channel
+ allocation strategy. (closes issue #13488) Reported by:
+ Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich
+ Tested by: crich, siepkes, festr ........ ................
+
+2009-03-30 16:47 +0000 [r185088] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185072 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar
+ 2009) | 45 lines Merged revisions 185031 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar
+ 2009) | 39 lines Fix queue weight behavior so that calls in
+ low-weight queues are not inappropriately blocked. (This is
+ copied and pasted from the review request I made for this patch)
+ Asterisk has some odd behavior when queue weights are used. The
+ current logic used when potentially calling a queue member is: If
+ the member we are going to call is part of another queue and
+ _that other queue has any callers in it_ and has a higher weight
+ than the queue we are calling from, then don't try to contact
+ that member. The issue here is what I have marked with
+ underscores. If the higher-weighted queue has any callers in it
+ at all, then the queue member will be unreachable from the
+ lower-weighted queue. This has the potential to be really really
+ bad if using a queue strategy, such as leastrecent or
+ fewestcalls, with the potential to call the same member
+ repeatedly. The fix proposed by garychen on issue 13220 is very
+ simple and, as far as I can see, works well for this situation.
+ With this set of changes, the logic used becomes: If the member
+ we are going to call is part of another queue, the other queue
+ has a higher weight than the queue we are calling from, and the
+ higher weight queue has at least as many callers as available
+ members, then do not try to contact the queue member. If the
+ higher weighted queue has fewer callers than available members,
+ then there is no reason to deny the call to this member since the
+ other queue can afford to spare a member. Since the fix involved
+ writing a generic function for determining the number of
+ available members in the queue, I also modified the is_our_turn
+ function to make use of the new num_available_members function to
+ determine if it is our turn to try calling a member. There is one
+ small behavior change. Before writing this patch, if you had
+ autofill disabled, then if you were the head caller in a queue,
+ you would automatically be told that it was your turn to try
+ calling a member. This did not take into account whether there
+ were actually any queue members available to take the call. Now
+ we actually make sure there is at least one member available to
+ take the call if autofill is disabled. (closes issue #13220)
+ Reported by: garychen Review:
+ http://reviewboard.digium.com/r/202/ ........ ................
+
+2009-03-30 14:41 +0000 [r184950] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) |
+ 21 lines Merged revisions 184947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) |
+ 14 lines Improve our handling of T38 in the initial INVITE from a
+ device. We now answer with matching media streams to what is
+ requested. If an INVITE is received with both a T38 and RTP media
+ stream this means we answer with both. For any outgoing calls
+ created as a result of this inbound one no T38 is requested in
+ the initial INVITE. Instead if we start receiving udptl packets
+ we trigger a reinvite on the outbound side. (closes issue #12437)
+ Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu
+ Review: http://reviewboard.digium.com/r/208/ ........
+ ................
+
+2009-03-30 13:57 +0000 [r184912] Russell Bryant <russell@digium.com>
+
+ * channels/h323/Makefile.in, /: Merged revisions 184910 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30
+ Mar 2009) | 4 lines Fix build error when chan_h323 is not being
+ built. (reported by cai1982 in #asterisk-dev) ........
+
+2009-03-29 05:52 +0000 [r184840-184845] Russell Bryant <russell@digium.com>
+
+ * apps/app_followme.c, /: Merged revisions 184843 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009)
+ | 13 lines Merged revisions 184842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009)
+ | 5 lines Ensure targs variable is fully initialized. (closes
+ issue #14758) Reported by: tim_ringenbach ........
+ ................
+
+ * channels/Makefile, /: Merged revisions 184838 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 |
+ russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines
+ Simplify chan_h323 build to not require a second run of "make".
+ (closes issue #14715) Reported by: jthurman Patches:
+ h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license
+ 614) Tested by: tzafrir, russell ........
+
+2009-03-27 19:17 +0000 [r184765] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c, main/timing.c, main/channel.c, /,
+ include/asterisk/timing.h, include/asterisk/channel.h: Merged
+ revisions 184762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 |
+ kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12
+ lines Improve timing interface to remember which provider
+ provided a timer The ability to load/unload timing interfaces is
+ nice, but it means that when a timer is allocated, it may come
+ from provider A, but later provider B becomes the 'preferred'
+ provider. If this happens, all timer API calls on the timer that
+ was provided by provider A will actually be handed to provider B,
+ which will say WTF and return an error. This patch changes the
+ timer API to include a pointer to the provider of the timer
+ handle so that future operations on the timer will be forwarded
+ to the proper provider. (closes issue #14697) Reported by: moy
+ Review: http://reviewboard.digium.com/r/211/ ........
+
+2009-03-27 18:09 +0000 [r184728] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27
+ Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure
+ we use the best RNG available. ........
+
+2009-03-27 15:54 +0000 [r184675] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 184673 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 |
+ file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix
+ speech structure leak in the AGI speech recognition integration.
+ The AGI dialplan applications did not destroy the speech
+ structure automatically if it was not destroyed by the running
+ AGI script. They will now do this. (issue LUMENVOX-15) ........
+
+2009-03-27 14:04 +0000 [r184631] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /,
+ res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions
+ 184630 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 |
+ russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines
+ Change g_eid to ast_eid_default. ........
+
+2009-03-27 13:22 +0000 [r184587] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) |
+ 16 lines Merged revisions 184565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9
+ lines Fix an issue where nat=yes would not always take effect for
+ the RTP session on outgoing calls. If calls were placed using an
+ IP address or hostname the global nat setting was copied over but
+ was not set on the RTP session itself. This caused the RTP stack
+ to not perform symmetric RTP actions. (closes issue #14546)
+ Reported by: acunningham ........ ................
+
+2009-03-27 02:25 +0000 [r184513-184547] Russell Bryant <russell@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009)
+ | 20 lines Fix some issues with rwlock corruption that caused
+ deadlock like symptoms. When dvossel and I were doing some load
+ testing last week, we noticed that we could make Asterisk trunk
+ lock up instantly when we started generating a bunch of calls.
+ The backtraces of locked threads were bizarre, and many were
+ stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a
+ number of places where a backtrace would be loaded into an
+ invalid index of the backtrace array. It's an off by one error,
+ which ends up writing over the rwlock itself. 2) Ensure that in
+ the array of held locks, we NULL out an index once it is not
+ being used so that it's not confusing when analyzing its
+ contents. 3) Remove a bunch of logging referring to an rwlock
+ operating being done with "deep reentrancy". It is normal for
+ _many_ threads to hold a read lock on an rwlock. ........
+
+ * /, main/file.c: Merged revisions 184515 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 |
+ russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines
+ Don't act surprised if we get a -1 indication. ........
+
+ * include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26
+ Mar 2009) | 2 lines Pass more useful information through to lock
+ tracking when DEBUG_THREADS is on. ........
+
+2009-03-26 22:19 +0000 [r184451] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 184448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar
+ 2009) | 9 lines Merged revisions 184447 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar
+ 2009) | 3 lines use new, improved 8kHz prompts ........
+ ................
+
+2009-03-26 21:18 +0000 [r184394] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_test.c: Merged revisions 184389 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184389 | dvossel | 2009-03-26 16:09:37 -0500 (Thu, 26 Mar 2009)
+ | 14 lines Merged revisions 184388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009)
+ | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF
+ 8 app_test was failing when sending the last DTMF digit, 8,
+ because of the 100ms pause issued after DTMF is sent. During this
+ pause the other side would hang up causing the test to look like
+ it failed. Now the other side waits a second before hanging up.
+ (closes issue #12442) Reported by: tzafrir ........
+ ................
+
+2009-03-25 22:13 +0000 [r184325-184345] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 184344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 |
+ russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines
+ Remove unneeded AST_LIST_ENTRY() and comment on the purpose of
+ ast_event_ref. ........
+
+ * channels/chan_iax2.c, channels/chan_dahdi.c,
+ include/asterisk/event.h, channels/chan_skinny.c, res/ais/evt.c,
+ main/event.c, include/asterisk/strings.h, main/asterisk.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c,
+ channels/chan_unistim.c, include/asterisk/devicestate.h, /,
+ channels/chan_sip.c, main/devicestate.c,
+ include/asterisk/_private.h: Merged revisions 184339 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009)
+ | 35 lines Improve performance of the ast_event cache
+ functionality. This code comes from
+ svn/asterisk/team/russell/event_performance/. Here is a summary
+ of the changes that have been made, in order of both invasiveness
+ and performance impact, from smallest to largest. 1) Asterisk
+ 1.6.1 introduces some additional logic to be able to handle
+ distributed device state. This functionality comes at a cost. One
+ relatively minor change in this patch is that the extra
+ processing required for distributed device state is now
+ completely bypassed if it's not needed. 2) One of the things that
+ I noticed when profiling this code was that a _lot_ of time was
+ spent doing string comparisons. I changed the way strings are
+ represented in an event to include a hash value at the front. So,
+ before doing a string comparison, we do an integer comparison on
+ the hash. 3) Finally, the code that handles the event cache has
+ been re-written. I tried to do this in a such a way that it had
+ minimal impact on the API. I did have to change one API call,
+ though - ast_event_queue_and_cache(). However, the way it works
+ now is nicer, IMO. Each type of event that can be cached (MWI,
+ device state) has its own hash table and rules for hashing and
+ comparing objects. This by far made the biggest impact on
+ performance. For additional details regarding this code and how
+ it was tested, please see the review request. (closes issue
+ #14738) Reported by: russell Review:
+ http://reviewboard.digium.com/r/205/ ........
+
+ * /: add reviewboard:url property.
+
+2009-03-25 19:26 +0000 [r184282] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 |
+ file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix
+ issue with a T38 reinvite being sent even if not configured to do
+ so. If we receive a T38 request negotiate control frame we should
+ only attempt to do so if the option is enabled on the dialog.
+ ........
+
+2009-03-25 15:12 +0000 [r184223] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/asterisk.c, /: Merged revisions 184220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) |
+ 19 lines Merged revisions 184188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) |
+ 13 lines Avoid destroying the CLI line when moving the cursor
+ backward and trying to autocomplete. When moving the cursor
+ backward and pressing TAB to autocomplete, a NULL is put in the
+ line and we are loosing what we have already wrote after the
+ actual cursor position. (closes issue #14373) Reported by: eliel
+ Patches: asterisk.c.patch uploaded by eliel (license 64) Tested
+ by: lmadsen ........ ................
+
+2009-03-25 01:55 +0000 [r184149] Russell Bryant <russell@digium.com>
+
+ * main/timing.c, utils/Makefile, /, include/asterisk/compat.h:
+ Merged revisions 184147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 |
+ russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines
+ Fix build issues on Mac OSX. (closes issue #14714) Reported by:
+ ygor ........
+
+2009-03-24 22:42 +0000 [r184081] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar
+ 2009) | 15 lines Merged revisions 184078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar
+ 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero.
+ The 'digit' variable is guaranteed to be non-NULL, so the if
+ statement could never evaluate true. Changing to ast_strlen_zero
+ makes the logic correct. This was found while reviewing
+ ast_channel_ao2 code review. ........ ................
+
+2009-03-24 21:47 +0000 [r184039] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009)
+ | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low
+ and =medium The default codec configuration for chan_iax2 is
+ bandwidth=low. I noticed slin16 being negotiated as the codec in
+ some test calls, but that no longer happens after this change.
+ ........
+
+2009-03-24 15:28 +0000 [r183867-183916] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 183914 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500
+ (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009)
+ | 3 lines Additionally note that the operator option needs an 'o'
+ extension. (Related to issue #14731) ........ ................
+
+ * /, main/http.c: Merged revisions 183865 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 |
+ tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines
+ Allow browsers to cache images and other static content. (This is
+ a regression over 1.4) ........
+
+2009-03-23 18:59 +0000 [r183768] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 183766 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar
+ 2009) | 13 lines Merged revisions 183700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar
+ 2009) | 7 lines Fix a memory leak in res_monitor.c The only way
+ that this leak would occur is if Monitor were started using the
+ Manager interface and no File: header were given. Discovered
+ while reviewing the ast_channel_ao2 review request. ........
+ ................
+
+2009-03-23 18:12 +0000 [r183703] Leif Madsen <lmadsen@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009)
+ | 7 lines Fixes a documentation error introduced during the CLI
+ cleanup at AstriDevCon 2008. (closes issue #14655) Reported by:
+ ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728)
+ Tested by: lmadsen ........
+
+2009-03-20 17:08 +0000 [r183563] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183560 | russell | 2009-03-20 12:00:58 -0500
+ (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009)
+ | 2 lines Fix a crash in IAX2 registration handling found during
+ load testing with dvossel. ........ ................
+
+2009-03-19 20:33 +0000 [r183438] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/features.h, apps/app_dial.c, /, main/features.c:
+ Merged revisions 183436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009)
+ | 13 lines Merged revisions 183386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009)
+ | 6 lines Cleaning up a few things in detect disconnect patch
+ Initialized ast_call_feature in detect_disconnect to avoid
+ accessing uninitialized memory. Cleaned up /param tags in
+ features.h. No longer send dynamic features in
+ ast_feature_detect. issue #11583 ........ ................
+
+2009-03-19 19:19 +0000 [r183333] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500
+ (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009)
+ | 8 lines Delay signalling progress until a PRI channel really
+ signals progress. (closes issue #13034) Reported by: klaus3000
+ Patches: 20090316__bug13034.diff.txt uploaded by tilghman
+ (license 14) patch_trunk_183progress_klaus3000.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000 ........
+ ................
+
+2009-03-19 18:14 +0000 [r183249] Russell Bryant <russell@digium.com>
+
+ * main/loader.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Merged revisions 183242 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009)
+ | 10 lines Merged revisions 183241 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009)
+ | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving
+ like expected. ........ ................
+
+2009-03-19 18:11 +0000 [r183246] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 183244 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 |
+ mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16
+ lines Fix a memory leak associated with queues. For every attempt
+ that app_queue made to place an outbound call to a queue member,
+ we would allocate a queue_end_bridge structure. When the bridge
+ for the call had completed, we would free the structure.
+ Unfortunately not all call attempts actually end up bridged to a
+ member, so we need to be more selective of when to allocate the
+ structure. With this change, the allocation occurs in an area
+ where we can guarantee that the call will be bridged. (closes
+ issue #14680) Reported by: caspy Patches: 14680.patch uploaded by
+ mmichelson (license 60) Tested by: caspy ........
+
+2009-03-19 17:08 +0000 [r183198] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/features.h, apps/app_dial.c, /, main/features.c:
+ Merged revisions 183172 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009)
+ | 20 lines Merged revisions 183126 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009)
+ | 17 lines Allow disconnect feature before a call is bridged
+ feature.conf has a disconnect option. By default this option is
+ set to '*', but it could be anything. If a user wishes to
+ disconnect a call before the other side answers, only '*' will
+ work, regardless if the disconnect option is set to something
+ else. This is because features are unavailable until bridging
+ takes place. The default disconnect option, '*', was hardcoded in
+ app_dial, which doesn't make any sense from a user perspective
+ since they may expect it to be something different. This patch
+ allows features to be detected from outside of the bridge, but
+ not operated on. In this case, the disconnect feature can be
+ detected before briding and handled outside of features.c.
+ (closes issue #11583) Reported by: sobomax Patches:
+ patch-apps__app_dial.c uploaded by sobomax (license 359)
+ 11583.latest-patch uploaded by murf (license 17)
+ detect_disconnect.diff uploaded by dvossel (license 671) Tested
+ by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/
+ ........ ................
+
+2009-03-19 16:09 +0000 [r183121] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar
+ 2009) | 20 lines Merged revisions 183115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar
+ 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls
+ would erroneously report the device as "in use." A user was
+ having an issue where if an outgoing SIP call was canceled, the
+ SIP device would remain in use if we had not received any
+ response to the initial INVITE we sent out. The SIP device would
+ remain in use until the autocongestion timer was exhausted. I
+ tracked down the cause of this to be the section of code I am
+ removing here. I asked several people what the purpose of this
+ code was meant to be, but no one could give me any sort of answer
+ as to why this was here. The person who was having this issue has
+ been using this patch for several months and it has stopped the
+ problems they have had. AST-196 ........ ................
+
+2009-03-19 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.1.0-rc3
+
+2009-03-19 15:43 +0000 [r183067-183110] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 |
+ file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
+ Improve our triggering of a T38 switchover internally when
+ triggered by a received reinvite. Previously we reached across
+ the channel bridge to get the other party's SIP dialog structure
+ in order to trigger an outgoing reinvite. This is extremely
+ dangerous to do and only works if bridged to another SIP channel.
+ This patch changes this to use the T38 control frame method of
+ requesting a switchover. This change also causes the SIP channel
+ driver to propogate back whether the switchover worked or not
+ instead of blindly accepting the incoming T38 reinvite. Review:
+ http://reviewboard.digium.com/r/200/ ........
+
+ * main/channel.c, /: Merged revisions 183057 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 |
+ file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix
+ an issue where a T38 control frame would get dropped. If two
+ channels were bridged together using a generic bridge the T38
+ control frame would get passed up instead of being indicated on
+ the other channel. ........
+
+2009-03-18 21:19 +0000 [r183030] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18
+ Mar 2009) | 4 lines Add some code removed by mistake from commit
+ 182722 that works around a file descriptor leak in versions of
+ PWLib prior to 1.12.0. ........
+
+2009-03-18 14:32 +0000 [r182946] Russell Bryant <russell@digium.com>
+
+ * main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c,
+ configure, apps/app_mp3.c, res/res_agi.c,
+ include/asterisk/poll-compat.h, channels/chan_alsa.c,
+ main/asterisk.c, apps/app_nbscat.c, /, main/Makefile,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/io.h, main/utils.c, include/asterisk/channel.h:
+ Merged revisions 182847 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009)
+ | 52 lines Merged revisions 182810 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009)
+ | 44 lines Fix cases where the internal poll() was not being used
+ when it needed to be. We have seen a number of problems caused by
+ poll() not working properly on Mac OSX. If you search around,
+ you'll find a number of references to using select() instead of
+ poll() to work around these issues. In Asterisk, we've had poll.c
+ which implements poll() using select() internally. However, we
+ were still getting reports of problems. vadim investigated a bit
+ and realized that at least on his system, even though we were
+ compiling in poll.o, the system poll() was still being used. So,
+ the primary purpose of this patch is to ensure that we're using
+ the internal poll() when we want it to be used. The changes are:
+ 1) Remove logic for when internal poll should be used from the
+ Makefile. Instead, put it in the configure script. The logic in
+ the configure script is the same as it was in the Makefile.
+ Ideally, we would have a functionality test for the problem, but
+ that's not actually possible, since we would have to be able to
+ run an application on the _target_ system to test poll()
+ behavior. 2) Always include poll.o in the build, but it will be
+ empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
+ throughout the source tree to ast_poll(). I feel that it is good
+ practice to give the API call a new name when we are changing its
+ behavior and not using the system version directly in all cases.
+ So, normally, ast_poll() is just redefined to poll(). On systems
+ where AST_POLL_COMPAT is defined, ast_poll() is redefined to
+ ast_internal_poll(). 4) Change poll() in main/poll.c to be
+ ast_internal_poll(). It's worth noting that any code that still
+ uses poll() directly will work fine (if they worked fine before).
+ So, for example, out of tree modules that are using poll() will
+ not stop working or anything. However, for modules to work
+ properly on Mac OSX, ast_poll() needs to be used. (closes issue
+ #13404) Reported by: agalbraith Tested by: russell, vadim
+ http://reviewboard.digium.com/r/198/ ........ ................
+
+2009-03-17 20:52 +0000 [r182724] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /,
+ channels/h323/ast_h323.cxx, configure,
+ autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h,
+ channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions
+ 182722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 |
+ jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
+ Allow H.323 Plus library to be used in addition to the OpenH323
+ library Chan_h323 can now be compiled against both the previously
+ supported versions of OpenH323 as well as the current H.323 Plus
+ (version 1.20.2). The configure script has been modified to look
+ in the default install location of h323 to hopefully help avoid
+ using the environment variables OPENH323DIR and PWLIBDIR. Also,
+ the CLI command "h323 show version" has been added which
+ indicates which version of h323 is in use. (closes issue #11261)
+ Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
+ uploaded by jthurman (license 614) ........
+
+2009-03-17 15:31 +0000 [r182570] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 182553 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 |
+ russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines
+ Tweak the handling of the frame list inside of ast_answer(). This
+ does not change any behavior, but moves the frames from the local
+ frame list back to the channel read queue using an O(n) algorithm
+ instead of O(n^2). ........
+
+2009-03-17 15:00 +0000 [r182527-182533] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/channel.c, /: Merged revisions 182530 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 |
+ kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2
+ lines correct logic flaw in ast_answer() changes in r182525
+ ........
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ Merged revisions 182525 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 |
+ kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11
+ lines Improve behavior of ast_answer() to not lose incoming
+ frames ast_answer(), when supplied a delay before returning to
+ the caller, use ast_safe_sleep() to implement the delay.
+ Unfortunately during this time any incoming frames are discarded,
+ which is problematic for T.38 re-INVITES and other sorts of
+ channel operations. When a delay is not passed to ast_answer(),
+ it still delays for up to 500 milliseconds, waiting for media to
+ arrive. Again, though, it discards any control frames, or
+ non-voice media frames. This patch rectifies this situation, by
+ storing all incoming frames during the delay period on a list,
+ and then requeuing them onto the channel before returning to the
+ caller. http://reviewboard.digium.com/r/196/ ........
+
+2009-03-17 05:54 +0000 [r182452] Tilghman Lesher <tlesher@digium.com>
+
+ * main/db.c, /: Merged revisions 182450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009)
+ | 14 lines Merged revisions 182449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009)
+ | 7 lines Fix race in astdb The underlying db1 implementation
+ does not fully isolate the pages retrieved from astdb, so the
+ lock protecting accesses needs to be extended until the copy from
+ the shared memory structure is done. (closes issue #14682)
+ Reported by: makoto ........ ................
+
+2009-03-16 17:53 +0000 [r182284] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 182282 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500
+ (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009)
+ | 7 lines Randomize IAX2 encryption padding The 16-32 byte random
+ padding at the beginning of an encrypted IAX2 frame turns out to
+ not be all that random at all. This patch calls ast_random to
+ fill the padding buffer with random data. The padding is
+ randomized at the beginning of every encrypted call and for every
+ encrypted retransmit frame. Review:
+ http://reviewboard.digium.com/r/193/ ........ ................
+
+2009-03-16 17:38 +0000 [r182280] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /, funcs/func_env.c: Merged revisions
+ 182211,182278 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009)
+ | 14 lines Merged revisions 182208 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009)
+ | 7 lines Fixup glare detection, to fix a memory leak of a local
+ pvt structure. (closes issue #14656) Reported by: caspy Patches:
+ 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy ........ ................ r182278 | tilghman |
+ 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines Fix an
+ off-by-one error in the FILE() function, and extend FILE()'s
+ length parameter to work like variable substitution. Previously,
+ FILE() returned one less character than specified, due to the
+ terminating NULL. Both the offset and length parameters now
+ behave identically to the way variable substitution offsets and
+ lengths also work. (closes issue #14670) Reported by: BMC
+ ................
+
+2009-03-16 14:00 +0000 [r182173] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 182171 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 |
+ file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix
+ a memory leak in the ast_answer / __ast_answer API call. For a
+ channel that is not yet answered this API call will wait until a
+ voice frame is received on the channel before returning. It does
+ this by waiting for frames on the channel and reading them in.
+ The frames read in were not freed when they should have been.
+ ........
+
+2009-03-13 21:27 +0000 [r182068-182123] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 182121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 |
+ mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6
+ lines Change faulty comparison used when announcing average hold
+ minutes and seconds (closes issue #14227) Reported by: caspy
+ ........
+
+ * /, main/features.c: Merged revisions 182029 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar
+ 2009) | 41 lines Merged revisions 181990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar
+ 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and
+ peer when interpreting DTMF. Dynamic features defined in the
+ applicationmap section of features.conf allow one to specify
+ whether the caller, callee, or both have the ability to use the
+ feature. The documentation in the features.conf.sample file could
+ be interpreted to mean that one only needs to set the
+ DYNAMIC_FEATURES channel variable on the calling channel in order
+ to allow for the callee to be able to use the features which he
+ should have permission to use. However, the DYNAMIC_FEATURES
+ variable would only be read from the channel of the participant
+ that pressed the DTMF sequence to activate the feature. The
+ result of this was that the callee was unable to use dynamic
+ features unless the dialplan writer had taken measures to be sure
+ that the DYNAMIC_FEATURES variable was set on the callee's
+ channel. This commit changes the behavior of
+ ast_feature_interpret to concatenate the values of
+ DYNAMIC_FEATURES from both parties involved in the bridge. The
+ features themselves determine who has permission to use them, so
+ there is no reason to believe that one side of the bridge could
+ gain the ability to perform an action that they should not have
+ the ability to perform. Kevin Fleming pointed out on the
+ asterisk-users list that the typical way that this was worked
+ around in the past was by setting _DYNAMIC_FEATURES on the
+ calling channel so that the value would be inherited by the
+ called channel. While this works, the documentation alone is not
+ enough to figure out why this is necessary for the callee to be
+ able to use dynamic features. In this particular case, changing
+ the code to match the documentation is safe, easy, and will
+ generally make things easier for people for future installations.
+ This bug was originally reported on the asterisk-users list by
+ David Ruggles. (closes issue #14657) Reported by: mmichelson
+ Patches: 14657.patch uploaded by mmichelson (license 60) ........
+ ................
+
+2009-03-13 17:29 +0000 [r182042] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 182022 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 |
+ file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix
+ an issue with requesting a T38 reinvite before the call is
+ answered. The code responsible for sending the T38 reinvite did
+ not check if an INVITE was already being handled. This caused
+ things to get confused and the call to fail. The code now defers
+ sending the T38 reinvite until the current INVITE is done being
+ handled. (issue AST-191) ........
+
+2009-03-13 16:58 +0000 [r181987] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 181985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181985 |
+ kpfleming | 2009-03-13 11:55:38 -0500 (Fri, 13 Mar 2009) | 1 line
+ improve a bit of suboptimal code ........
+
+2009-03-12 21:45 +0000 [r181771-181849] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 181846 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 |
+ mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3
+ lines Run the macro on the queue member's channel when he
+ answers, not the caller's channel. ........
+
+ * /, channels/chan_sip.c: Merged revisions 181769 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar
+ 2009) | 28 lines Merged revisions 181768 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar
+ 2009) | 22 lines Properly send a 487 on an INVITE we have not
+ responded to if we receive a BYE. If we receive an INVITE from an
+ endpoint and then later receive a BYE from that same endpoint
+ before we have sent a final response for the INVITE, then we need
+ to respond to the INVITE with a 487. There was logic in the code
+ prior to this commit which seemed to exist solely to handle this
+ situation, but there was one condition in an if statement which
+ was incorrect. The only way we would send a 487 was if the
+ sip_pvt had no owner channel. This made no sense since we created
+ the owner channel when we received the INVITE, meaning that the
+ majority of the time we would never send the 487. The 487 being
+ sent should not rely on whether we have created a channel. Its
+ delivery should be dependent on the current state of the initial
+ INVITE transaction. With this commit, that logic is now correctly
+ in place. (closes issue #14149) Reported by: legranjl Patches:
+ 14149.patch uploaded by mmichelson (license 60) Tested by:
+ legranjl ........ ................
+
+2009-03-12 18:07 +0000 [r181733] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/translate.c: Merged revisions 181731 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181731 |
+ tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines
+ Adjust translation table column widths based upon the translation
+ times. Previously, only 5 columns were displayed, and if a
+ translation time exceeded 99,999 useconds, it would be displayed
+ as 0, instead of its actual time. (closes issue #14532) Reported
+ by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman
+ (license 14) Tested by: pj ........
+
+2009-03-12 16:58 +0000 [r181614-181667] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 181665 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu,
+ 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2
+ lines Fix incorrect usage of strncasecmp... I really meant to use
+ strcasecmp. ........ ................
+
+ * /, res/res_musiconhold.c: Merged revisions 181661 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu,
+ 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8
+ lines Fix another scenario where depending on configuration the
+ stream would not get read. For custom commands we don't know
+ whether the audio is coming from a stream or not so we are going
+ to have to read the data despite no channels. (closes issue
+ #14416) Reported by: caspy ........ r181660 | file | 2009-03-12
+ 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in
+ previous commit. ........ ................
+
+ * /, res/res_musiconhold.c: Merged revisions 181656 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu,
+ 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) |
+ 10 lines Fix issue with streaming MOH failing if nobody is
+ listening. When a music class is setup to actually provide music
+ on hold from a stream we need to constantly read audio from it
+ since it will constantly be providing audio. This is now done
+ despite there being no channels listening to it. (closes issue
+ #14416) Reported by: caspy ........ ................
+
+ * apps/app_dial.c, /: Merged revisions 181612 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 |
+ file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix
+ crash when sleep and retries argument was not given to RetryDial
+ application. (closes issue #14647) Reported by: sherpya ........
+
+2009-03-12 01:05 +0000 [r181544] Richard Mudgett <rmudgett@digium.com>
+
+ * /, build_tools/make_version: Merged revisions 181542 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009)
+ | 1 line Use the correct branch integrated property when
+ generating the version string ........
+
+2009-03-11 23:21 +0000 [r181521] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, configs/sip.conf.sample: Merged revisions 181499 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk Provide
+ correct hint to debug SIP trouble in the default config (closes
+ issue #14646) Reported by: strk
+
+2009-03-11 22:27 +0000 [r181474] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 181465 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181465 |
+ russell | 2009-03-11 17:25:57 -0500 (Wed, 11 Mar 2009) | 2 lines
+ Make handling of the BRIDGE_PLAY_SOUND variable thread-safe.
+ ........
+
+2009-03-11 22:23 +0000 [r181457] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 181444 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r181444 | qwell | 2009-03-11 17:20:13 -0500
+ (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) |
+ 4 lines Allow prefix to set localstatedir (when used and
+ different from the default). This is similar to the /etc change
+ that was made for the non-FreeBSD case. ........ ................
+
+2009-03-11 22:16 +0000 [r181426-181430] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 181428 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 |
+ russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines
+ Make handling of the BRIDGEPVTCALLID variable thread-safe.
+ ........
+
+ * main/channel.c, /: Merged revisions 181424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009)
+ | 17 lines Merged revisions 181423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009)
+ | 9 lines Make code that updates BRIDGEPEER variable thread-safe.
+ It is not safe to read the name field of an ast_channel without
+ the channel locked. This patch fixes some places in channel.c
+ where this was being done, and lead to crashes related to
+ masquerades. (closes issue #14623) Reported by: guillecabeza
+ ........ ................
+
+2009-03-11 17:40 +0000 [r181373] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, channels/iax2-parser.h, /: Merged revisions
+ 181371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009)
+ | 17 lines Merged revisions 181340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009)
+ | 11 lines encrypted IAX2 during packet loss causes decryption to
+ fail on retransmitted frames If an iax channel is encrypted, and
+ a retransmit frame is sent, that packet's iseqno is updated while
+ it is encrypted. This causes the entire frame to be corrupted.
+ When the corrupted frame is sent, the other side decrypts it and
+ sends a VNAK back because the decrypted frame doesn't make any
+ sense. When we get the VNAK, we look through the sent queue and
+ send the same corrupted frame causing a loop. To fix this,
+ encrypted frames requiring retransmission are decrypted, updated,
+ then re-encrypted. Since key-rotation may change the key held by
+ the pvt struct, the keys used for encryption/decryption are held
+ within the iax_frame to guarantee they remain correct. (closes
+ issue #14607) Reported by: stevenla Tested by: dvossel Review:
+ http://reviewboard.digium.com/r/192/ ........ ................
+
+2009-03-11 17:29 +0000 [r181298-181359] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 181345 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) |
+ 21 lines Merged revisions 181328 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) |
+ 14 lines Fix issue where an attended transfer could not be
+ completed under a rare scenario. When completing an attended
+ transfer chan_sip does a check to make sure the extension in the
+ URI portion of the Refer-To header is a local valid extension. We
+ don't actually need to check this since we know for sure the
+ other channel is already up and talking to the extension. Some
+ devices do not put the extension in the Refer-To header either,
+ which can cause the extension check to fail. We now no longer do
+ this check if it is an attended transfer. (closes issue #14628)
+ Reported by: sverre Patches: 14628.diff uploaded by file (license
+ 11) ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 181296 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) |
+ 16 lines Merged revisions 181295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9
+ lines Fix a problem with inband DTMF detection on outgoing SIP
+ calls when dtmfmode=auto. When dtmfmode was set to auto the
+ inband DTMF detector was not setup on outgoing SIP calls. This
+ caused inband DTMF detection to fail. The inband DTMF detector is
+ now setup for both dtmfmode inband and auto. (closes issue
+ #13713) Reported by: makoto ........ ................
+
+2009-03-11 15:54 +0000 [r181199-181283] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/ast_h323.cxx: add missing header file
+
+ * pbx/pbx_config.c, utils/Makefile, include/asterisk/utils.h,
+ include/asterisk/astmm.h, /, channels/chan_sip.c,
+ channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c:
+ Merged revisions 181135 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 |
+ jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines
+ Fix malloc debug macros to work properly with h323. The main
+ problem here was that cstdlib was undefining free thereby causing
+ the proper debug macros to not be used. ast_h323.cxx has been
+ changed to call ast_free instead to avoid the issue. A few other
+ issues were addressed: - There were a few instances of functions
+ improperly passing ast_free instead of ast_free_ptr. - Some clean
+ up was done to avoid the debug macros intentionally being
+ redefined. (copied below from Kevin's commit, appreciate the
+ help) - disable astmm.h from doing anything when STANDALONE is
+ defined, which is used by the tools in the utils/ directory that
+ use parts of Asterisk header files in hackish ways; also ensure
+ that utils/extconf.c and utils/conf2ael.c are compiled with
+ STANDALONE defined. (closes issue #13593) Reported by: pj
+ ........
+
+2009-03-11 01:04 +0000 [r181035] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 181032-181033 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500
+ (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar
+ 2009) | 9 lines Fix incorrect tag checking on transfers when
+ pedantic=yes is enabled. (closes issue #14611) Reported by:
+ klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt
+ uploaded by klaus3000 (license 65) Tested by: klaus3000 ........
+ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar
+ 2009) | 3 lines Remove unused variables. ........
+ ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500
+ (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC
+ 3891 ................
+
+2009-03-10 22:07 +0000 [r180947] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac, autoconf/ast_prog_sed.m4,
+ autoconf/ast_check_gnu_make.m4: Merged revisions 180944 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r180944 | qwell | 2009-03-10 17:03:41 -0500
+ (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar
+ 2009) | 1 line Make things happier when using autoconf 2.62+
+ ........ ................
+
+2009-03-10 14:42 +0000 [r180802] Joshua Colp <jcolp@digium.com>
+
+ * main/manager.c, /: Merged revisions 180800 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 |
+ file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines
+ Reset the thread local string buffer when handling the UserEvent
+ action. (closes issue #14593) Reported by: JimDickenson ........
+
+2009-03-09 21:22 +0000 [r180740] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/heap.h, include/asterisk/http.h,
+ include/asterisk/logger.h, main/tcptls.c,
+ include/asterisk/res_odbc.h, include/asterisk/doxyref.h,
+ include/asterisk/event.h, include/asterisk/audiohook.h,
+ include/asterisk/dsp.h, include/asterisk/lock.h,
+ include/asterisk/udptl.h, include/asterisk/dnsmgr.h,
+ include/asterisk/utils.h, include/asterisk/devicestate.h, /,
+ include/asterisk/taskprocessor.h, include/asterisk/astobj2.h,
+ include/asterisk/channel.h, include/asterisk/tcptls.h,
+ include/asterisk/manager.h, main/enum.c,
+ include/asterisk/callerid.h, include/asterisk/app.h,
+ include/asterisk/linkedlists.h, include/asterisk/sched.h,
+ include/asterisk/datastore.h, include/asterisk/timing.h,
+ include/asterisk/dlinkedlists.h, include/asterisk/pbx.h,
+ include/asterisk/enum.h, include/asterisk/config.h,
+ include/asterisk/rtp.h, include/asterisk/extconf.h,
+ main/devicestate.c: Merged revisions 180719 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180719 |
+ jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines
+ Add Doxygen documentation for API changes from 1.6.0 to 1.6.1
+ Copied from my review board description: This is a continuation
+ of the API changes documentation started for describing changes
+ between releases. Most of the API changes were pretty simple
+ needing only to be brought to attention via the new "Asterisk API
+ Changes" list. However, if you see anything that needs further
+ explanation feel free to supplement what is there. The current
+ method of documenting is to add (in the header file): \version
+ <ver number> <description of changes> and then to add the
+ function to the change list in doxyref.h on the AstAPIChanges
+ page. I also made sure all the functions that were newly added
+ were tagged with \since 1.6.1. I think this is a good habit to
+ start both for the historical aspect as well as for the future
+ ability to easily add a "New Asterisk API" page. Review:
+ http://reviewboard.digium.com/r/190/ ........
+
+2009-03-06 18:26 +0000 [r180585] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 180579 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600
+ (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri,
+ 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when
+ IMAP storage is enabled. ........ ................
+
+2009-03-06 17:35 +0000 [r180537] David Vossel <dvossel@digium.com>
+
+ * main/enum.c, /: Merged revisions 180534 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009)
+ | 15 lines Merged revisions 180532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009)
+ | 9 lines Fix handling of backreferences for ENUM lookups enum.c
+ did not handle regex backtraces correctly. The '\1' in the regex
+ is a backreference that requires a pattern match to be inserted.
+ The way the code used to work is that it would find the
+ backreference and insert the entire input string minus the '+'.
+ This is incorrect. The regexec() function takes in a variable
+ called pmatch which is an array of structs containing the start
+ and end indexes for each backreference substring. The original
+ code actually passed the pmatch array pointer into regexec but
+ never did anything with it. Now when a backtrace is found, the
+ backtrace number is looked up in the pmatch array and the correct
+ substring is inserted. (closes issue #14576) Reported by:
+ chris-mac Review: http://reviewboard.digium.com/r/187/ ........
+ ................
+
+2009-03-05 23:28 +0000 [r180425-180467] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600
+ (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar
+ 2009) | 16 lines [IMAP] Fix message retrieval issues when
+ identical mailbox names were defined in separate contexts. There
+ was a fix put in a while back so that an X-Asterisk-VM-Context
+ message header was added to stored IMAP voicemails. This would
+ allow for us to differentiate if the same mailbox name was used
+ in multiple contexts. The problem still left was that not all
+ places where messages were retrieved actually attempted to use
+ this header for information when retrieving messages. This commit
+ fixes that so that MWI and message retrieval from VoiceMailMain
+ work as expected. (closes issue #13853) Reported by: vicks1
+ Patches: 13853_v2.patch uploaded by mmichelson (license 60)
+ Tested by: lmadsen ........ ................
+
+ * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
+ revisions 180383 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar
+ 2009) | 31 lines Merged revisions 180380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar
+ 2009) | 25 lines Fix broken mailbox parsing when searchcontexts
+ option is enabled. When using the searchcontexts option in
+ voicemail.conf, the code made the assumption that all mailbox
+ names defined were unique across all contexts. However, the code
+ did nothing to actually enforce this assumption, nor did it do
+ anything to alert a user that he may have created an ambiguity in
+ his voicemail.conf file by defining the same mailbox name in
+ multiple contexts. With this change, we now will issue a nice
+ long warning if searchcontexts is on and we encounter the same
+ mailbox name in multiple contexts and ignore any duplicates after
+ the first box. Whether searchcontexts is enabled or not, if we
+ come across a duplicate mailbox in the same context, then we will
+ issue a warning and ignore the duplicated mailbox. I have also
+ added a small note to voicemail.conf.sample in the explanation
+ for searchcontexts explaining that you cannot define the same
+ mailbox in multiple contexts if you have enabled the option.
+ (closes issue #14599) Reported by: lmadsen Patches: 14599.patch
+ uploaded by mmichelson (license 60) (with slight modification)
+ Tested by: lmadsen ........ ................
+
+2009-03-05 18:40 +0000 [r180378] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, main/rtp.c, main/frame.c, /: Merged
+ revisions 180373 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar
+ 2009) | 15 lines Merged revisions 180372 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar
+ 2009) | 9 lines Fix problems when RTP packet frame size is
+ changed During some code analysis, I found that calling
+ ast_rtp_codec_setpref() on an ast_rtp session does not work as
+ expected; it does not adjust the smoother that may on the RTP
+ session, in fact it summarily drops it, even if it has data in
+ it, even if the current format's framing size has not changed.
+ This is not good. This patch changes this behavior, so that if
+ the packetization size for the current format changes, any
+ existing smoother is safely updated to use the new size, and if
+ no smoother was present, one is created. A new API call for
+ smoothers, ast_smoother_reconfigure(), was required to implement
+ these changes. Review: http://reviewboard.digium.com/r/184/
+ ........ ................
+
+2009-03-04 Leif Madsen <lmadsen@digium.com>
+
+ * Released Asterisk 1.6.1.0-rc2
+
+2009-03-04 21:09 +0000 [r180263] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 180261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180261 |
+ russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines
+ Resolve object matching issues related to the removal of the
+ sip_user object. Previously, chan_sip had both sip_peer and
+ sip_user objects in memory. A patch went in to remove sip_user to
+ simplify the code, since everything could be done with just
+ sip_peer. This patch resolves some regressions found that were
+ introduced by those changes. This code comes from
+ svn/asterisk/team/group/sip-object-matching/. Here is a list of
+ the changes that have been made: 1) When doing a match by name
+ with the find_peer() function, make it much easier to specify
+ which objects should be matched by having a parameter that
+ specifies exactly which object types should be considered. Also,
+ update find_by_name() to handle this parameter. Finally, update
+ all code to use the new option values. 2) When looking up an
+ object for an outbound request by name, consider peers only.
+ (create_addr()) 3) Only match peers on an incoming registration
+ request. 4) When doing authentication (except for SUBSCRIBE),
+ look up users by name, instead of all objects by name. 5) When
+ doing authentication (except for SUBSCRIBE), after looking for a
+ user by name, look for a peer by IP address, instead of all
+ objects by IP address. 6) When handling the SIP qualify CLI
+ command or manager action, look for a peer by name, instead of
+ any object by name. 7) When handling the SIP unregister CLI
+ command, look for a peer by name, instead of any object by name.
+ 9) In sip_do_debug_peer(), search for a peer by name, instead of
+ any object by name. 9) When handling the SIPPEER() dialplan
+ function, search for a peer by name, instead of any object by
+ name. 10) In the following session timer related functions,
+ st_get_se(), st_get_refresher(), and st_get_mode(), when looking
+ for an object for a given sip_pvt using pvt->peername, look for a
+ peer by name, instead of any object by name. 11) Fix build_peer()
+ to properly handle the case where separate type=peer and
+ type=user entries were specified in sip.conf. (closes issue
+ #14505) Reported by: lmadsen Review:
+ http://reviewboard.digium.com/r/172/ ........
+
+2009-03-04 19:27 +0000 [r180122-180197] Joshua Colp <jcolp@digium.com>
+
+ * /, main/callerid.c: Merged revisions 180195 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) |
+ 11 lines Merged revisions 180194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4
+ lines Look for the number in a callerid string starting from the
+ end. This way a value using <> can exist in the name portion.
+ (issue #AST-194) ........ ................
+
+ * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 |
+ file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines
+ Remove duplicate 'k' and 'K' Dial options. (closes issue #14601)
+ Reported by: alecdavis Patches: app_dial.optionk.diff.txt
+ uploaded by alecdavis (license 585) ........
+
+2009-03-03 23:39 +0000 [r180080] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, include/asterisk/app.h, apps/app_read.c, /,
+ main/app.c: Merged revisions 180032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 |
+ dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines
+ app_read does not break from prompt loop with user terminated
+ empty string In app.c, ast_app_getdata is called to stream the
+ prompts and receive DTMF input. If ast_app_getdata() receives an
+ empty string caused by the user inputing the end of string
+ character, in this case '#', it should break from the prompt loop
+ and return to app_read, but instead it cycles through all the
+ prompts. I've added a return value for this special case in
+ ast_readstring() which uses an enum I've delcared in apps.h. This
+ enum is now used as a return value for ast_app_getdata(). (closes
+ issue #14279) Reported by: Marquis Patches: fix_app_read.patch
+ uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded
+ by dvossel (license 671) Tested by: Marquis, dvossel Review:
+ http://reviewboard.digium.com/r/177/ ........
+
+2009-03-03 23:31 +0000 [r180077] Steve Murphy <murf@digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile,
+ utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y,
+ main/ast_expr2f.c: Merged revisions 179973 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) |
+ 33 lines Merged revisions 179807 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some
+ work to do to port these changes to trunk; the check_expr stuff
+ hasn't been updated here for quite some time, it appears. I added
+ some more tests to the check_expr2 suite. I had to play around
+ with the makefile a bit, etc. I added STANDALONE2 #ifdefs to
+ ast_expr2.y so as not to conflict structure with aelparse.
+ ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar
+ 2009) | 19 lines These changes allow AEL to better check ${}
+ constructs within $[...], that are concatenated with text. I
+ modified and added rules in ast_expr2.fl to better handle the
+ concatenations. I added some default routines to ast_expr2.y so
+ the standalone would compile. It also looks like I haven't run
+ this thru bison since 2.1, so it's good to get this updated. The
+ Makefile has comments added now for check_expr2 and check_expr to
+ explain what they are for, and how to run them. The testexpr2s
+ stuff has been removed, in favor of check_expr2. expr2.testinput
+ has been updated to include the two expressions that inspired
+ these changes (from mcnobody on #asterisk this morning) The
+ regression has been run and all looks well. ........
+ ................
+
+2009-03-03 22:49 +0000 [r179939-180009] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions
+ 180007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar
+ 2009) | 22 lines Merged revisions 180006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar
+ 2009) | 17 lines Clarify some documentation of queues.conf.sample
+ It had always been possible to explicitly specify a "blank" value
+ for a sound file in queues.conf and have no sound played back.
+ The problem with this is that it would result in some ugly CLI
+ warnings from file.c. This commit introduces a check when playing
+ a file in app_queue to see if the name of the file is zero-length
+ and return early if that is the case. Also, the ability to
+ specify the blank sound files in queues.conf is now mentioned
+ more clearly in queues.conf.sample (closes issue #14227) Reported
+ by: caspy ........ ................
+
+ * doc/timing.txt (added), /, res/res_timing_dahdi.c: Merged
+ revisions 179937 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179937 |
+ mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20
+ lines Add documentation for timing modules used in Asterisk This
+ document specifies the timing modules available in Asterisk
+ beginning with Asterisk 1.6.1. The document goes into detail
+ about the differences between each and gives a general overview
+ of what timing is used for in Asterisk. There is also a section
+ which can be used to help customize your setup or to troubleshoot
+ timing issues you may have. I also added messages to the DAHDI
+ timing test used in res_timing_dahdi.c that points to this new
+ documentation if people experience problems. Big thanks to all
+ who contributed comments on this. (closes issue #14490) Reported
+ by: mmichelson Patches: timing.txt uploaded by mmichelson
+ (license 60) Review: http://reviewboard.digium.com/r/164/
+ ........
+
+2009-03-03 20:09 +0000 [r179905] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 179903 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar
+ 2009) | 1 line fix a leaked channel lock (and future deadlock)
+ when we try to pick up our own channel ........
+
+2009-03-03 18:30 +0000 [r179843] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 179841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) |
+ 16 lines Merged revisions 179840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9
+ lines Do not assume that the bridge_cdr is still attached to the
+ channel when the 'h' exten is finished executing. It is possible
+ for a masquerade operation to occur when the 'h' exten is
+ operating. This operation moves the CDR records around causing
+ the bridge_cdr to no longer exist on the channel where it is
+ expected to. We can not safely modify it afterwards because of
+ this, so don't even try. (closes issue #14564) Reported by: meric
+ ........ ................
+
+2009-03-03 16:48 +0000 [r179744] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 179742 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009)
+ | 14 lines Merged revisions 179741 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009)
+ | 6 lines Ensure chan->fdno always gets reset to -1 after
+ handling a channel fd event. Since setting fdno to -1 had to be
+ moved, a couple of other code paths that do process an fd event
+ return early and do not pass through the code path where it was
+ moved to. So, set it to -1 in a few other places, too. ........
+ ................
+
+2009-03-03 14:41 +0000 [r179674] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 179672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) |
+ 10 lines Merged revisions 179671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3
+ lines Move where fdno is set to the default value to *after* the
+ read callback of the channel driver is called. We have to do this
+ as the underlying channel driver may need the fdno value to
+ determine what to read. ........ ................
+
+2009-03-03 13:56 +0000 [r179611] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 179609 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009)
+ | 17 lines Merged revisions 179608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009)
+ | 9 lines Make it easier to detect an improper call to
+ ast_read(). When you call ast_waitfor() on a channel, the index
+ into the channel fds array that holds the file descriptor that
+ poll() determines has input available is stored in fdno. This
+ patch clears out this value after a call to ast_read() and also
+ reports errors if ast_read() is called without an fdno set. From
+ a discussion on the asterisk-dev list. ........ ................
+
+2009-03-03 00:04 +0000 [r179539] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: Merged revisions 179537 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009)
+ | 21 lines Merged revisions 179536 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009)
+ | 15 lines Fix bridging regression from commit 176701 This fixes
+ a bad regression where the bridge would exit after an attended
+ transfer was made. The problem was due to nexteventts getting set
+ after the masquerade which caused the bridge to return
+ AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
+ tim_ringenbach ........ ................
+
+2009-03-02 23:39 +0000 [r179535] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009)
+ | 48 lines Merged revisions 179532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009)
+ | 40 lines Move ast_waitfor() down to avoid the results of the
+ API call becoming stale. This call to ast_waitfor() was being
+ done way too soon in this section of code. Specifically, there
+ was code in between the call to waitfor and the code that uses
+ the result that puts the channel in autoservice. By putting the
+ channel in autoservice, the previous results of ast_waitfor()
+ become meaningless, as the autoservice thread will do it's own
+ ast_waitfor() and ast_read() on the channel. So, when we came
+ back out of autoservice and eventually hit the block of code that
+ calls ast_read() on the channel, there may not actually be any
+ input on the channel available. Even though the previous call to
+ ast_waitfor() in app_meetme said there was input, the autoservice
+ thread has since serviced the channel for some period of time.
+ This bug manifested itself while dvossel was doing some testing
+ of MeetMe in Asterisk trunk. He was using the timerfd timing
+ module. When the code hit ast_read() erroneously, it determined
+ that it must have been called because of input on the timer fd,
+ as chan->fdno was set to AST_TIMING_FD, since that was the cause
+ of the last legitimate call to ast_read() done by autoservice. In
+ this test, an IAX2 channel was calling into the MeetMe
+ conference. It was _much_ more likely to be seen with an IAX2
+ channel because of the way audio is handled. Every audio frame
+ that comes in results in a call to ast_queue_frame(), which then
+ uses ast_timer_enable_continuous() to notify the channel thread
+ that a frame is waiting to be handled. So, the chances of
+ ast_waitfor() indicating that a channel needs servicing due to a
+ timer event on an IAX2 event is very high. Finally, it is
+ interesting to note that if a different timing interface was
+ being used, this bug would probably not be noticed. When
+ ast_read() is called and erroneously thinks that there is a timer
+ event to handle, it calls the ast_timer_ack() function. The
+ pthread and dahdi timing modules handle the ack() function being
+ called when there is no event by simply ignoring it. In the case
+ of the timerfd module, it results in a read() on the timer fd
+ that will block forever, as there is no data to read. This caused
+ Asterisk to lock up very quickly. Thanks to dvossel and
+ mmichelson for the fun debugging session. :-) ........
+ ................
+
+2009-03-02 23:12 +0000 [r179471] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/app.c: Merged revisions 179469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009)
+ | 17 lines Merged revisions 179468 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009)
+ | 10 lines When ending a recording with silence detection,
+ remember to reduce the duration. The end of the recording is
+ correspondingly trimmed, but the duration was not trimmed by the
+ number of seconds trimmed, so the saved duration was necessarily
+ longer than the actual soundfile duration. (closes issue #14406)
+ Reported by: sasargen Patches: 20090226__bug14406.diff.txt
+ uploaded by tilghman (license 14) Tested by: sasargen ........
+ ................
+
+2009-03-02 23:04 +0000 [r179464] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 179462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009)
+ | 16 lines Merged revisions 179461 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009)
+ | 8 lines Ensure that only one thread is calling ast_settimeout()
+ on a channel at a time. For example, with an IAX2 channel, you
+ can have both the channel thread and the chan_iax2 processing
+ threads calling this function, and doing so twice at the same
+ time is a bad thing. (Found in a debugging session with dvossel
+ and mmichelson) ........ ................
+
+2009-03-02 20:18 +0000 [r179407] Jason Parker <jparker@digium.com>
+
+ * /, main/editline/configure, main/editline/np/unvis.c,
+ main/editline/sys.h, main/editline/configure.in: Merged revisions
+ 179396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) |
+ 9 lines Merged revisions 179395 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) |
+ 1 line Remove several silly warnings in editline. One about a
+ broken preprocessor directive, and another about strlcpy/strlcat.
+ (closes issue #14264) Reported by: dimas ........
+ ................
+
+2009-03-02 17:19 +0000 [r179362] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179361 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r179361 | tilghman | 2009-03-02 11:18:48 -0600 (Mon, 02 Mar 2009)
+ | 2 lines Backport 1.6.0 fix to trunk (failsafe if db is not
+ loaded) ........
+
+2009-03-02 14:14 +0000 [r179293] Joshua Colp <jcolp@digium.com>
+
+ * /, main/audiohook.c: Merged revisions 179291 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179291 |
+ file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines Fix
+ issue where changing the volume of both directions of audio did
+ not work. (closes issue #14574) Reported by: KNK Patches:
+ audiohook_volume_fix.diff uploaded by KNK (license 545) ........
+
+2009-03-01 23:28 +0000 [r179221-179256] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_speech_utils.c, /: Merged revisions 179254 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar
+ 2009) | 5 lines Swap reversed timevals. This was pointed out by
+ ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! ........
+
+ * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 |
+ mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18
+ lines Properly free memory and remove scheduler entries when a
+ transmission failure occurs. Previously, only the "data" field of
+ the sip_pkt created during __sip_reliable_xmit was freed when
+ XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was
+ called, this inevitably resulted in the reading and writing of
+ freed memory. XMIT_ERROR is a condition meaning that we don't
+ want to attempt resending the packet at all. The proper action to
+ take is to remove the scheduler entry we just created, free the
+ packet's data as well as the packet itself, and unlink it from
+ the list of packets on the sip_pvt structure. (closes issue
+ #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by
+ mmichelson (license 60) Tested by: Nick_Lewis ........
+
+2009-02-27 21:48 +0000 [r179166] Russell Bryant <russell@digium.com>
+
+ * configs/ais.conf.sample, res/res_ais.c, /,
+ doc/distributed_devstate.txt: Merged revisions 179164 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27
+ Feb 2009) | 2 lines Mark res_ais as experimental, as the binary
+ event format is subject to change. ........
+
+2009-02-27 21:34 +0000 [r179163] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009)
+ | 3 lines If config file is blank, don't load module. (Closes
+ issue #14563) ........
+
+2009-02-27 21:25 +0000 [r179160] Russell Bryant <russell@digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 179154 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r179154 |
+ russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines
+ Add a note about the ordering of entries in sip.conf in 1.6.1.
+ ........
+
+2009-02-27 19:06 +0000 [r179059] Jason Parker <jparker@digium.com>
+
+ * /, doc/tex/channelvariables.tex: Merged revisions 179057 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb
+ 2009) | 8 lines Update documentation for DIALEDTIME and
+ ANSWEREDTIME variables. (closes issue #14566) Reported by:
+ klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
+ klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
+ klaus3000 (license 65) ........
+
+2009-02-27 03:56 +0000 [r178988] Steve Murphy <murf@digium.com>
+
+ * configs/features.conf.sample, /, main/features.c: Merged
+ revisions 178986 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) |
+ 26 lines Merged revisions 178956 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 In this
+ case, it's just a matter of reducing the default timeouts from
+ 2000 to 1000 msec, as the max def feature digit timeout is no
+ longer halved. ........ r178956 | murf | 2009-02-26 14:27:32
+ -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default
+ feature digit timeout to 1000 ms from the previous default of
+ 500. As per bug 14515, a dev discussion arrived at a "mediated
+ concensus" of a default feature digit timeout of 1.0 sec. Some
+ voted for 1300; ctooley thought 1500 for distracted phone users
+ in phone booths; kpfleming put his foot down at 1.0 sec. Users
+ who found the previous default max delay of 250 msec perfect, are
+ welcome to override the new default. Notice that I said that 250
+ msec was the default; wait a minute, you might say, the config
+ file said it was 500 msec!; well, because of the bug fix for
+ 14515, we found that 500 msec was actually enforcing a max of
+ 250. The bug fix would restore 500 msec, but we felt even that
+ was a bit tight for most users... 2000 msec was pushed earlier by
+ mmichelson, so that reduces to 1000 msec after the bug fix.
+ Enjoy! ........ ................
+
+2009-02-26 17:50 +0000 [r178875] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 178871 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009)
+ | 6 lines IAX2 prune realtime, minor tweak to last fix A return
+ statement was missing which caused unexpected cli output. issue
+ #14479 ........
+
+2009-02-26 17:38 +0000 [r178869] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 178828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) |
+ 34 lines Merged revisions 178804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) |
+ 28 lines This patch prevents the feature detection timeout from
+ being cut in half. Because the ast_channel_bridge() call will
+ return 0 and pass a frame pointer for both DTMF_BEGIN and
+ DTMF_END, the feature_timer field in hte config struct is getting
+ decremented twice, which effectively cuts the digittimeout in
+ half. I added conditions to the if statement to only let DTMF_END
+ frames to flow thru, which solved the problem. Also, when the
+ frame pointer is null, let control flow thru-- this usually
+ happens on timeouts. I added a comment to the code to explain
+ what's going on and why. Many thanks to sodom for reporting this
+ problem. Personnally, it always seemed like something was wrong
+ with the featuredigittimeout, but I never could quite decide
+ what... and was too busy to investigate. This bug forced the
+ issue, and now we know. Sodom had other issues in 14515, but I
+ couldn't reproduce them. If he still has problems, and wants to
+ get them solved, he is welcome to reopen 14515. (closes issue
+ #14515) Reported by: sodom Patches: 14515.patch uploaded by murf
+ (license 17) Tested by: murf, sodom ........ ................
+
+2009-02-26 16:44 +0000 [r178803] Joshua Colp <jcolp@digium.com>
+
+ * /, main/file.c: Merged revisions 178801 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178801 |
+ file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix
+ an issue where the timer for file playback would not be stopped
+ if DAHDI was not installed. (closes issue #14541) Reported by:
+ grant ........
+
+2009-02-26 16:07 +0000 [r178769] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 178767 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009)
+ | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues.
+ If "iax2 prune realtime all" was called, it would appear like the
+ command was successful, but in reality nothing happened. This is
+ because the reload that was supposed to take place checks the
+ config files, sees no changes, and does nothing. If there had
+ been a change in the the config file, the realtime users would
+ have been marked for deletion and everything would have been
+ fine. Now prune_users() and prune_peers() are called instead of
+ reload_config() to prune all users/peers that are realtime. These
+ functions remove all users/peers with the rtfriend and delme
+ flags set. iax2_prune_realtime() also lacked the code to properly
+ delete a single friend. For example. if iax2 prune realtime
+ <friend> was called, only the peer instance would be removed. The
+ user would still remain. (closes issue #14479) Reported by:
+ mousepad99 Review: http://reviewboard.digium.com/r/176/ ........
+
+2009-02-25 12:46 +0000 [r178511] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 178509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009)
+ | 10 lines Merged revisions 178508 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009)
+ | 2 lines Update the copyright year for the main page of the
+ doxygen documentation. ........ ................
+
+2009-02-24 23:28 +0000 [r178383-178448] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 178446 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600
+ (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009)
+ | 5 lines Add section about the #exec command in configuration
+ files. (closes issue #14540) Reported by: jtodd Patch by: jtodd,
+ with additional notes by tilghman (license 14) ........
+ ................
+
+ * main/asterisk.c, /: Merged revisions 178381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 |
+ tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines
+ Apparently, a void cast doesn't override warn_unused_result.
+ ........
+
+2009-02-24 20:44 +0000 [r178379-178380] Russell Bryant <russell@digium.com>
+
+ * Makefile: revert accidental Makefile change.
+
+ * main/rtp.c, Makefile, /: Merged revisions 178374 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r178374 | russell | 2009-02-24 14:39:57 -0600
+ (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009)
+ | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset
+ to 0 properly. (issue #14460) Reported by: moliveras Tested by:
+ russell ........ ................
+
+2009-02-24 20:41 +0000 [r178305-178377] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 178375 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 |
+ tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines
+ The 3 possible errors with pipe(2) are all impossible in this
+ situation. ........
+
+ * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24
+ Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of
+ depending upon the astcanary process being inherited by init.
+ ........
+
+ * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 |
+ tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines
+ Cause astcanary to exit if Asterisk exits abnormally and doesn't
+ kill astcanary. Also, add some documentation supporting the use
+ of astcanary. (closes issue #14538) Reported by: KNK Patches:
+ asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545)
+ ........
+
+2009-02-24 15:22 +0000 [r178232] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) |
+ 16 lines Merged revisions 178205 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9
+ lines Skip check for extension when subscribing for MWI. Since
+ the remote side is not actually subscribing to a specific
+ extension when subscribing for MWI just skip the check to see if
+ the extension exists. They can't use it to specify the mailbox
+ either since we require configuration of that in sip.conf (closes
+ issue #14531) Reported by: festr ........ ................
+
+2009-02-23 23:22 +0000 [r178172] Russell Bryant <russell@digium.com>
+
+ * main/rtp.c, /: Merged revisions 178142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009)
+ | 22 lines Merged revisions 178141 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009)
+ | 14 lines Fix infinite DTMF when a BEGIN is received without an
+ END. This commit is related to rev 175124 of 1.4 where a previous
+ attempt was made to fix this problem. The problem with the
+ previous patch was that the inserted code needed to go _before_
+ setting the lastrxts to the current timestamp. Because those were
+ the same, the dtmfcount variable was never decremented, and so
+ the END was never sent. In passing, I removed the dtmfsamples
+ variable which was completed unused. I also removed a redundant
+ setting of the lastrxts variable. (closes issue #14460) Reported
+ by: moliveras ........ ................
+
+2009-02-21 16:04 +0000 [r177945] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 177944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177944 |
+ tilghman | 2009-02-21 09:59:49 -0600 (Sat, 21 Feb 2009) | 2 lines
+ On update, test against the existence of sipregs. ........
+
+2009-02-21 12:51 +0000 [r177851] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_sip.c: Merged revisions 177849 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177849 |
+ mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines
+ make chan_sip.c compile on OpenBSD again. ........
+
+2009-02-20 23:05 +0000 [r177789] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 177787 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009)
+ | 16 lines Merged revisions 177786 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009)
+ | 9 lines Don't print the CR-NL combination when we aren't
+ outputting to the manager. An embedded CR-NL in a CLI command
+ screws up several AMI parsers that don't expect to see that
+ combination in the middle of output. (Closes issue #14305)
+ Reported by: martins Patch by: tilghman ........ ................
+
+2009-02-20 22:27 +0000 [r177785] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 177699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177699 |
+ dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines
+ Make app_fax compatible with spandsp-0.0.6pre4 Prior to
+ spandsp-0.0.6pre4 the t30_stats_t structure used a
+ pages_transferred integer to indicate the number of pages
+ transferred (so far) during the fax session. The
+ spandsp-0.0.6pre4 release removed the pages_transferred integer
+ and replaced it with two different integers - pages_tx and
+ pages_rx. This revision uses the new integers for
+ spandsp-0.0.6pre4 while maintaining backwards compatibility for
+ previous spandsp releases. ........
+
+2009-02-20 22:15 +0000 [r177760-177764] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/strings.h: Oops, last merge broke 1.6.1 branch
+
+ * apps/app_system.c, include/asterisk/app.h, /, main/app.c: Merged
+ revisions 177664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 |
+ tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines
+ Allow semicolons to be escaped, when passing arguments to the
+ System command. (closes issue #14231) Reported by: jcovert
+ Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76
+ (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by
+ jcovert (license 551) Tested by: jcovert ........
+
+ * include/asterisk/threadstorage.h, /: Merged revisions 177732 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177732 | tilghman | 2009-02-20 15:25:37 -0600
+ (Fri, 20 Feb 2009) | 10 lines Merged revisions 177701 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009)
+ | 3 lines This exception does not appear to still be true for
+ Solaris 10, and OpenSolaris definitely needs it to be removed.
+ Fixed for snuff-home on -dev channel. ........ ................
+
+2009-02-20 20:34 +0000 [r177700] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with
+ undefined audio codecs in chan_iax2 During iax2 call negotiation,
+ supported codecs are passed in an Information Element containing
+ a 2 byte field where each bit correlates to a specific codec. In
+ 1.6 only audio codec bits 0-12 are defined, leaving bits 13-14
+ undefined. By default all bits are enabled unless specified
+ otherwise. Since its a 2 byte field and 13-14 are not defined,
+ these bits are never turned off. In trunk, bits 13-14 are
+ defined, which means 1.6 is advertising support for codecs it
+ does not have when talking to trunk. I fixed this by adding
+ #define for undefined audio codec bits. These bits are then
+ removed from iax2's full bandwidth capabilities. (closes issue
+ #14283) Reported by: jcovert
+
+2009-02-20 17:28 +0000 [r177663] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 177661 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r177661 | tilghman | 2009-02-20 11:22:19 -0600 (Fri, 20 Feb 2009)
+ | 2 lines Oops, merge broke trunk ........
+
+2009-02-20 00:38 +0000 [r177626] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 177624 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177624 |
+ jpeeler | 2009-02-19 18:35:53 -0600 (Thu, 19 Feb 2009) | 7 lines
+ Set sip_request ast_str data to NULL so ast_str_copy allocates
+ space properly in copy_request (issue #14478) Reported by:
+ erik_dedecker ........
+
+2009-02-20 00:26 +0000 [r177623] Steve Murphy <murf@digium.com>
+
+ * /, main/Makefile, main/ast_expr2f.c: Merged revisions 177595 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177595 | murf | 2009-02-19 16:56:50 -0700 (Thu,
+ 19 Feb 2009) | 32 lines Merged revisions 177540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was
+ already pretty 8-bit clean; but I'm still removing the --full
+ from the flex command so everything is uniform. ........ r177540
+ | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
+ This patch fixes a problem with 8-bit input to the ast_expr2
+ scanner. The real culprit was the --full argument to flex in the
+ Makefile! This causes a 7-bit scanner to be generated. I reviewed
+ the rules and found one rule where I needed to specifically
+ include 8-bit chars for a token. I tested against the text
+ supplied by ibercom, and all looks very well. This has been there
+ a surprisingly long time! (closes issue #14498) Reported by:
+ ibercom Patches: 14498.patch uploaded by murf (license 17) Tested
+ by: murf ........ ................
+
+2009-02-19 22:35 +0000 [r177539] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 177537 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177537 | tilghman | 2009-02-19 16:33:00 -0600
+ (Thu, 19 Feb 2009) | 14 lines Merged revisions 177536 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009)
+ | 7 lines Fix up potential crashes, by reducing the sharing
+ between interactive and non-interactive threads. (closes issue
+ #14253) Reported by: Skavin Patches: 20090219__bug14253.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: Skavin ........
+ ................
+
+2009-02-19 16:46 +0000 [r177389] Jeff Peeler <jpeeler@digium.com>
+
+ * /, include/asterisk/channel.h: Merged revisions 177387 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r177387 | jpeeler | 2009-02-19 10:45:02 -0600 (Thu, 19
+ Feb 2009) | 3 lines Fix another merge error from 176708 ........
+
+2009-02-19 16:40 +0000 [r177386] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_speech_utils.c, /: Merged revisions 177384 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177384 | file | 2009-02-19 12:38:41 -0400 (Thu,
+ 19 Feb 2009) | 10 lines Merged revisions 177383 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3
+ lines If we are able to create a speech structure unset the ERROR
+ variable in case it was previously set. (issue #LUMENVOX-13)
+ ........ ................
+
+2009-02-19 15:57 +0000 [r177358] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 177356 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177356 |
+ jpeeler | 2009-02-19 09:56:31 -0600 (Thu, 19 Feb 2009) | 4 lines
+ Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev
+ on the asterisk-dev mailing list. Thanks! ........
+
+2009-02-19 00:17 +0000 [r177294] Steve Murphy <murf@digium.com>
+
+ * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177286 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177286 | murf | 2009-02-18 16:50:57 -0700 (Wed,
+ 18 Feb 2009) | 39 lines Merged revisions 177225 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) |
+ 34 lines This patch fixes a regression of sorts that was
+ introduced in rev 24425. It basically fixes AST-190/ABE-1782.
+ What was wrong: the user has 6000 extensions in one context; and
+ then 6000 contexts, one per extension. The parser could only
+ handle about 4893 of the 6000 extens in the single context. This
+ was due to the regression I mentioned. To get rid of shift/reduce
+ conflicts, Luigi set up right-recursive lists for globals,
+ context elements, switch lists, and statements. Right recursive
+ lists got rid of the warnings, but instead, they use up a
+ tremendous amount of stack space when the lists are long. I saw
+ this a few years back, and resolved not to fix it until someone
+ complained. That day has arrived! After the changes were made, I
+ ran the regression test suite, and there were no problems. I took
+ the test case the user provided, and added 100,000 extensions to
+ the single context, that already had 6,000 extens in it. (I'll
+ see your 6, and raise you 100!) It takes a few minutes to read it
+ all in, check it and generate code for it, but no problems. So, I
+ think I can say that fundamentally, there are no longer any
+ limits on the number of items you can place in contexts,
+ statement blocks, switches, or globals, beyond your virt mem
+ constraints. ........ ................
+
+2009-02-18 23:15 +0000 [r177230] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c, /: Merged revisions 177229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177229 |
+ kpfleming | 2009-02-18 17:09:58 -0600 (Wed, 18 Feb 2009) | 3
+ lines fix two very minor bugs: if anyone ever uses SLINEAR16 as a
+ format in RTP, ensure that the samples are byte-swapped to
+ network order if needed. also, when a smoother is operating on a
+ format that has a sample rate other than 8000 samples per second,
+ use the proper sample rate for computing delivery timestamps.
+ ........
+
+2009-02-18 23:03 +0000 [r177228] David Vossel <dvossel@digium.com>
+
+ * /, main/features.c: Merged revisions 177226 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177226 |
+ dvossel | 2009-02-18 16:51:38 -0600 (Wed, 18 Feb 2009) | 9 lines
+ Locking issue in action_bridge and bridge_exec action_bridge()
+ and bridge_exec() both search for the channels to bridge to, and
+ then immediately drop the lock. Instead, they should hold the
+ lock until the masquerade is complete. This will guarantee the
+ channel remains and prevent any other weirdness from occurring.
+ In action_bridge() some more weirdness comes into play. Both
+ channels are needlessly locked at the same time and perform the
+ exact same logic. It makes sense from a coding organizational
+ standpoint, but could cause a theoretical deadlock so I split the
+ code up. There is an issue associated with this, but since its a
+ rather complicated thing to reproduce I'm not certain this alone
+ will close it. issue# 14296 Review:
+ http://reviewboard.digium.com/r/167/ ........
+
+2009-02-18 20:16 +0000 [r177164] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/chan_h323.h, channels/h323/cisco-h225.cxx,
+ channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4,
+ channels/h323/cisco-h225.h, /, channels/h323/caps_h323.cxx,
+ channels/h323/ast_ptlib.h (added), channels/h323/ast_h323.cxx,
+ configure, channels/h323/compat_h323.h, configure.ac,
+ channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
+ channels/h323/ast_h323.h: Merged revisions 177162 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r177162 | jpeeler | 2009-02-18 14:11:57 -0600 (Wed, 18 Feb 2009)
+ | 14 lines Modify h323 to build against PTLib as well as the
+ older PWLib Several changes in PTLib have occurred requiring
+ build time detection. Changes accounted for include the library
+ name change, config option change, install location change, and a
+ boolean type change which is handled by ast_ptlib.h. Also, the
+ sed check has been modified to properly work with autoconf >=
+ 2.62. (closes issue #14224) Reported by: bergolth Patches:
+ asterisk-autoconf-sed.patch uploaded by bergolth (license 661)
+ asterisk-pwlib-v3.patch uploaded by bergolth (license 661) Tested
+ by: jpeeler ........
+
+2009-02-18 19:30 +0000 [r177158] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 177101 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177101 |
+ russell | 2009-02-18 13:12:49 -0600 (Wed, 18 Feb 2009) | 8 lines
+ Re-add 'o' option to MeetMe, reverting rev 62297. Enabling this
+ option by default proved to be a bad idea, as the talker
+ detection is not very reliable. So, make it optional again, and
+ off by default. (issue #13801) Reported by: justdave ........
+
+2009-02-18 19:09 +0000 [r177100] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/config.h: Merged revisions 177098 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r177098 | tilghman | 2009-02-18 13:05:15 -0600
+ (Wed, 18 Feb 2009) | 9 lines Merged revisions 177096 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18
+ Feb 2009) | 2 lines Document the return value of the update
+ method (as requested on -dev list) ........ ................
+
+2009-02-18 17:26 +0000 [r177037] Doug Bailey <dbailey@digium.com>
+
+ * /, main/utils.c: Merged revisions 177035 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 |
+ dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines
+ Fixed error where a check for an zero length, terminated string
+ was needed. ........
+
+2009-02-18 17:14 +0000 [r177007] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 177005 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177005 |
+ file | 2009-02-18 13:11:52 -0400 (Wed, 18 Feb 2009) | 6 lines Fix
+ ordering of output for a ChannelUpdate manager event. (closes
+ issue #14497) Reported by: vinsik Patches:
+ chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)
+ ........
+
+2009-02-18 16:20 +0000 [r176962] Doug Bailey <dbailey@digium.com>
+
+ * /, main/utils.c: Merged revisions 176948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176948 |
+ dbailey | 2009-02-18 10:09:12 -0600 (Wed, 18 Feb 2009) | 2 lines
+ Need to take into account the \0 terminator of the old string to
+ determine the amount available. ........
+
+2009-02-18 15:59 +0000 [r176946] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /: Merged revisions 176943 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176943 |
+ murf | 2009-02-18 08:35:26 -0700 (Wed, 18 Feb 2009) | 45 lines
+ This patch fixes merge_contexts_and_delete so it does not
+ deadlock when hints are present. Reason: when I re-engineered the
+ merge_and_delete func to reduce its lock time, I failed to notice
+ that the functions it calls still also do locking as before. This
+ leads to deadlocks on dialplan reloads, when there are actually
+ living, subscribed hints registered in the system. While the
+ reporter come across this problem while using AEL, I might note
+ that these deadlocks should also happen if extensions.conf were
+ used. Here I added these routines to pbx.c:
+ ast_add_extension_nolock add_pri_lockopt
+ ast_add_extension2_lockopt find_context add_hint_nolock All of
+ the above routines are static and restricted to be used only
+ within pbx.c, and more specifically within the
+ merge_contexts_and_delete routine. They are pretty much the same
+ as their counterparts except they don't lock contexts or hints.
+ Most of them now do the real work of their name-alike, with
+ optional locking via extra arguments, and are called by their
+ name-alike. The goal was to have the original functions so they
+ would behave exactly as before. Both PJ and I tested these fixes,
+ and the deadlocking problem is no longer encountered. (closes
+ issue #14357) Reported by: pj Patches: 14357.diff uploaded by
+ murf (license 17) Tested by: pj, murf ........
+
+2009-02-18 06:15 +0000 [r176903-176906] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/heap.h, /: Merged revisions 176904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r176904 | russell | 2009-02-18 00:14:47 -0600 (Wed, 18 Feb 2009)
+ | 2 lines Add example code for a heap traversal. ........
+
+ * main/pbx.c, /: Merged revisions 176901 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176901 |
+ russell | 2009-02-18 00:00:40 -0600 (Wed, 18 Feb 2009) | 9 lines
+ Fix a number of incorrect uses of strncpy(). The big problem here
+ is that the 3rd argument provided in these uses of strncpy() did
+ not reserve a byte for the null terminator, leaving the potential
+ for writing one byte past the end of the buffer. Aside from this,
+ there were coding guidelines violations with regards to spacing,
+ as well as hard coded lengths being used instead of sizeof().
+ ........
+
+2009-02-18 00:23 +0000 [r176809] Shaun Ruffell <sruffell@digium.com>
+
+ * /, codecs/codec_dahdi.c: Merged revisions 176760 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r176760 | sruffell | 2009-02-17 16:28:41 -0600 (Tue, 17 Feb 2009)
+ | 10 lines Several changes to codec_dahdi to play nice with G723.
+ This commit brings in the changes that were living out on the
+ svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch.
+ codec_dahdi.c now always uses signed linear as the simple codec
+ so that a soft g729 codec will not end up being preferred to the
+ hardware codec. There are also changes to allow codec_dahdi.c to
+ feed packets to the hardware in the native sample size of the
+ codec. This solves problems with choppy audio when using G723.
+ ........
+
+2009-02-17 22:21 +0000 [r176731] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 176705 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176705 |
+ dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11
+ lines create a UDPTL structure in create_addr_from_peer() if it
+ does not already exist for T38 This is required to create a UDPTL
+ structure in create_addr_from_peer() to handle the scenario where
+ 't38pt_udptl=yes' is not defined in the [general] section of
+ sip.conf but is defined the peer's context. I tested this patch
+ by enabling t38pt_udptl in the [general] section on one system
+ and only enabling t38pt_udptl in a peer's context on the system
+ sending a fax. Without the patch, the sending system will fail to
+ initiate T38 negotiation with the warning message, "No way to add
+ SDP without an UDPTL structure". When this patch is applied the
+ sending side will successfully initiate T38 negotiation. ........
+
+2009-02-17 22:15 +0000 [r176711] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ Merged revisions 176708 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009)
+ | 23 lines Merged revisions 176701 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009)
+ | 17 lines Modify bridging to properly evaluate DTMF after first
+ warning is played The main problem is currently if the Dial flag
+ L is used with a warning sound, DTMF is not evaluated after the
+ first warning sound. To fix this, a flag has been added in
+ ast_generic_bridge for playing the warning which ensures that if
+ a scheduled warning is missed, multiple warrnings are not played
+ back (due to a feature evaluation or waiting for digits).
+ ast_channel_bridge was modified to store the nexteventts in the
+ ast_bridge_config structure as that information was lost every
+ time ast_channel_bridge was reentered, causing a hangup due to
+ incorrect time calculations. (closes issue #14315) Reported by:
+ tim_ringenbach Reviewed on reviewboard:
+ http://reviewboard.digium.com/r/163/ ........ ................
+
+2009-02-17 21:41 +0000 [r176699] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/frame.h, /: Merged revisions 176697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r176697 | mmichelson | 2009-02-17 15:40:09 -0600 (Tue, 17 Feb
+ 2009) | 3 lines Clear up documentation of AST_FRIENDLY_OFFSET in
+ frame.h ........
+
+2009-02-17 21:24 +0000 [r176675] Russell Bryant <russell@digium.com>
+
+ * main/timing.c, main/channel.c, /, res/res_timing_pthread.c,
+ res/res_timing_dahdi.c, include/asterisk/timing.h: Merged
+ revisions 176666 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176666 |
+ russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines
+ Update the timing API to have better support for multiple timing
+ interfaces. 1) Add module use count handling so that timing
+ modules can be unloaded. 2) Implement unload_module() functions
+ for the timing interface modules. 3) Allow multiple timing
+ modules to be loaded, and use the one with the highest priority
+ value. 4) Report which timing module is being use in the "timing
+ test" CLI command. (closes issue #14489) Reported by: russell
+ Review: http://reviewboard.digium.com/r/162/ ........
+
+2009-02-17 21:16 +0000 [r176644] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c, channels/chan_local.c, /: Merged revisions
+ 176592,176642 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176592 |
+ tilghman | 2009-02-17 12:49:20 -0600 (Tue, 17 Feb 2009) | 4 lines
+ Add assertions in the quest to track down a refcount leak.
+ (closes issue #14485) Reported by: davevg ........ r176642 |
+ tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines
+ Prior to masquerade, move the group definitions to the channel
+ performing the masq, so that the group count lingers past the
+ bridge. (closes issue #14275) Reported by: kowalma Patches:
+ 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: kowalma ........
+
+2009-02-17 20:57 +0000 [r176559-176637] Russell Bryant <russell@digium.com>
+
+ * tests/test_heap.c (added), /: Merged revisions 176635 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r176635 | russell | 2009-02-17 14:56:26 -0600 (Tue, 17
+ Feb 2009) | 4 lines Add a test module for the heap
+ implementation. Review: http://reviewboard.digium.com/r/160/
+ ........
+
+ * include/asterisk/heap.h (added), /, main/Makefile, main/heap.c
+ (added): Merged revisions 176632 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176632 |
+ russell | 2009-02-17 14:51:10 -0600 (Tue, 17 Feb 2009) | 8 lines
+ Add an implementation of the heap data structure. A heap is a
+ convenient data structure for implementing a priority queue. Code
+ from svn/asterisk/team/russell/heap/. Review:
+ http://reviewboard.digium.com/r/160/ ........
+
+ * apps/app_queue.c, main/pbx.c, /: Merged revisions 176557 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r176557 | russell | 2009-02-17 11:33:38 -0600 (Tue, 17
+ Feb 2009) | 12 lines Fix a race condition that caused device
+ states to become incorrect for hints. The problem here is that
+ the hint processing code was subscribed to the wrong event type.
+ So, it started processing state for a hint too soon, before the
+ device state cache had been updated. Also, fix a similar bug in
+ app_queue, as it was also subscribed to the wrong event type.
+ (closes issue #14461) Reported by: alecdavis ........
+
+2009-02-17 14:48 +0000 [r176461-176503] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 176501 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176501 |
+ tilghman | 2009-02-17 08:39:36 -0600 (Tue, 17 Feb 2009) | 3 lines
+ In this version, we can combine the queries, because we support
+ dropping nonexistent columns. ........
+
+ * /, channels/chan_sip.c: Merged revisions 176459 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009)
+ | 17 lines Merged revisions 176426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009)
+ | 10 lines After a 'sip reload', qualifies for realtime peers
+ weren't immediately restarted, instead waiting until the next
+ registration. We're now caching the qualify across a
+ reload/restart and starting the qualify immediately upon loading
+ the peer. (closes issue #14196) Reported by: pdf Patches:
+ 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
+ Tested by: pdf ........ ................
+
+2009-02-16 23:57 +0000 [r176362] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 176355 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r176355 | dvossel | 2009-02-16 17:33:55 -0600
+ (Mon, 16 Feb 2009) | 13 lines Merged revisions 176354 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009)
+ | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not being
+ relayed correctly during bridging This should have been committed
+ with rev176247, but I missed it. srcupdate frames no longer break
+ out of the native bridge, but are not being sent to the other
+ call leg either. This fixs that. issue #13749 ........
+ ................
+
+2009-02-16 23:17 +0000 [r176321] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_skinny.c: Merged revisions 176320 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r176320 | tilghman | 2009-02-16 17:14:08 -0600 (Mon, 16 Feb 2009)
+ | 7 lines Use the correct list macros for deleting an item from
+ the middle of a list. (issue #13777) Reported by: pj Patches:
+ 20090203__bug13777.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: pj ........
+
+2009-02-16 22:00 +0000 [r176259] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/stringfields.h, /, main/utils.c: Merged
+ revisions 176255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb
+ 2009) | 13 lines Merged revisions 176216 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb
+ 2009) | 3 lines fix a flaw in the ast_string_field_build() family
+ of API calls; these functions made no attempt to reuse the space
+ already allocated to a field, so every time the field was written
+ it would allocate new space, leading to what appeared to be a
+ memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46
+ -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the
+ last stringfields commit... don't mark additional space as
+ allocated if the string was built using already-allocated space
+ ........ ................
+
+2009-02-16 21:50 +0000 [r176257] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 176253 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r176253 | mmichelson | 2009-02-16 15:40:40 -0600 (Mon, 16 Feb
+ 2009) | 24 lines Merged revisions 176249,176252 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb
+ 2009) | 14 lines Open the DAHDI pseudo device and set it to be
+ nonblocking atomically Apparently on FreeBSD, attempting to set
+ the O_NONBLOCKING flag separately from opening the file was
+ causing an "inappropriate ioctl for device" error. While I cannot
+ fathom why this would be happening, I certainly am not opposed to
+ making the code a bit more compact/efficient if it also fixes a
+ bug. (closes issue #14482) Reported by: ys Patches: meetme.patch
+ uploaded by ys (license 281) Tested by: ys ........ r176252 |
+ mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3
+ lines Remove unused variable and make dev-mode compilation happy
+ ........ ................
+
+2009-02-16 21:36 +0000 [r176251] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 176248 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r176248 | dvossel | 2009-02-16 15:30:17 -0600
+ (Mon, 16 Feb 2009) | 11 lines Merged revisions 175597 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13
+ Feb 2009) | 4 lines Fixed iax2 key rotation backwards
+ compatibility Turns key rotation back on by default. Added bit
+ into encryption IE to indicate whether or not key rotation is
+ supported or not. If it is not supported then it is not enabled,
+ which insures backwards compatibility. This eliminates the need
+ for the keyrotate option in iax.conf, so it has been removed.
+ ........ ................
+
+2009-02-16 18:38 +0000 [r176176] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/logger.c: Merged revisions 176174 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r176174 |
+ mmichelson | 2009-02-16 12:25:57 -0600 (Mon, 16 Feb 2009) | 11
+ lines Assist proper thread synchronization when stopping the
+ logger thread. I was finding that on my dev box, occasionally
+ attempting to "stop now" in trunk would cause Asterisk to hang. I
+ traced this to the fact that the logger thread was waiting on a
+ condition which had already been signalled. The logger thread
+ also need to be sure to check the value of the
+ close_logger_thread variable. The close_logger_thread variable is
+ only checked when the list of logmessages is empty. This allows
+ for the logger thread to print and free any pending messages
+ before exiting. ........
+
+2009-02-16 17:10 +0000 [r176102] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_features.c (removed): Merged revisions 176100
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r176100 | russell | 2009-02-16 11:09:24 -0600 (Mon, 16
+ Feb 2009) | 4 lines Remove chan_features. Review:
+ http://reviewboard.digium.com/r/161/ ........
+
+2009-02-16 17:07 +0000 [r176099] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/func_odbc.conf.sample: Eliminate mention of a variable
+ which exists only in trunk. (Thanks, jsmith)
+
+2009-02-16 15:38 +0000 [r176032] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 176030 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) |
+ 16 lines Merged revisions 176029 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9
+ lines Don't have the Via header stored as a stringfield as it can
+ change often during the lifetime of a dialog. This issue crept up
+ with subscriptions on the AA50. When an outgoing NOTIFY is sent a
+ new branch value is created and the Via header is changed to
+ reflect it. Since this was a stringfield a new spot in the pool
+ was used for the value while the old was left untouched/unused.
+ If the current pool was full a new pool was created. This would
+ cause memory usage to increase steadily. (issue #AA50-2332)
+ ........ ................
+
+2009-02-16 09:42 +0000 [r176023] Michiel van Baak <michiel@vanbaak.info>
+
+ * include/asterisk/manager.h, doc/unistim.txt,
+ channels/chan_unistim.c, /, channels/chan_sip.c: Merged revisions
+ 175952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009)
+ | 10 lines Merged revisions 175921 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009)
+ | 3 lines fix mis-spelling of the word registered. Reported by
+ De_Mon on #asterisk-dev. ........ ................
+
+2009-02-15 21:28 +0000 [r175831-175890] Russell Bryant <russell@digium.com>
+
+ * main/sched.c, /, include/asterisk/sched.h: Merged revisions
+ 175882 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175882 |
+ russell | 2009-02-15 15:27:33 -0600 (Sun, 15 Feb 2009) | 2 lines
+ Make ast_sched_report() and ast_sched_dump() thread safe.
+ ........
+
+ * main/sched.c, /, channels/chan_sip.c, include/asterisk/sched.h:
+ Merged revisions 175829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175829 |
+ russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines
+ Fix a number of problems with ast_sched_report(). 1) It had
+ numerous coding guidelines violations with regards to formatting.
+ 2) It allocated memory using ast_calloc() that was never freed.
+ 3) It didn't check for failure from the allocation. 4) It used
+ sprintf() and strcat() to build the result, doing zero checking
+ to prevent writing past the end of the provided buffer. The
+ function also lacks API documentation, but that has not been
+ addressed in this commit. ........
+
+2009-02-13 20:48 +0000 [r175662] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, configs/iax.conf.sample, channels/iax2.h:
+ Merged revisions 175597 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 |
+ dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
+ Fixed iax2 key rotation backwards compatibility Turns key
+ rotation back on by default. Added bit into encryption IE to
+ indicate whether or not key rotation is supported or not. If it
+ is not supported then it is not enabled, which insures backwards
+ compatibility. This eliminates the need for the keyrotate option
+ in iax.conf, so it has been removed. ........
+
+2009-02-13 19:52 +0000 [r175593] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 175591 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r175591 | mmichelson | 2009-02-13 13:49:38 -0600
+ (Fri, 13 Feb 2009) | 22 lines Merged revisions 175590 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb
+ 2009) | 16 lines Fix a potential crash situation when using IMAP
+ voicemail If calling into VoiceMailMain when using IMAP storage,
+ it was possible to crash Asterisk by hanging up the phone when
+ prompted for a voicemail mailbox. This patch fixes the issue.
+ While it may appear that this patch is superficial, it allows
+ code execution to continue to the failure case just below the
+ IMAP_STORAGE code block where this patch has been applied (closes
+ issue #14473) Reported by: dwpaul Patches:
+ voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license
+ 689) ........ ................
+
+2009-02-13 16:44 +0000 [r175551] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_record.c: Merged revisions 175549 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 |
+ file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add
+ an option to keep the recorded file upon hangup. (closes issue
+ #14341) Reported by: fnordian ........
+
+2009-02-12 21:41 +0000 [r175370] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 |
+ russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines
+ Remove useless string copy, and make sscanf safe again ........
+
+2009-02-12 21:27 +0000 [r175342] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c, /: Merged revisions 175334 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009)
+ | 16 lines Merged revisions 175311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009)
+ | 9 lines Fix crashes when receiving certain T.38 packets. Also,
+ increase the maximum size of T.38 packets and warn users when
+ they try to set the limits above those maximums. (closes issue
+ #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: schern ........
+ ................
+
+2009-02-12 20:51 +0000 [r175300] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 175298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009)
+ | 15 lines Merged revisions 175294 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009)
+ | 9 lines Fix ParkedCall event information for From field in the
+ case of a blind transfer If the parker information can not be
+ obtained from the peer, try and see if the BLINDTRANSFER channel
+ variable has been set. Previously, a blind transfer to the
+ ParkAndAnnounce app would return nothing for the From. Closes
+ AST-189 ........ ................
+
+2009-02-12 20:48 +0000 [r175257-175297] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 |
+ russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines
+ Avoid using ast_strdupa() in a loop. ........
+
+ * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009)
+ | 4 lines Don't enable something by default that has a dependency
+ on something _not_ enabled by default. menuselect was not happy
+ with this. ........
+
+2009-02-12 18:50 +0000 [r175251] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 175250 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r175250 | kpfleming | 2009-02-12 12:48:52 -0600 (Thu, 12 Feb
+ 2009) | 1 line correct warning message to not refer specifically
+ to DAHDI ........
+
+2009-02-12 18:01 +0000 [r175190] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 175188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009)
+ | 12 lines Merged revisions 175187 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009)
+ | 6 lines Fix crash in event of failed attempt to transfer to
+ parking The peer may not necessarily exist, such as in the case
+ of a transfer to ParkAndAnnounce. In this case don't try to play
+ a sound to it. ........ ................
+
+2009-02-12 17:09 +0000 [r175130] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 175127 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r175127 | dvossel | 2009-02-12 11:07:17 -0600 (Thu, 12 Feb 2009)
+ | 4 lines Setting key rotation to be off by default Key rotation
+ breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0).
+ As a follow up to this, I am investigating possible ways to allow
+ key rotation to be on by default and not affect the other
+ branches, but for now it must be turned off. ........
+
+2009-02-12 17:08 +0000 [r175129] Russell Bryant <russell@digium.com>
+
+ * main/rtp.c, /: Merged revisions 175125 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009)
+ | 35 lines Merged revisions 175124 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009)
+ | 27 lines Don't send DTMF for infinite time if we do not receive
+ an END event. I thought that this was going to end up being a
+ pretty gnarly fix, but it turns out that there was actually
+ already a configuration option in rtp.conf, dtmftimeout, that was
+ intended to handle this situation. However, in between Asterisk
+ 1.2 and Asterisk 1.4, the code that processed the option got
+ lost. So, this commit brings it back to life. The default timeout
+ is 3 seconds. However, it is worth noting that having this be
+ configurable at all is not really the recommended behavior in RFC
+ 2833. From Section 3.5 of RFC 2833: Limiting the time period of
+ extending the tone is necessary to avoid that a tone "gets
+ stuck". Regardless of the algorithm used, the tone SHOULD NOT be
+ extended by more than three packet interarrival times. A slight
+ extension of tone durations and shortening of pauses is generally
+ harmless. Three seconds will pretty much _always_ be far more
+ than three packet interarrival times. However, that behavior is
+ not required, so I'm going to leave it with our legacy behavior
+ for now. Code from svn/asterisk/team/russell/issue_14460 (closes
+ issue #14460) Reported by: moliveras ........ ................
+
+2009-02-12 16:35 +0000 [r174947-175123] Mark Michelson <mmichelson@digium.com>
+
+ * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
+ 175121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 |
+ mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11
+ lines Make lock information for ao2_trylock be more useful and
+ gnarly Core show locks information involving an ao2_trylock did
+ not show the function that called ao2_trylock, but would instead
+ show ao2_trylock as the source of the lock. This is not useful
+ when trying to debug locking issues. One bizarre note is that
+ this logic is already in 1.4 but somehow did not get merged to
+ trunk or the 1.6.X branches. ........
+
+ * apps/app_queue.c, /: Merged revisions 174951 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174951 |
+ mmichelson | 2009-02-11 17:12:57 -0600 (Wed, 11 Feb 2009) | 3
+ lines Fix a bit of odd logic for announcing position. Sync with
+ 1.6.0's logic ........
+
+ * apps/app_queue.c, /: Merged revisions 174948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174948 |
+ mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) | 20
+ lines Fix odd "thank you" sound playing behavior in app_queue.c
+ If someone has configured the queue to play an position or
+ holdtime announcement, then it is odd and potentially unexpected
+ to hear a "Thank you for your patience" sound when no position or
+ holdtime was actually announced. This fixes the announcement so
+ that the "thanks" sound is only played in the case that a
+ position or holdtime was actually announced. There is a way that
+ the "thank you" sound can be played without a position or
+ holdtime, and that is to set announce-frequency to a value but
+ keep announce-position and announce-holdtime both turned off.
+ (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch
+ uploaded by putnopvut (license 60) Tested by: caspy ........
+
+ * apps/app_dial.c, main/channel.c, main/pbx.c, /,
+ apps/app_dictate.c, apps/app_waitforsilence.c,
+ include/asterisk/channel.h: Merged revisions 174945 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb
+ 2009) | 29 lines Fix 'd' option for app_dial and add new option
+ to Answer application The 'd' option would not work for channel
+ types which use RTP to transport DTMF digits. The only way to
+ allow for this to work was to answer the channel if we saw that
+ this option was enabled. I realized that this may cause issues
+ with CDRs, specifically with giving false dispositions and answer
+ times. I therefore modified ast_answer to take another parameter
+ which would tell if the CDR should be marked answered. I also
+ extended this to the Answer application so that the channel may
+ be answered but not CDRified if desired. I also modified
+ app_dictate and app_waitforsilence to only answer the channel if
+ it is not already up, to help not allow for faulty CDR answer
+ times. All of these changes are going into Asterisk trunk. For
+ 1.6.0 and 1.6.1, however, all the changes except for the change
+ to the Answer application will go in since we do not introduce
+ new features into stable branches (closes issue #14164) Reported
+ by: DennisD Patches: 14164.patch uploaded by putnopvut (license
+ 60) Tested by: putnopvut Review:
+ http://reviewboard.digium.com/r/145 ........
+
+2009-02-11 14:46 +0000 [r174846] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 174844 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174844 |
+ file | 2009-02-11 10:44:47 -0400 (Wed, 11 Feb 2009) | 10 lines
+ Tell the device state core a change happened when a channel is
+ freed but not a specific state. We need to do this because while
+ we know that the freeing of the channel may cause something to
+ become not in use we do not know this for sure. There may be
+ another channel that is still up which would cause it to be in
+ use. (closes issue #13238) Reported by: kowalma Patches:
+ 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: alecdavis ........
+
+2009-02-10 23:21 +0000 [r174769-174823] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 |
+ mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11
+ lines Fix potential for stack overflows in app_chanspy.c When
+ using the 'g' or 'e' options, the stack allocations that were
+ used could cause a stack overflow if a spyer stayed on the line
+ long enough without actually successfully spying on anyone. The
+ problem has been corrected by using static buffers and copying
+ the contents of the appropriate strings into them instead of
+ using functions like alloca or ast_strdupa ........
+
+ * main/manager.c, /: Merged revisions 174764 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 |
+ mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21
+ lines Fix an fd leak that would occur in HTTP AMI sessions The
+ explanation behind this fix is a bit complicated, and I've
+ already typed it up in the code as a huge comment inside of
+ manager.c, so I'll give the abridged version here. We needed a
+ way to separate action-specific data from session-specific data.
+ Unfortunately, the only way to maintain API compatibility and to
+ not have to change every single manager action was to rename the
+ current mansession structure and wrap it inside a new mansession
+ structure which actually contains action- specific data. (closes
+ issue #14364) Reported by: awk Patches: 14364_better.patch
+ uploaded by putnopvut (license 60) Tested by: putnopvut Review:
+ http://reviewboard.digium.com/r/148/ ........
+
+2009-02-10 20:17 +0000 [r174714] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 |
+ file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines
+ Only decrease inringing count if above zero. (issue #13238)
+ Reported by: kowalma ........
+
+2009-02-10 18:18 +0000 [r174590] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb
+ 2009) | 25 lines Merged revisions 174583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb
+ 2009) | 18 lines Improve behavior of jitterbuffer when
+ maxjitterbuffer is set. This change improves the way the
+ jitterbuffer handles maxjitterbuffer and dramatically reduces the
+ number of frames dropped when maxjitterbuffer is exceeded. In the
+ previous jitterbuffer, when maxjitterbuffer was exceeded, all new
+ frames were dropped until the jitterbuffer is empty. This change
+ modifies the code to only drop frames until maxjitterbuffer is no
+ longer exceeded. Also, previously when maxjitterbuffer was
+ exceeded, dropped frames were not tracked causing stats for
+ dropped frames to be incorrect, this change also addresses that
+ problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
+ by mnicholson (license 96) Tested by: mnicholson Review:
+ http://reviewboard.digium.com/r/144/ ........ ................
+
+2009-02-10 17:49 +0000 [r174545-174582] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174580 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174580 |
+ file | 2009-02-10 13:48:29 -0400 (Tue, 10 Feb 2009) | 4 lines Set
+ the type for the peer structure to be a peer as the default.
+ (closes issue #14447) Reported by: triccyx ........
+
+ * /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 |
+ file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines
+ Make the logic for inuse and inringing manipluation match that of
+ 1.4. The old broken logic would reset the values back to 0 during
+ certain scenarios causing the wrong state to be reported. (closes
+ issue #14399) Reported by: caspy (issue #13238) Reported by:
+ kowalma ........
+
+2009-02-10 07:07 +0000 [r174471-174504] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, apps/app_voicemail.c, /: Merged revisions
+ 174503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174503 |
+ tilghman | 2009-02-10 01:06:29 -0600 (Tue, 10 Feb 2009) | 2 lines
+ Fix0ring build ........
+
+ * apps/app_stack.c, /: Merged revisions 174470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174470 |
+ tilghman | 2009-02-09 23:39:33 -0600 (Mon, 09 Feb 2009) | 2 lines
+ Remove the usage of the KeepAlive app, as it no longer exists.
+ ........
+
+2009-02-10 05:13 +0000 [r174428-174440] Steve Murphy <murf@digium.com>
+
+ * apps/app_osplookup.c: This patch corrects warnings which seem to
+ appear only on 64-bit compilers, gcc-4.3.2.
+
+ * apps/app_rpt.c: One final fix in the 1.6.1 release only; some
+ variables the compiler worries "may not be initialized".
+
+ * apps/app_rpt.c, /: Merged revisions 174435 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174435 |
+ murf | 2009-02-09 21:49:02 -0700 (Mon, 09 Feb 2009) | 8 lines
+ This patch removes the use of AST_PBX_KEEPALIVE from app_rpt.c.
+ (closes issue #14435) Reported by: D_McNaul ........
+
+ * apps/app_rpt.c, /: Merged revisions 174432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174432 |
+ murf | 2009-02-09 21:36:22 -0700 (Mon, 09 Feb 2009) | 3 lines
+ More intptr_t work. ........
+
+ * apps/app_rpt.c, /: Merged revisions 174370 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174370 | murf | 2009-02-09 19:45:56 -0700 (Mon, 09 Feb 2009) |
+ 10 lines Merged revisions 174369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5
+ lines This patch solves some compiler complaints in both 32 and
+ 64-bit environments. ........ ................
+
+2009-02-09 17:47 +0000 [r174330] David Vossel <dvossel@digium.com>
+
+ * /, apps/app_externalivr.c: Merged revisions 174325 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r174325 | dvossel | 2009-02-09 11:26:02 -0600 (Mon, 09 Feb 2009)
+ | 9 lines Fixes issue with hangups not being sent and external
+ process never terminating. The ignore_hangup, run_dead, and
+ noanswer flags were never initilized to zero causing hangups to
+ never be issued. If the external script expects to be notified of
+ a hangup and never receives one, it runs indefinitely. (closes
+ issue #14251) Reported by: chris-mac Tested by: dvossel ........
+
+2009-02-09 17:30 +0000 [r174326-174329] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 |
+ mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3
+ lines Fix something I messed up in the merge I just did ........
+
+ * /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb
+ 2009) | 20 lines Merged revisions 174282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb
+ 2009) | 12 lines Don't do an SRV lookup if a port is specified
+ RFC 3263 says to do A record lookups on a hostname if a port has
+ been specified, so that's what we're going to do. See section
+ 4.2. (closes issue #14419) Reported by: klaus3000 Patches:
+ patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000
+ (license 65) ........ ................
+
+2009-02-09 14:50 +0000 [r174221] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon,
+ 09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4
+ lines Don't overwrite our pointer to the music class when music
+ on hold stops. We will use this if it starts again to see if we
+ can resume the music where it left off. (closes issue #14407)
+ Reported by: mostyn ........ ................
+
+2009-02-07 16:18 +0000 [r174154] Russell Bryant <russell@digium.com>
+
+ * /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009)
+ | 10 lines Merged revisions 174148 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009)
+ | 2 lines Fix a race condition that could cause a crash. ........
+ ................
+
+2009-02-07 00:09 +0000 [r174086] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009)
+ | 13 lines Merged revisions 174082 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009)
+ | 5 lines check ast_strlen_zero() before calling ast_strdupa() in
+ sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter
+ didn't actually upload a properly-formed patch, instead a
+ modified chan_sip.c file was uploaded. I created a patch to
+ determine the changes, then modified the suggested changes to
+ create a proper fix. The summary above is a complete description
+ of the changes. (closes issue #13547) Reported by: tecnoxarxa
+ Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258)
+ Tested by: tecnoxarxa ........ ................
+
+2009-02-06 19:30 +0000 [r173994-174043] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4
+ lines Don't subscribe to a mailbox on pseudo channels. It is
+ futile. This solves an issue where duplicated pseudo channels
+ would cause a crash because the first one would unsubscribe and
+ the next one would also try to unsubscribe the same subscription.
+ (closes issue #14322) Reported by: amessina ........
+
+ * /, channels/chan_sip.c: Merged revisions 173974 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) |
+ 15 lines Merged revisions 173967-173968 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4
+ lines Some clients do not put the call-id for replaces at the
+ beginning, so support it being anywhere in the string. (closes
+ issue #14350) Reported by: fhackenberger ........ r173968 | file
+ | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a
+ debug message I put in by accident. ........ ................
+
+2009-02-06 17:05 +0000 [r173964-173966] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb
+ 2009) | 14 lines Merged revisions 173917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
+ 2009) | 7 lines Limit the addition of the Contact header in SIP
+ responses according to various SIP RFCs. (closes issue #13602)
+ Reported by: hjourdain Tested by: mnicholson ........
+ ................
+
+ * main/ast_expr2.c, /, channels/chan_sip.c, main/ast_expr2.h:
+ revert revision 173964
+
+ * main/ast_expr2.c, /, channels/chan_sip.c, main/ast_expr2.h:
+ Merged revisions 173952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb
+ 2009) | 14 lines Merged revisions 173917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
+ 2009) | 7 lines Limit the addition of the Contact header in SIP
+ responses according to various SIP RFCs. (closes issue #13602)
+ Reported by: hjourdain Tested by: mnicholson ........
+ ................
+
+2009-02-06 16:01 +0000 [r173904] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_chanspy.c, /, main/audiohook.c: Merged revisions 173902
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r173902 | file | 2009-02-06 11:59:17 -0400 (Fri, 06 Feb
+ 2009) | 4 lines Always detach and destroy the whisper and barge
+ audiohooks. Additionally also allow an audiohook to be detached
+ if it has not been attached. (closes issue #14414) Reported by:
+ bluecrow76 ........
+
+2009-02-06 10:26 +0000 [r173850] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, /: Merged revisions 173848 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173848 |
+ russell | 2009-02-06 04:25:09 -0600 (Fri, 06 Feb 2009) | 2 lines
+ Resolve a memory leak that would occur on an invalid channel
+ given to Action: Status ........
+
+2009-02-05 23:53 +0000 [r173779] Mark Michelson <mmichelson@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 173776 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu,
+ 05 Feb 2009) | 14 lines Update extensions.conf.sample to be
+ correct. In trunk, the only necessary change pointed out was that
+ the call to ChanIsAvail uses an option that has been removed. For
+ the 1.6.1 branch, however, it appears that the sample file is
+ badly in need of updating since there are |'s used all over the
+ place there. My tentative plan is just to copy trunk's sample
+ config file to those branches since the info there is most
+ up-to-date and should be correct for use in 1.6.1 Thanks to macli
+ in #asterisk-dev for bringing this up ........
+
+2009-02-05 23:51 +0000 [r173778] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_sqlite.c: Oops, merge from trunk broke 1.6.1
+
+2009-02-05 23:31 +0000 [r173775] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 173773 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb
+ 2009) | 7 lines Properly set "seen" and "unseen" flags when
+ moving messages from the new to the old folder when using IMAP
+ for voicemail storage (closes issue #13905) Reported by: jaroth
+ Patches: foldermove_v2.patch uploaded by jaroth (license 50)
+ ........
+
+2009-02-05 21:06 +0000 [r173699] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 173697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600
+ (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009)
+ | 12 lines Add new configuration option to make shared IMAP
+ mailboxes function as expected. The new option is "imapvmshareid"
+ which is an ID to tag multiple mailboxes using the same IMAP
+ storage location to function as one mailbox. This allows all
+ messages to be retrieved for any user in the group. The patch
+ alters the 'X-Asterisk-VM-Extension' header that is responsible
+ for matching voicemails for a given user. (closes issue #13673)
+ Reported by: howardwilkinson ........ ................
+
+2009-02-05 20:35 +0000 [r173695] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 173693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb
+ 2009) | 20 lines Merged revisions 173692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb
+ 2009) | 12 lines Fix situations where queue members could be
+ autopaused unexpectedly Specifically, this patch prevents us from
+ autopausing members when we receive a busy or congestion frame
+ from them. (closes issue #14376) Reported by: fiddur Patches:
+ 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
+ ........ ................
+
+2009-02-05 19:37 +0000 [r173658] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_sqlite.c, /: Merged revisions 173657 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r173657 | tilghman | 2009-02-05 13:36:29 -0600 (Thu, 05 Feb 2009)
+ | 2 lines Change the first field, or we don't get the necessary
+ field separation. ........
+
+2009-02-05 18:50 +0000 [r173541-173595] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 173593 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600
+ (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb
+ 2009) | 3 lines Add some missing cleanup to app_mixmonitor
+ ........ ................
+
+ * apps/app_mixmonitor.c, /: Merged revisions 173589 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600
+ (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb
+ 2009) | 25 lines Fix a problem where a channel pointer becomes
+ invalid due to masquerading or hanging up. app_mixmonitor runs
+ its own thread to monitor the channel's activity and write the
+ mixed audio to a file. Since this thread runs independently of
+ the channel, it is possible that the mixmonitor thread's channel
+ pointer will point to freed memory when the channel either is
+ masqueraded or hangs up (technically, both cases are hangups, but
+ we need to handle the cases slightly differently). The solution
+ for this is to employ a datastore, which has the nice benefit of
+ allowing us to hook into channel masquerades and hangups and
+ update our pointer as necessary. If this looks familiar, this
+ same technique is employed in app_chanspy. app_chanspy is a bit
+ more involved since it does a lot more operations on the channel
+ that is being spied upon. app_mixmonitor does have an extra touch
+ that app_chanspy doesn't have, though. Since there is a thread
+ race between the channel's thread and the mixmonitor thread on a
+ hangup, we em- ploy a condition-and-boolean combination to ensure
+ that the channel thread finishes with our structure before the
+ mixmonitor thread attempts to free it. No crashes! (closes issue
+ #14374) Reported by: aragon Patches: 14374.patch uploaded by
+ putnopvut (license 60) Tested by: aragon, putnopvut ........
+ ................
+
+ * apps/app_queue.c, /: Merged revisions 173507 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 |
+ mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7
+ lines Fix some areas where the incorrect interface was passed to
+ ast_device_state I swear it feels like I already did this once...
+ (closes issue #14359) Reported by: francesco_r ........
+
+2009-02-04 21:32 +0000 [r173506] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, channels/iax2-parser.h, /: Merged revisions
+ 173502 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173502 |
+ dvossel | 2009-02-04 15:25:14 -0600 (Wed, 04 Feb 2009) | 9 lines
+ Fixes issue with IAX2 transfer not handing off calls. Reverts
+ changes in 116884 Fixes issue with IAX2 transfers not taking
+ place. As it was, a call that was being transfered would never be
+ handed off correctly to the call ends because of how call numbers
+ were stored in a hash table. The hash table,
+ "iax_peercallno_pvt", storing all the current call numbers did
+ not take into account the complications associated with
+ transferring a call, so a separate hash table was required. This
+ second hash table "iax_transfercallno_pvt" handles calls being
+ transfered, once the call transfer is complete the call is
+ removed from the transfer hash table and added to the peer hash
+ table resuming normal operations. Addition functions were created
+ to handle storing, removing, and comparing items in the
+ iax_transfercallno_pvt table. The changes reverted in 116884
+ caused backwards compatibility issues involving iax2 transfer
+ with 1.6.0, 1.4, and 1.2. (closes issue #13468) Reported by:
+ nicox Tested by: dvossel ........
+
+2009-02-04 21:28 +0000 [r173505] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/features.h, /, main/features.c: Merged revisions
+ 173500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009)
+ | 23 lines Merged revisions 173211 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009)
+ | 17 lines Parking attempts made to one end of a bridge no longer
+ will hang up due to a parking failure. Parking attempts made
+ using either one-touch, or doing either a blind or assisted
+ transfer to the parking extension now keep up the bridge instead
+ of hanging up the attempted parked party. Normal causes for the
+ parking attempt to fail includes the specific specified extension
+ (via PARKINGEXTEN) not being available or if all the parking
+ spaces are currently in use. To avoid having to reverse a
+ masquerade park_space_reserve was made to provide foresight if a
+ parking attempt will succeed and if so reserve the parking space.
+ (closes issue #13494) Reported by: mdu113 Reviewed by Russell:
+ http://reviewboard.digium.com/r/133/ ........ ................
+
+2009-02-04 18:52 +0000 [r173459] Tilghman Lesher <tlesher@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 173458 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 |
+ tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines
+ When using a socket as a FILE *, the stdio functions will
+ sometimes try to do an fseek() on the stream, which is an invalid
+ operation for a socket. Turning off buffering explicitly lets the
+ stdio functions know they cannot do this, thus avoiding a
+ potential error. (closes issue #14400) Reported by: fnordian
+ Patches: tcptls.patch uploaded by fnordian (license 110) ........
+
+2009-02-04 17:46 +0000 [r173356-173399] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 173397 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb
+ 2009) | 11 lines Merged revisions 173396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb
+ 2009) | 3 lines Revert my previous change because it was stupid
+ ........ ................
+
+ * apps/app_chanspy.c, /: Merged revisions 173393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb
+ 2009) | 11 lines Merged revisions 173392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb
+ 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever
+ matter, but it's needed. ........ ................
+
+ * /, main/file.c: Merged revisions 173354 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 |
+ mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30
+ lines Fix a problem where file playback would cause fds to remain
+ open forever The problem came from the fact that a frame read
+ from a format interpreter was not freed. Adding a call to
+ ast_frfree fixed this. The explanation for why this caused the
+ problem is a bit complex, but here goes: There was a problem in
+ all versions of Asterisk where the embedded frame of a filestream
+ structure was referenced after the filestream was freed. This was
+ fixed by adding reference counting to the filestream structure.
+ The refcount would increase every time that a filestream's frame
+ pointer was pointing to an actual frame of data. When the frame
+ was freed, the refcount would decrease. Once the refcount reached
+ 0, the filestream was freed, and as part of the operation, the
+ open files were closed as well. Thus it becomes more clear why a
+ missing ast_frfree would cause a reference leak and cause the
+ files to not be closed. You may ask then if there was a frame
+ leak before this patch. The answer to that is actually no! The
+ filestream code was "smart" enough to know that since the frame
+ we received came from a format interpreter, the frame had no
+ malloced data and thus didn't need to be freed. Now, however,
+ there is cleanup that needs to be done when we finish with the
+ frame, so we do need to call ast_frfree on the frame to be sure
+ that the refcount for the filestream is decremented
+ appropriately. (closes issue #14384) Reported by: fiddur Patches:
+ 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur,
+ putnopvut ........
+
+2009-02-04 00:46 +0000 [r173313] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 173311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 |
+ tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10
+ lines Ensure that commas placed in the middle of extension
+ character classes do not interfere with correct parsing of the
+ extension. Also, if an unterminated character class DOES make its
+ way into the pbx core (through some other method), ensure that it
+ does not crash Asterisk. (closes issue #14362) Reported by:
+ Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: Corydon76 ........
+
+2009-02-03 00:26 +0000 [r173115] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 173104 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600
+ (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009)
+ | 5 lines Add warning to standard config, that globals may be
+ overridden by other dialplan configuration files. (closes issue
+ #14388) Reported by: macli ........ ................
+
+2009-02-03 00:01 +0000 [r173069] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 173067 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009)
+ | 9 lines Merged revisions 173066 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009)
+ | 2 lines Fix a feature inheritance bug I added after code review
+ ........ ................
+
+2009-02-02 18:15 +0000 [r172895] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/res_ldap.conf.sample: Merged revisions 172894 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02
+ Feb 2009) | 7 lines Update the res_ldap.conf file with a better
+ working example. (closes issue #13861) Reported by: scramatte
+ Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage
+ (license 10) Tested by: jcovert ........
+
+2009-02-01 02:45 +0000 [r172708-172743] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 172741 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009)
+ | 4 lines Blank argument crashes Asterisk (closes issue #14377)
+ Reported by: amorsen ........
+
+ * /, funcs/func_strings.c: Merged revisions 172706 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009)
+ | 7 lines Don't increment the loop, now that incrementing is
+ taken care of by the decoder function. (closes issue #14363)
+ Reported by: andrew53 Patches: func_strings_filter.patch uploaded
+ by andrew53 (license 519) ........
+
+2009-01-31 00:07 +0000 [r172636-172638] Terry Wilson <twilson@digium.com>
+
+ * configs/features.conf.sample, /: Merged revisions 172581 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30
+ Jan 2009) | 2 lines Remove incorret line from sample config
+ ........
+
+ * CHANGES, configs/features.conf.sample, apps/app_dial.c,
+ main/global_datastores.c, /, main/features.c,
+ include/asterisk/global_datastores.h: Merged revisions 172580 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r172580 | twilson | 2009-01-30 15:29:12 -0600
+ (Fri, 30 Jan 2009) | 44 lines Merged revisions 172517 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009)
+ | 37 lines Fix feature inheritance with builtin features When
+ using builtin features like parking and transfers, the
+ AST_FEATURE_* flags would not be set correctly for all instances
+ when either performing a builtin attended transfer, or parking a
+ call and getting the timeout callback. Also, there was no way on
+ a per-call basis to specify what features someone should have on
+ picking up a parked call (since that doesn't involve the Dial()
+ command). There was a global option for setting whether or not
+ all users who pickup a parked call should have
+ AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
+ PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
+ variable which can be set either in the dialplan or with setvar
+ in channels that support it. This variable can be set to any
+ combination of 't', 'k', 'w', and 'h' (case insensitive matching
+ of the equivalent dial options), to set what features should be
+ activated on this channel. The patch moves the setting of the
+ features datastores into the bridging code instead of app_dial to
+ help facilitate this. 2) adds global options parkedcallparking,
+ parkedcallhangup, and parkedcallrecording to be similar to the
+ parkedcalltransfers option for globally setting features. 3) has
+ builtin_atxfer call builtin_parkcall if being transfered to the
+ parking extension since tracking everything through multiple
+ masquerades, etc. is difficult and error-prone 4) attempts to fix
+ all cases of return calls from parking and completed builtin
+ transfers not having the correct permissions (closes issue
+ #14274) Reported by: aragon Patches:
+ fix_feature_inheritence.diff.txt uploaded by otherwiseguy
+ (license 396) Tested by: aragon, otherwiseguy Review
+ http://reviewboard.digium.com/r/138/ ........ ................
+
+2009-01-30 22:24 +0000 [r172609] Mark Michelson <mmichelson@digium.com>
+
+ * /, include/asterisk/channel.h: Merged revisions 172598 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri,
+ 30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h
+ ........
+
+2009-01-30 08:27 +0000 [r172509] Olle Johansson <oej@edvina.net>
+
+ * CHANGES: Remove an extra "the" and restructure a bit
+
+2009-01-29 23:53 +0000 [r172504] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_rpt.c, main/asterisk.c, /, autoconf/ast_func_fork.m4,
+ configure, main/app.c: Merged revisions 172441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009)
+ | 16 lines Merged revisions 172438 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009)
+ | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at
+ startup. Otherwise, if Asterisk runs as a non-root user and the
+ administrator does a 'restart now', Asterisk loses the ability to
+ set QOS on packets. (closes issue #14004) Reported by: nemo
+ Patches: 20090105__bug14004.diff.txt uploaded by Corydon76
+ (license 14) Tested by: Corydon76 ........ ................
+
+2009-01-29 22:05 +0000 [r172435] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 172400 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 |
+ rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12
+ lines channels/chan_dahdi.c * Added doxygen comments to the major
+ dahdi structures. * Fixed PRI and SS7 using an incorrect string
+ value if the extension delimiter is not present in the Dial()
+ function. * Fixed SS7 not checking if the dialed extension is at
+ least as long as the stripmsd option. * Fixed PRI not handling
+ unknown TON/NPI prefix letters correctly. * Fixed some
+ uninitialized string variables on FXS ports.
+ configs/chan_dahdi.conf.sample * Updated some documentation.
+ ........
+
+2009-01-29 20:54 +0000 [r172317-172402] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/muted.c, /: Merged revisions 146514 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk (closes issue
+ #14360) Reported by: oej ........ r146514 | russell | 2008-10-05
+ 17:11:30 -0500 (Sun, 05 Oct 2008) | 2 lines Make this build on my
+ mac. ........
+
+ * configs/func_odbc.conf.sample, /: Merged revisions 172315 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29
+ Jan 2009) | 2 lines Better document mode=multirow, based upon a
+ conversation with Jared. ........
+
+2009-01-29 13:50 +0000 [r172272] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 172271
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29
+ Jan 2009) | 5 lines The realtime_pgsql.sql script is missing a
+ couple of fields. closes issue #14339) Reported by: fiddur
+ Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678)
+ ........
+
+2009-01-29 11:24 +0000 [r172218-172235] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 172234 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r172234 |
+ oej | 2009-01-29 12:19:29 +0100 (Tor, 29 Jan 2009) | 7 lines Make
+ sure register= line supports both port and expiry at the same
+ time. (closes issue #14185) Reported by: Nick_Lewis Patches:
+ chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657)
+ Tested by: Nick_Lewis ........
+
+ * /, channels/chan_sip.c: Merged revisions 172173 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24
+ lines Merged revisions 172169 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16
+ lines Make sure that we always add the hangupcause headers. In
+ some cases, the owner was disconnected before we checked for the
+ cause. This patch implements a temporary storage in the pvt and
+ use that instead. The code is based on ideas from code from
+ Adomjan in issue #13385 (Add support for Reason: header) Thanks
+ to Klaus Darillion for testing! (closes issue #14294) related to
+ issue #13385 Reported by: klaus3000 and adomjan Patches:
+ bug14294b.diff uploaded by oej (license 306) Based on
+ 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan
+ (license 487) Tested by: oej, klaus3000 ........ ................
+
+ * /, configs/sip.conf.sample: Merged revisions 171880 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r171880 | oej | 2009-01-28 14:26:31 +0100 (Ons, 28 Jan 2009) | 2
+ lines Add some more notes about device matching. ........
+
+2009-01-28 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.1-rc1 released
+
+2009-01-28 22:52 +0000 [r172133] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c, /: Merged revisions 172131 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r172131 | tilghman | 2009-01-28 16:48:01 -0600 (Wed, 28 Jan 2009)
+ | 7 lines Fix how we skip fields (to avoid fields which don't
+ exist) when doing an UPDATE. (closes issue #14205) Reported by:
+ maxgo Patches: 20090128__bug14205__5.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: blitzrage ........
+
+2009-01-28 20:56 +0000 [r172067] Steve Murphy <murf@digium.com>
+
+ * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /,
+ main/features.c, include/asterisk/channel.h: Merged revisions
+ 172063 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) |
+ 52 lines Merged revisions 172030 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) |
+ 46 lines This patch fixes h-exten running misbehavior in
+ manager-redirected situations. What it does: 1. A new Flag value
+ is defined in include/asterisk/channel.h,
+ AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
+ bridge hangup exten code not to run the h-exten there (nor
+ publish the bridge cdr there). It will done at the pbx-loop level
+ instead. 2. In the manager Redirect code, I set this flag on the
+ channel if the channel has a non-null pbx pointer. I did the same
+ for the second (chan2) channel, which gets run if name2 is set...
+ and the first succeeds. 3. I restored the ending of the cdr for
+ the pbx loop h-exten running code. Don't know why it was removed
+ in the first place. 4. The first attempt at the fix for this bug
+ was to place code directly in the async_goto routine, which was
+ called from a large number of places, and could affect a large
+ number of cases, so I tested that fix against a fair number of
+ transfer scenarios, both with and without the patch. In the
+ process, I saw that putting the fix in async_goto seemed not to
+ affect any of the blind or attended scenarios, but still, I was
+ was highly concerned that some other scenarios I had not tested
+ might be negatively impacted, so I refined the patch to its
+ current scope, and jmls tested both. In the process, tho, I saw
+ that blind xfers in one situation, when the one-touch blind-xfer
+ feature is used by the peer, we got strange h-exten behavior. So,
+ I inserted code to swap CDRs and to set the HANGUP_DONT field, to
+ get uniform behavior. 5. I added code to the bridge to obey the
+ HANGUP_DONT flag, skipping both publishing the bridge CDR, and
+ running the h-exten; they will be done at the pbx-loop (higher)
+ level instead. 6. I removed all the debug logs from the patch
+ before committing. 7. I moved the AUTOLOOP set/reset in the
+ h-exten code in res_features so it's only done if the h-exten is
+ going to be run. A very minor performance improvement, but
+ technically correct. (closes issue #14241) Reported by: jmls
+ Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer
+ uploaded by murf (license 17) Tested by: murf, jmls ........
+ ................
+
+2009-01-28 17:29 +0000 [r171966] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 171964 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600
+ (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28
+ Jan 2009) | 2 lines Clarify log message (suggested by manxpower
+ on #asterisk-dev) ........ ................
+
+2009-01-28 13:21 +0000 [r171857] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Merged revisions 171838 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons,
+ 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2
+ lines Add a better explanation of the difference between the
+ device namespace and the dialplan for newbies. ........
+ ................
+
+2009-01-27 22:01 +0000 [r171620-171693] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 171691 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600
+ (Tue, 27 Jan 2009) | 47 lines Merged revisions 171689 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan
+ 2009) | 39 lines Fix devicestate problems for "always-on" agent
+ channels A revision to chan_agent attempted to "inherit" the
+ device state of the underlying channel in order to report the
+ device state of an agent channel more accurately. The problem
+ with the logic here is that it makes no sense to use this for
+ always-on agents. If the agent is logged in, then to the
+ underlying channel, the agent will always appear to be "in use,"
+ no matter if the agent is on a call or not. The reason is that to
+ the underlying channel, the channel is currently in use on a call
+ to the AgentLogin application. The most common cause that I found
+ for this issue to occur was for a SIP channel to be the
+ underlying channel type for an Agent channel. If the SIP phone
+ re-registers, then the registration will cause the device state
+ core to query the device state of the SIP channel. Since the SIP
+ channel is in use, the Agent channel would also inherit this
+ status. Once the agent channel was set to "in use" there was no
+ way that the device state could change on that channel unless the
+ agent logged out. The solution for this problem is a bit
+ different in 1.4 than it is in the other branches. In 1.4, there
+ will be a one-line fix to make sure that only callback agents
+ will inherit device state from their underlying channel type. For
+ the other branches of Asterisk, since callback support has been
+ removed, there is also no need for device state inheritance in
+ chan_agent, so I will simply be removing it from the code. In
+ addition, the 1.4 source is getting a new comment to help the
+ next person who edits chan_agent.c. I'm adding a comment that a
+ agent_pvt's loginchan field may be used to determine if the agent
+ is a callback agent or not. (closes issue #14173) Reported by:
+ nathan Patches: 14173.patch uploaded by putnopvut (license 60)
+ Tested by: nathan, aramirez ........ ................
+
+ * /, main/slinfactory.c: Merged revisions 171622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan
+ 2009) | 26 lines Merged revisions 171621 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan
+ 2009) | 18 lines Prevent a crash from occurring when a jitter
+ buffer interpolated frame is removed from a slinfactory
+ slinfactory used the "samples" field of an ast_frame in order to
+ determine the amount of data contained within the frame. In
+ certain cases, such as jitter buffer interpolated frames, the
+ frame would have a non-zero value for "samples" but have NULL
+ "data" This caused a problem when a memcpy call in
+ ast_slinfactory_read would attempt to access invalid memory. The
+ solution in use here is to never feed frames into the slinfactory
+ if they have NULL "data" (closes issue #13116) Reported by:
+ aragon Patches: 13116.diff uploaded by putnopvut (license 60)
+ ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 171618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 |
+ mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24
+ lines Fix queue crashes that would occur after the calling
+ channel was masqueraded. The data passed to the
+ end_bridge_callback was assumed to be data which was still
+ stack'd. The problem was that with some call features, attended
+ transfers in particular, a new bridge thread is started once the
+ feature completes, meaning that when the end_bridge_callback is
+ called, the end_bridge_callback_data was invalid. To fix this
+ problem, there are two measures taken 1. Instead of pointing to
+ stacked data, we now used heap-allocated data for passing to the
+ end_bridge_callback in app_queue 2. Since bridges can end
+ multiple times on a single logical call, we wait until the final
+ bridge is broken to actually set any queue variables. This is
+ accomplished through reference-counting and the use of an
+ end_bridge_callback_data_fixup function in app_queue.c (closes
+ issue #14260) Reported by: ccesario Patches: 14260.patch uploaded
+ by putnopvut (license 60) Tested by: ccesario ........
+
+2009-01-27 15:19 +0000 [r171540] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 171528 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23
+ lines Solving the same issue, but a bit different in trunk...
+ Merged revisions 171527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13
+ lines Use the same branch tag in CANCEL as in INVITE Originally
+ putnopvut implemented some changes in revision 142079 that
+ according to the bug report seemed to have worked then, but
+ somehow fails now. I guess code, as humans, get old and forget
+ stuff. Anyway, this bug caused CANCEL not to work with picky
+ systems. Thanks Fredrik for pointing out where the bug in the SIP
+ messaging was. (closes issue #14346) Reported by: oej Patches:
+ bug14346.diff uploaded by oej (license 306) Tested by: oej
+ ........ ................
+
+2009-01-26 14:58 +0000 [r171361] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 171326 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) |
+ 17 lines Merged revisions 171264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9
+ lines Don't retransmit 401 on REGISTER requests when
+ alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000
+ Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000 ........
+ ................
+
+2009-01-26 00:04 +0000 [r171190] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_oss.c, /: Merged revisions 171188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009)
+ | 13 lines Merged revisions 171187 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009)
+ | 6 lines Correctly track the hookstate (closes issue #13686)
+ Reported by: itiliti Patches: 20081013__bug13686.diff.txt
+ uploaded by Corydon76 (license 14) ........ ................
+
+2009-01-25 13:40 +0000 [r170982] Sean Bright <sean.bright@gmail.com>
+
+ * /, apps/app_page.c: Merged revisions 170980 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan
+ 2009) | 16 lines Merged revisions 170979 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan
+ 2009) | 9 lines Resolve a logic error that was causing Page() to
+ crash when more than one channel was specified. (closes issue
+ #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
+ uploaded by seanbright (license 71) Tested by: kc0bvu ........
+ ................
+
+2009-01-25 02:52 +0000 [r170945] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 170943 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009)
+ | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second
+ part of this macro is written as 0[a] instead of a[0], it will
+ force a failure if the macro is used on a C++ object that
+ overloads the [] operator. ........
+
+2009-01-24 13:57 +0000 [r170839] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/res_odbc.conf.sample, /: Merged revisions 170837 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600
+ (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24
+ Jan 2009) | 2 lines Remove superfluous implementation note
+ (closes issue #14319) ........ ................
+
+2009-01-23 23:53 +0000 [r170831] Richard Mudgett <rmudgett@digium.com>
+
+ * /, doc/tex/Makefile: Merged revisions 170794 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 |
+ rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line
+ Fix asterisk.pdf generation if branch name has an underscore in
+ it. ........
+
+2009-01-23 22:59 +0000 [r170792] Russell Bryant <russell@digium.com>
+
+ * /, doc/tex/Makefile: Merged revisions 170790 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r170790 |
+ russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines
+ Don't blow up if a branch name has an underscore in it ........
+
+2009-01-23 20:57 +0000 [r170693-170722] Mark Michelson <mmichelson@digium.com>
+
+ * configs/res_odbc.conf.sample, /: Merged revisions 170720 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600
+ (Fri, 23 Jan 2009) | 16 lines Merged revisions 170719 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan
+ 2009) | 8 lines Add notes to the idlecheck explanation in
+ res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000
+ Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by
+ klaus3000 (license 65) ........ ................
+
+ * contrib/i18n.testsuite.conf, /: Merged revisions 170677 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r170677 | mmichelson | 2009-01-23 14:23:00 -0600
+ (Fri, 23 Jan 2009) | 22 lines Merged revisions 170671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan
+ 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use
+ deprecated syntax * Convert Wait,1 to Wait(1) * Convert
+ SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
+ priorities beyond the first Also added test for Chinese numbers,
+ too. (closes issue #14320) Reported by: dant Patches:
+ i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
+ 670) ........ ................
+
+2009-01-23 20:20 +0000 [r170664] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 170652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) |
+ 11 lines Merged revisions 170648 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4
+ lines When a channel is answered make sure any indications
+ currently playing stop. Usually the phone would do this but if
+ the channel was already answered then they are being generated by
+ Asterisk and we darn well need to stop them. (closes issue
+ #14249) Reported by: RadicAlish ........ ................
+
+2009-01-23 19:37 +0000 [r170637] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 170608 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r170608 | tilghman | 2009-01-23 13:25:10 -0600
+ (Fri, 23 Jan 2009) | 9 lines Merged revisions 170588 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23
+ Jan 2009) | 2 lines Additions to AST-2009-001 ........
+ ................
+
+2009-01-23 19:10 +0000 [r170507-170571] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 170569 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) |
+ 11 lines Merged revisions 170568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4
+ lines When a call is forwarded stop any active indications. The
+ new channel will provide an indication, if need be, itself.
+ (closes issue #14310) Reported by: RadicAlish ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 170505 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170505 | file | 2009-01-23 14:09:45 -0400 (Fri, 23 Jan 2009) |
+ 11 lines Merged revisions 170504 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4
+ lines Use the on hold flag to see if the call is on hold or not.
+ It is possible that our address for them will still be valid even
+ though they are on hold. (closes issue #14295) Reported by:
+ klaus3000 ........ ................
+
+2009-01-23 17:49 +0000 [r170502] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_h323.c: Merged revisions 170501 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r170501 | mvanbaak | 2009-01-23 18:46:02 +0100 (Fri, 23 Jan 2009)
+ | 1 line let's use SENTINEL where needed ........
+
+2009-01-23 16:35 +0000 [r170458] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: MWI messages included in CID spill was not
+ being properly handled and prevented the call from being
+ processed (issue #14313) Reported by: seandarcy Tested by:
+ dbailey
+
+2009-01-23 15:51 +0000 [r170395] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 170393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170393 | mmichelson | 2009-01-23 09:44:27 -0600 (Fri, 23 Jan
+ 2009) | 36 lines Merged revisions 170392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan
+ 2009) | 28 lines Fix broken call pickup There was a subtle change
+ in ast_do_masquerade which resulted in failed attempts to pickup
+ calls. The problem was that the value of the AST_FLAG_OUTGOING
+ flag was copied from the clone to the original channel. In the
+ case of call pickup, this meant that the AST_FLAG_OUTGOING flag
+ ended up being cleared on the channel that was attempting to
+ execute the pickup. Because this flag was not set, when ast_read
+ came across an answer frame, it ignored it. The result of this
+ was that the calling channel was never properly answered. This
+ fix changes the behavior in ast_do_masquerade to set the flags on
+ the original channel to the union of the flags on the clone
+ channel. This way, if the AST_FLAG_OUTGOING flag is set on either
+ of the two channels involved in the masquerade, the resulting
+ channel will have the flag set as well. (closes issue #14206)
+ Reported by: francesco_r Patches: 14206.patch uploaded by
+ putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut
+ ........ ................
+
+2009-01-22 20:06 +0000 [r170242] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 170240 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170240 | file | 2009-01-22 16:04:39 -0400 (Thu, 22 Jan 2009) |
+ 14 lines Merged revisions 170239 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7
+ lines Don't crash if RTCP is not enabled on an RTP structure but
+ statistics are output. (closes issue #14234) Reported by: jcovert
+ Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551)
+ rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........
+ ................
+
+2009-01-22 17:21 +0000 [r170178] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 170165 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170165 | tilghman | 2009-01-22 11:19:28 -0600 (Thu, 22 Jan 2009)
+ | 13 lines Merged revisions 170158 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009)
+ | 6 lines Allow global variables after substitution to be as long
+ as other variables. (closes issue #14263) Reported by: markd
+ Patches: 20090120__bug14263.diff.txt uploaded by Corydon76
+ (license 14) ........ ................
+
+2009-01-22 16:54 +0000 [r170049-170150] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 170148 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) |
+ 11 lines Merged revisions 170147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4
+ lines If we are unable to request a DAHDI pseudo channel and we
+ are using the user introduction without review option make sure
+ it gets unset so other code does not blindly assume a DAHDI
+ pseudo channel exists. (closes issue #14282) Reported by:
+ cheesegrits ........ ................
+
+ * main/pbx.c, /: Merged revisions 170051 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170051 | file | 2009-01-22 11:14:50 -0400 (Thu, 22 Jan 2009) |
+ 13 lines Merged revisions 170050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6
+ lines Do a string comparison instead of pointer comparison since
+ some people specify the context they are actually in as an
+ argument to get around some funkiness. (closes issue #14011)
+ Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga
+ (license 665) ........ ................
+
+ * apps/app_parkandannounce.c, /: Merged revisions 170047 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan
+ 2009) | 4 lines Clear the autoloop flag when parsing and setting
+ the context/extension/priority to go back to. When the channel
+ executes a PBX again we want it to start out at the point we
+ explicitly say and at that point it will not yet be doing
+ autoloop. (closes issue #14304) Reported by: jcovert ........
+
+2009-01-22 00:46 +0000 [r169946] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 169944 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r169944 | tilghman | 2009-01-21 18:44:52 -0600
+ (Wed, 21 Jan 2009) | 16 lines Merged revisions 169943 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009)
+ | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really
+ wanted to ask is whether autoconf detected a static initializer
+ value. This fixes rwlocks on all such platforms (mainly, Mac OS
+ X). (closes issue #13767) Reported by: jcovert Patches:
+ 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: jcovert, Corydon76 ........ ................
+
+2009-01-21 23:28 +0000 [r169871] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, /: Merged revisions 169869 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r169869 | file | 2009-01-21 19:25:27 -0400 (Wed, 21 Jan 2009) |
+ 11 lines Merged revisions 169867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4
+ lines Read lock the contexts to maintain the locking order when
+ we are notified that the state of a device has changed. (closes
+ issue #13839) Reported by: mcallist ........ ................
+
+2009-01-21 22:23 +0000 [r169830] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, doc/tex/extensions.tex: Merged revisions 169793 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r169793 | mvanbaak | 2009-01-21 23:04:16 +0100 (Wed, 21 Jan 2009)
+ | 2 lines remove duplicated sentence. ........
+
+2009-01-21 22:11 +0000 [r169792-169796] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/say.c: Merged revisions 169794 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169794 |
+ mmichelson | 2009-01-21 16:10:02 -0600 (Wed, 21 Jan 2009) | 17
+ lines Fix a crash when saying certain numbers in Chinese This
+ commit fixes a crash that was occurring when attempting to say a
+ number between 10000 and 100000 due to dividing by 0. This also
+ removes some places where a "zero" is spoken when it should not
+ be. (closes issue #14291) Reported by: dant Patches:
+ say.c-14291.diff uploaded by dant (license 670) Tested by: dant
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 169791 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169791 |
+ mmichelson | 2009-01-21 15:53:55 -0600 (Wed, 21 Jan 2009) | 18
+ lines Further fix some oddities in sip show users and sip show
+ peers logic ccesario on IRC pointed out that his sip peers were
+ not displayed properly when he would issue the command "sip show
+ peers." The problem was that the onlymatchonip field was used to
+ determine if the endpoint was a "peer" or "user." The tricky part
+ is that a "friend" is supposed to be treated as both a "user" and
+ a "peer" but the logic would not allow "friends" to show up as
+ "peers" since onlymatchonip was set to FALSE for friends. I have
+ modified the sip_peer structure to more explicitly keep track of
+ what type endpoint it is so that the various manager and CLI
+ commands will display the expected information Reported by
+ ccesario via IRC Tested by ccesario ........
+
+2009-01-21 21:05 +0000 [r169725] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 169723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r169723 | tilghman | 2009-01-21 15:03:40 -0600 (Wed, 21 Jan 2009)
+ | 15 lines Merged revisions 169722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009)
+ | 8 lines Extra NULLs in the output cause some terminal types to
+ abort in the middle of a color code, causing terminal weirdness.
+ (closes issue #14130) Reported by: coolmig Patches:
+ 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, coolmig ........ ................
+
+2009-01-21 17:40 +0000 [r169674] Steve Murphy <murf@digium.com>
+
+ * utils/refcounter.c, /: Merged revisions 169673 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169673 |
+ murf | 2009-01-21 10:21:40 -0700 (Wed, 21 Jan 2009) | 14 lines
+ This patch corrects a segfault reported in 14289, due to a null
+ ptr being refd. Yes, seanbright is right in the bug comments,
+ that is the fix. Sorry for this oversight; I guess my personal
+ usage didn't have this happen! murf (closes issue #14289)
+ Reported by: jamesgolovich ........
+
+2009-01-21 10:49 +0000 [r169622-169626] Russell Bryant <russell@digium.com>
+
+ * /: Merged revisions 169625 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169625 |
+ russell | 2009-01-21 04:49:00 -0600 (Wed, 21 Jan 2009) | 2 lines
+ Remove properties that erroneously got merged into trunk ........
+
+ * main/tcptls.c, /: Merged revisions 169620 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169620 |
+ russell | 2009-01-21 04:26:07 -0600 (Wed, 21 Jan 2009) | 10 lines
+ Fix a regression in TCP support. This patch fixes a problem that
+ caused chan_sip to think that every open TCP session was to a
+ remote address of 0.0.0.0:0. (closes issue #14287) Reported by:
+ jamesgolovich Patches: bug-14287.diff.txt uploaded by
+ jamesgolovich (license 176) ........
+
+2009-01-21 00:35 +0000 [r169559-169613] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 169611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169611 |
+ mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22
+ lines Fix device state parsing issues for channel names with
+ multiple slashes The fix being applied is a bit different for
+ trunk and the 1.6.X branches. For trunk, we only wish to strip
+ off the characters beyond the second slash if the channel is a
+ Local channel (i.e. we are removing the /n from the device name).
+ Other channel technologies with multiple slashes (e.g. DAHDI)
+ need the information after the second slash in order to get the
+ proper device state information. In addition to this fix, the
+ 1.6.X branches are receiving a much more important fix as well.
+ The problem in 1.6.X is that the member's device name was being
+ directly changed instead of having a copy changed. This meant
+ that we would strip off the second slash and trailing characters
+ and then leave the member's device name like that permanently
+ thereafter. (closes issue #14014) Reported by: kebl0155 Patches:
+ 14014_number2.patch uploaded by putnopvut (license 60) Tested by:
+ kebl0155 ........
+
+ * apps/app_queue.c, /: Merged revisions 169574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169574 |
+ mmichelson | 2009-01-20 15:57:24 -0600 (Tue, 20 Jan 2009) | 6
+ lines Use the default timeout for a queue instead of -1 (closes
+ issue #14272) Reported by: timking ........
+
+ * /, channels/chan_sip.c: Merged revisions 169557 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169557 |
+ mmichelson | 2009-01-20 14:10:31 -0600 (Tue, 20 Jan 2009) | 19
+ lines Convert the character pointers in a sip_request to be
+ pointer offsets When an ast_str expands to hold more data, any
+ pointers that were pointing to the data prior to the expansion
+ will be pointing at invalid memory. This change makes such
+ pointers used in chan_sip.c instead be offsets from the beginning
+ of the string so that the same math may be applied no matter
+ where in memory the string resides. To help ease this transition,
+ a macro called REQ_OFFSET_TO_STR has been added to chan_sip.c so
+ that given a sip_request and an offset, the string at that offset
+ is returned. (closes issue #14220) Reported by: riksta Tested by:
+ putnopvut Review http://reviewboard.digium.com/r/126/ ........
+
+2009-01-20 19:31 +0000 [r169488-169554] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 169510 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169510 |
+ twilson | 2009-01-20 13:22:24 -0600 (Tue, 20 Jan 2009) | 7 lines
+ Make a proper builtin attended transfer to parking work This is
+ an ugly hack from 1.4 that allows the timeout callback from a
+ parked call to use the right channel name for the callback when
+ the park is done with a builtin attended transfer (that isn't
+ completed early). This hasn't ever worked in trunk and no one has
+ complained yet, so eh. ........
+
+ * /, main/features.c: Merged revisions 169486 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r169486 | twilson | 2009-01-20 12:48:14 -0600 (Tue, 20 Jan 2009)
+ | 13 lines Merged revisions 169485 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009)
+ | 6 lines Don't play audio to the channel if we've masqueraded
+ (closes issue #14066) Reported by: bluefox Tested by:
+ otherwiseguy, bluefox ........ ................
+
+2009-01-19 20:10 +0000 [r169368] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /, apps/app_userevent.c: Merged revisions 169365
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r169365 | tilghman | 2009-01-19 14:05:52 -0600
+ (Mon, 19 Jan 2009) | 11 lines Merged revisions 169364 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009)
+ | 4 lines Truncate userevents at the end of a line, when the
+ command exceeds the buffer. (closes issue #14278) Reported by:
+ fnordian ........ ................
+
+2009-01-19 15:55 +0000 [r169213] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 169211 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r169211 | mmichelson | 2009-01-19 09:54:06 -0600
+ (Mon, 19 Jan 2009) | 21 lines Merged revisions 169210 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan
+ 2009) | 13 lines Prevent a crash in chan_local due to a potential
+ NULL pointer dereference Move the check for if both channels on a
+ local_pvt have generators to below where p->chan is checked for
+ NULLity (NULLness?). This prevents a crash from occurring if
+ p->chan is NULL. (closes issue #14189) Reported by: sascha
+ Patches: 14189.patch uploaded by putnopvut (license 60) Tested
+ by: sascha ........ ................
+
+2009-01-17 18:46 +0000 [r169154] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
+ discriminator for when ring pulse alert signal is used to preface
+ MWI spills This prevents the situation when MWI messages are
+ added to caller ID spills causing the channel to be hung up
+
+2009-01-17 01:59 +0000 [r168981-169082] Terry Wilson <twilson@digium.com>
+
+ * main/tcptls.c, /, main/http.c, include/asterisk/tcptls.h: Merged
+ revisions 169080 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169080 |
+ twilson | 2009-01-16 19:56:36 -0600 (Fri, 16 Jan 2009) | 8 lines
+ Fix qualify for TCP peer (closes issue #14192) Reported by:
+ pabelanger Patches: asterisk-bug14192.diff.txt uploaded by
+ jamesgolovich (license 176) Tested by: jamesgolovich ........
+
+ * /, channels/chan_sip.c: Merged revisions 169044 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169044 |
+ twilson | 2009-01-16 18:03:39 -0600 (Fri, 16 Jan 2009) | 8 lines
+ Fix port :0 added to SIP INVITE URI when outboundproxy used
+ (closes issue #14233) Reported by: chris-mac Patches:
+ asterisk-bug14233.diff.txt uploaded by jamesgolovich (license
+ 176) Tested by: jamesgolovich, chris-mac, otherwiseguy ........
+
+ * /, main/features.c: Merged revisions 168941 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168941 | twilson | 2009-01-16 16:16:23 -0600 (Fri, 16 Jan 2009)
+ | 19 lines Merged revisions 168716 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009)
+ | 12 lines Convert call to park_call_full to
+ masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
+ return value, we need to use masqueraded parking, otherwise we
+ will try to call ast_hangup() in __pbx_run() and in
+ do_parking_thread() and then promptly crash. (closes issue
+ #14215) Reported by: waverly360 Tested by: otherwiseguy (closes
+ issue #14228) Reported by: kobaz Tested by: otherwiseguy ........
+ ................
+
+2009-01-16 22:46 +0000 [r168979] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168976 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168976 | mmichelson | 2009-01-16 16:43:09 -0600 (Fri, 16 Jan
+ 2009) | 26 lines Merged revisions 168975 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan
+ 2009) | 18 lines Account for possible NULL pointer when we
+ receive a 408 in response to a REGISTER It may be that by the
+ time we receive a reply to a REGISTER request, the attempt has
+ timed out and thus the registry structure pointed to by the
+ corresponding sip_pvt has gone away. This situation was handled
+ properly for a 200 OK response, but the 408 case assumed that the
+ sip_registry struct was non-NULL, thus potentially causing a
+ crash This commit fixes this assumption and prints out a message
+ to the console if we should receive a late 408 response to a
+ REGISTER (closes issue #14211) Reported by: aborghi Patches:
+ 14211.diff uploaded by putnopvut (license 60) Tested by: aborghi
+ ........ ................
+
+2009-01-16 18:55 +0000 [r168836] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/say.h, apps/app_voicemail.c, /, main/say.c:
+ Merged revisions 168832 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168832 | tilghman | 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009)
+ | 13 lines Merged revisions 168828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009)
+ | 6 lines Fix the conjugation of Russian and Ukrainian languages.
+ (related to issue #12475) Reported by: chappell Patches:
+ vm_multilang.patch uploaded by chappell (license 8) ........
+ ................
+
+2009-01-16 00:47 +0000 [r168739-168748] Steve Murphy <murf@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 168746 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168746 | murf | 2009-01-15 17:34:31 -0700 (Thu, 15 Jan 2009) |
+ 20 lines Merged revisions 168745 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) |
+ 14 lines This patch fixes a problem where a goto (or jump, in
+ this case) fails a consistency check because it can't find a
+ matching extension. The problem was a missing instruction to end
+ the range notation in the code where it converts the pattern into
+ a regex and uses the regex code to determine the match. I tested
+ using the AEL code the user supplied, and now, the consistency
+ check passes. (closes issue #14141) Reported by: dimas ........
+ ................
+
+ * main/ast_expr2.c, /, main/ast_expr2.h, main/ast_expr2.y: Merged
+ revisions 168737 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168737 |
+ murf | 2009-01-15 13:54:59 -0700 (Thu, 15 Jan 2009) | 16 lines
+ This patch allows null args in ast_expr2 func calls, and fixes
+ commas being converted to pipes, which was 1.4 type stuff. If the
+ user says count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld);
+ then it won't complain about the empty arg (c,,...) and fabled's
+ patch won't let it swap the commas for pipes. Ran it thru my
+ dialplan and no complaints. (closes issue #14169) Reported by:
+ fabled Patches: function-argument-separator-fix.diff uploaded by
+ fabled (license 448) ........
+
+2009-01-15 19:17 +0000 [r168729] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Merged revisions 168728 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168728 |
+ mmichelson | 2009-01-15 13:16:29 -0600 (Thu, 15 Jan 2009) | 3
+ lines Fix the compactheaders option in sip.conf ........
+
+2009-01-15 19:05 +0000 [r168727] Olle Johansson <oej@edvina.net>
+
+ * /, configs/extconfig.conf.sample: Merged revisions 168722 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r168722 | oej | 2009-01-15 19:47:14 +0100 (Tor,
+ 15 Jan 2009) | 10 lines Merged revisions 168721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2
+ lines Meetme actually has realtime but wasn't documented ........
+ ................
+
+2009-01-15 19:00 +0000 [r168726] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168725 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168725 |
+ mmichelson | 2009-01-15 13:00:06 -0600 (Thu, 15 Jan 2009) | 17
+ lines Remove an unneeded condition for line addition to a SIP
+ request/response In Asterisk 1.4 and 1.6.0, the sip_request
+ structure had a statically allocated buffer to hold the text of
+ the request. There was a check in the add_line function to not
+ attempt to write the line into the buffer if we did not have room
+ for it. In trunk and Asterisk versions starting with 1.6.1, an
+ expandable ast_str structure is used to hold the text. Since it
+ may grow to fit an arbitrarily sized string, this check in
+ add_line is no longer valid. I found this oddity while attempting
+ to fix issue #14220; however, I do not believe that this is the
+ fix for that issue since the output supplied by the reporter did
+ not contain the warning message that would be printed had this
+ condition been satisfied. ........
+
+2009-01-15 18:20 +0000 [r168714-168715] Olle Johansson <oej@edvina.net>
+
+ * /, configs/sip.conf.sample: Merged revisions 168711 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r168711 | oej | 2009-01-15 18:55:53 +0100 (Tor, 15 Jan 2009) | 4
+ lines Clarify some misunderstandings and make it even more clear
+ that you can refer to a peer in the register= line. ........
+
+ * /, channels/chan_sip.c: Merged revisions 168712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168712 |
+ oej | 2009-01-15 19:08:59 +0100 (Tor, 15 Jan 2009) | 3 lines Make
+ sure that we have the same terminology in sip.conf.sample and the
+ source code warning. Thanks Nick Lewis for pointing this out in
+ the bug tracker. ........
+
+2009-01-15 15:37 +0000 [r168707] Sean Bright <sean.bright@gmail.com>
+
+ * /, apps/app_meetme.c: Merged revisions 168705 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168705 |
+ seanbright | 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) | 11
+ lines Add a missing unlock and properly handle the 'maxusers'
+ setting on MeetMe conferences. We were using the 'user number'
+ field to compare against the maximum allowed users, which works
+ assuming users with lower user numbers didn't leave the
+ conference. (closes issue #14117) Reported by: sergedevorop
+ Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright
+ (license 71) Tested by: sergedevorop ........
+
+2009-01-15 00:15 +0000 [r168631] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 168629 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168629 | mmichelson | 2009-01-14 18:14:17 -0600 (Wed, 14 Jan
+ 2009) | 24 lines Merged revisions 168628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan
+ 2009) | 16 lines Fix some crashes from bad datastore handling in
+ app_queue.c * The queue_transfer_fixup function was searching for
+ and removing the datastore from the incorrect channel, so this
+ was fixed. * Most datastore operations regarding the
+ queue_transfer datastore were being done without the channel
+ locked, so proper channel locking was added, too. (closes issue
+ #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
+ putnopvut (license 60) Tested by: ZX81, festr ........
+ ................
+
+2009-01-14 21:55 +0000 [r168625] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, /: Merged revisions 168623 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r168623 | rmudgett | 2009-01-14 15:51:06 -0600
+ (Wed, 14 Jan 2009) | 11 lines Merged revisions 168622 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009)
+ | 4 lines * Fixed create_process() allocation of process ID
+ values. The allocated process IDs could overflow their respective
+ NT and TE fields. Affects outgoing calls. ........
+ ................
+
+2009-01-14 21:30 +0000 [r168621] Steve Murphy <murf@digium.com>
+
+ * /, apps/app_page.c: Merged revisions 168613 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168613 | murf | 2009-01-14 13:51:26 -0700 (Wed, 14 Jan 2009) | 9
+ lines Merged revisions 168608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1
+ line app_page was failing to compile in dev-mode on my gcc-4.2.4
+ system. This change gets rid of the warning. ........
+ ................
+
+2009-01-14 21:00 +0000 [r168618] Sean Bright <sean.bright@gmail.com>
+
+ * contrib/scripts/autosupport, /: Merged revisions 168615 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r168615 | seanbright | 2009-01-14 15:58:26 -0500
+ (Wed, 14 Jan 2009) | 16 lines Merged revisions 168614 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan
+ 2009) | 9 lines Update autosupport script to supply info for both
+ Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x
+ and trunk instead of zttest. (closes issue #14132) Reported by:
+ dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by
+ dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded
+ by dsedivec (license 638) ........ ................
+
+2009-01-14 20:18 +0000 [r168611] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168610 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168610 |
+ mmichelson | 2009-01-14 14:13:48 -0600 (Wed, 14 Jan 2009) | 9
+ lines Restore the "sip show users" and "sip show user" CLI
+ commands (closes issue #14180) Reported by: amorsen Patches:
+ sip_show_users_161v3.diff uploaded by putnopvut (license 60)
+ Tested by: blitzrage, amorsen ........
+
+2009-01-14 19:12 +0000 [r168606] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c, /: Merged revisions 168604 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168604 | tilghman | 2009-01-14 13:11:14 -0600 (Wed, 14 Jan 2009)
+ | 14 lines Merged revisions 168603 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009)
+ | 7 lines Don't read into a buffer without first checking if a
+ value is beyond the end. (closes issue #13600) Reported by: atis
+ Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
+ (license 14) Tested by: atis ........ ................
+
+2009-01-14 02:11 +0000 [r168582-168596] Terry Wilson <twilson@digium.com>
+
+ * /, apps/app_page.c: Merged revisions 168594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168594 | twilson | 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009)
+ | 27 lines Merged revisions 168593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009)
+ | 20 lines Don't overflow when paging more than 128 extensions
+ The number of available slots for calls in app_page was hardcoded
+ to 128. Proper bounds checking was not in place to enforce this
+ limit, so if more than 128 extensions were passed to the Page()
+ app, Asterisk would crash. This patch instead dynamically
+ allocates memory for the ast_dial structures and removes the
+ (non-functional) arbitrary limit. This issue would have special
+ importance to anyone who is dynamically creating the argument
+ passed to the Page application and allowing more than 128
+ extensions to be added by an outside user via some external
+ interface. The patch posted by a_villacis was slightly modified
+ for some coding guidelines and other cleanups. Thanks,
+ a_villacis! (closes issue #14217) Reported by: a_villacis
+ Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
+ uploaded by a (license 660) Tested by: otherwiseguy ........
+ ................
+
+ * /, res/res_http_post.c: Merged revisions 168588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168588 |
+ twilson | 2009-01-13 17:05:43 -0600 (Tue, 13 Jan 2009) | 5 lines
+ Fully overwrite a same-named file when uploading (closes issue
+ #14190) Reported by: timking ........
+
+ * /, channels/chan_sip.c: Merged revisions 168578 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168578 | twilson | 2009-01-13 16:22:34 -0600 (Tue, 13 Jan 2009)
+ | 14 lines Merged revisions 168551 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009)
+ | 7 lines Don't pass a value with a side effect to a macro
+ (closes issue #14176) Reported by: paraeco Patches:
+ chan_sip.c.diff uploaded by paraeco (license 658) ........
+ ................
+
+2009-01-13 19:35 +0000 [r168565] Russell Bryant <russell@digium.com>
+
+ * main/indications.c, main/channel.c, channels/chan_misdn.c,
+ channels/chan_skinny.c, funcs/func_channel.c, main/app.c,
+ res/snmp/agent.c, res/res_indications.c, channels/chan_unistim.c,
+ main/pbx.c, apps/app_read.c, /, include/asterisk/indications.h,
+ apps/app_readexten.c, apps/app_disa.c,
+ include/asterisk/channel.h: Merged revisions 168562 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r168562 | russell | 2009-01-13 13:22:13 -0600
+ (Tue, 13 Jan 2009) | 10 lines Merged revisions 168561 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009)
+ | 2 lines Revert unnecessary indications API change from rev
+ 122314 ........ ................
+
+2009-01-13 17:52 +0000 [r168528-168549] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_logic.c: Merged revisions 168547 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168547 | tilghman | 2009-01-13 11:51:12 -0600 (Tue, 13 Jan 2009)
+ | 13 lines Merged revisions 168546 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009)
+ | 6 lines If either conditional is NULL, don't try copying it.
+ (closes issue #14226) Reported by: caspy Patches:
+ 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
+ ........ ................
+
+ * /, channels/chan_alsa.c: Merged revisions 168526 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r168526 | tilghman | 2009-01-12 17:45:51 -0600
+ (Mon, 12 Jan 2009) | 12 lines Merged revisions 167095 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31 Dec 2008)
+ | 5 lines Repeat attempts to write when we receive -EAGAIN from
+ the driver, as detailed in the ALSA sample code (see
+ http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
+ Reported by: Jerry Geis (via the -users list) Fixed by: me
+ (license 14) ........ ................
+
+2009-01-12 23:13 +0000 [r168524] Mark Michelson <mmichelson@digium.com>
+
+ * main/srv.c, /: Merged revisions 168523 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168523 |
+ mmichelson | 2009-01-12 17:12:30 -0600 (Mon, 12 Jan 2009) | 11
+ lines bump the verbosity of a message in srv.c up by one. It used
+ to be at this level prior to a large patch merge which converted
+ ast_verbose calls to ast_verb (closes issue #14221) Reported by:
+ jcovert Patches: srv.c.patch uploaded by jcovert (license 551)
+ ........
+
+2009-01-12 22:00 +0000 [r168510-168519] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 168517 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168517 | jpeeler | 2009-01-12 15:51:46 -0600 (Mon, 12 Jan 2009)
+ | 12 lines Merged revisions 168516 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009)
+ | 5 lines (closes issue #13881) Reported by: hoowa Update the app
+ CDR field for AGI commands that are not executing an application
+ via "exec". ........ ................
+
+ * /, channels/chan_agent.c: Merged revisions 168508 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r168508 | jpeeler | 2009-01-12 14:53:04 -0600
+ (Mon, 12 Jan 2009) | 15 lines Merged revisions 168507 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009)
+ | 9 lines (closes issue #12269) Reported by: IgorG Tested by:
+ denisgalvao This gits rid of the notion of an owning_app allowing
+ the request and hangup to be initiated by different threads.
+ Originating from an active agent channel requires this. The
+ implementation primarily changes __login_exec to wait on a
+ condition variable rather than a lock. Review:
+ http://reviewboard.digium.com/r/35/ ........ ................
+
+2009-01-12 17:26 +0000 [r168500] Olle Johansson <oej@edvina.net>
+
+ * /, apps/app_minivm.c: Merged revisions 168497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168497 |
+ oej | 2009-01-12 17:31:27 +0100 (MÃ¥n, 12 Jan 2009) | 2 lines
+ Better to use the proper app name ........
+
+2009-01-12 15:05 +0000 [r168488] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168486 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ ........
+
+2009-01-12 14:58 +0000 [r168484] Russell Bryant <russell@digium.com>
+
+ * /, configs/indications.conf.sample: Merged revisions 168481 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r168481 | russell | 2009-01-12 08:57:49 -0600
+ (Mon, 12 Jan 2009) | 10 lines Merged revisions 168480 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009)
+ | 2 lines s/ringdance/ringcadence/ for Bulgaria ........
+ ................
+
+2009-01-10 01:44 +0000 [r168336] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168334 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168334 |
+ tilghman | 2009-01-09 19:42:45 -0600 (Fri, 09 Jan 2009) | 2 lines
+ sizeof for a stringfield is 4. Kinda low for reconstructing a
+ field value. ........
+
+2009-01-09 23:18 +0000 [r168272] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 168270 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168270 | kpfleming | 2009-01-09 17:16:08 -0600 (Fri, 09 Jan
+ 2009) | 9 lines Merged revisions 168267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan
+ 2009) | 1 line update to use new sound file packages that include
+ license files ........ ................
+
+2009-01-09 23:12 +0000 [r168266] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 168192 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r168192 | rmudgett | 2009-01-09 15:43:30 -0600
+ (Fri, 09 Jan 2009) | 10 lines Merged revisions 168191 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009)
+ | 3 lines * Fix for JIRA AST-175/ABE-1757 * Miscellaneous doxygen
+ comments added. ........ ................
+
+2009-01-09 22:23 +0000 [r168209] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 168200 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r168200 | russell | 2009-01-09 16:21:05 -0600
+ (Fri, 09 Jan 2009) | 10 lines Merged revisions 168198 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09 Jan 2009)
+ | 2 lines Make this compile for mvanbaak ........
+ ................
+
+2009-01-09 21:57 +0000 [r168196] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168193 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168193 | mmichelson | 2009-01-09 15:53:26 -0600 (Fri, 09 Jan
+ 2009) | 21 lines Merged revisions 168128 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan
+ 2009) | 13 lines Add check_via calls to more request handlers
+ INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
+ checking the topmost Via to determine where to send the response.
+ Adding check_via calls to those request handlers solves this.
+ (closes issue #13071) Reported by: baron Patches: check_via.patch
+ uploaded by baron (license 531) Tested by: baron ........
+ ................
+
+2009-01-09 20:30 +0000 [r168157] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_phoneprov.c: Merged revisions 168142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168142 |
+ twilson | 2009-01-09 14:25:25 -0600 (Fri, 09 Jan 2009) | 7 lines
+ Don't leak memory if phoneprov.conf does not exist (closes issue
+ #14203) Reported by: jamesgolovich Patches:
+ asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich
+ (license 176) ........
+
+2009-01-09 18:42 +0000 [r168092] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 168090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168090 |
+ tilghman | 2009-01-09 12:30:55 -0600 (Fri, 09 Jan 2009) | 3 lines
+ When using ast_str with a non-ast_str-enabled API, we need to
+ update the buffer or otherwise, we cannot use ast_str_strlen().
+ ........
+
+2009-01-09 16:41 +0000 [r168015] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/logger.c: Merged revisions 168014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r168014 |
+ mnicholson | 2009-01-09 10:32:34 -0600 (Fri, 09 Jan 2009) | 5
+ lines Use ast_safe_system() in logger.c instead of system()
+ (closes issue #14194) Reported by: pabelanger ........
+
+2009-01-09 00:45 +0000 [r167972] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 167935 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r167935 |
+ twilson | 2009-01-08 18:13:12 -0600 (Thu, 08 Jan 2009) | 2 lines
+ Set peer context and exten values so MACRO_EXTEN and
+ MACRO_CONTEXT will be set ........
+
+2009-01-08 22:45 +0000 [r167836-167905] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 167894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r167894 | tilghman | 2009-01-08 16:37:20 -0600 (Thu, 08 Jan 2009)
+ | 13 lines Merged revisions 167840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009)
+ | 6 lines Don't truncate database results at 255 chars. (closes
+ issue #14069) Reported by: evandro Patches:
+ 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
+ ........ ................
+
+ * /, apps/app_minivm.c: Merged revisions 167835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r167835 |
+ tilghman | 2009-01-08 15:32:45 -0600 (Thu, 08 Jan 2009) | 6 lines
+ Textual changes, consistency in status variable naming, and other
+ minor bugs. (closes issue #13943) Reported by: Marquis Patches:
+ minivm_trunk_fixes3.patch uploaded by Marquis (license 32)
+ ........
+
+2009-01-08 17:28 +0000 [r167701-167727] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 167720 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r167720 | kpfleming | 2009-01-08 11:26:03 -0600 (Thu, 08 Jan
+ 2009) | 9 lines Merged revisions 167714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan
+ 2009) | 1 line remove an unnecessary argument to queue_request()
+ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 167700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r167700 | kpfleming | 2009-01-08 10:43:26 -0600 (Thu, 08 Jan
+ 2009) | 12 lines Merged revisions 167620 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan
+ 2009) | 5 lines When a SIP request or response arrives for a
+ dialog with an associated Asterisk channel, and the lock on that
+ channel cannot be obtained because it is held by another thread,
+ instead of dropping the request/response, queue it for later
+ processing when the channel lock becomes available.
+ http://reviewboard.digium.com/r/123/ ........ ................
+
+2009-01-08 14:30 +0000 [r167663] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/sip-friends.sql, /: Merged revisions 167662 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r167662 | lmadsen | 2009-01-08 09:27:53 -0500 (Thu, 08
+ Jan 2009) | 1 line Oops... fix the fieldname I changed yesterday
+ to be right. ........
+
+2009-01-07 22:37 +0000 [r167544-167573] Russell Bryant <russell@digium.com>
+
+ * /, main/file.c: Merged revisions 167569 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r167569 | russell | 2009-01-07 16:36:34 -0600 (Wed, 07 Jan 2009)
+ | 10 lines Merged revisions 167566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009)
+ | 2 lines Fix the last couple of places where free() was
+ improperly used directly. ........ ................
+
+ * /, main/file.c: Merged revisions 167555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r167555 | russell | 2009-01-07 16:27:23 -0600 (Wed, 07 Jan 2009)
+ | 10 lines Merged revisions 167554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009)
+ | 2 lines Don't fclose() the file early, the filestream
+ destructor will handle it. ........ ................
+
+ * /, main/file.c: Merged revisions 167546 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r167546 | russell | 2009-01-07 16:20:31 -0600 (Wed, 07 Jan 2009)
+ | 10 lines Merged revisions 167545 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009)
+ | 2 lines Only try to close the file if one was actually opened
+ ........ ................
+
+ * /, main/file.c: Merged revisions 167542 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r167542 | russell | 2009-01-07 16:05:29 -0600 (Wed, 07 Jan 2009)
+ | 12 lines Merged revisions 167541 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009)
+ | 4 lines Don't use free() directly. This caused a crash since
+ ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
+ #asterisk-dev ........ ................
+
+2009-01-07 18:32 +0000 [r167502] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_followme.c, /: Merged revisions 167478 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r167478 |
+ bweschke | 2009-01-07 13:20:31 -0500 (Wed, 07 Jan 2009) | 7 lines
+ Answer the channel if it has not already been answered and we've
+ already found a valid profile for followme. (closes issue #14140)
+ Reported by: dimas Patches: 14140.patch uploaded by dimas
+ ........
+
+2009-01-07 18:27 +0000 [r167491] Leif Madsen <lmadsen@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 167477 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r167477 | lmadsen | 2009-01-07 13:18:45 -0500 (Wed, 07
+ Jan 2009) | 8 lines Update queues.conf.sample documentation.
+ Update the queues.conf.sample documentation to mention that you
+ need to preload chan_local.so as well if you plan on using Local
+ channels for queue members, and you're preloading pbx_config.so.
+ (closes issue #14179) Reported by: CrashHD Tested by: CrashHD
+ ........
+
+2009-01-07 17:46 +0000 [r167456] Russell Bryant <russell@digium.com>
+
+ * main/indications.c, /: Merged revisions 167442 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r167442 | russell | 2009-01-07 11:35:39 -0600 (Wed, 07 Jan 2009)
+ | 12 lines Merged revisions 167432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009)
+ | 4 lines Treat an empty string the same way as a NULL country
+ argument. In passing, simplify the handling of returning a
+ default tone zone. ........ ................
+
+2009-01-07 14:41 +0000 [r167376] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/sip-friends.sql, /: Merged revisions 167373 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r167373 | lmadsen | 2009-01-07 09:26:19 -0500 (Wed, 07
+ Jan 2009) | 1 line Update the sip-friends.sql file to use the
+ non-deprecated 'defaultname' instead of 'username' and remove an
+ extra comma that would cause the script to fail as-is ........
+
+2009-01-06 21:38 +0000 [r167306] Mark Michelson <mmichelson@digium.com>
+
+ * main/db.c, /: Merged revisions 167301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r167301 | mmichelson | 2009-01-06 15:36:44 -0600 (Tue, 06 Jan
+ 2009) | 16 lines Merged revisions 167299 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan
+ 2009) | 8 lines Use the correct variable when creating the format
+ string (closes issue #14177) Reported by: nic_bellamy Patches:
+ asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
+ (license 299) ........ ................
+
+2009-01-06 21:10 +0000 [r167268] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 167265 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r167265 | tilghman | 2009-01-06 15:02:33 -0600
+ (Tue, 06 Jan 2009) | 16 lines Merged revisions 167260 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r167260 | tilghman | 2009-01-06 14:48:05 -0600
+ (Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
+ Jan 2009) | 2 lines Security fix AST-2009-001. ........
+ ................ ................
+
+2009-01-05 17:10 +0000 [r167182] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 167180 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r167180 | mmichelson | 2009-01-05 10:59:36 -0600 (Mon, 05 Jan
+ 2009) | 49 lines Merged revisions 167179 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan
+ 2009) | 41 lines A couple of changes to T.38 SDP attribute
+ handling There are some boolean attributes for T.38 such as
+ T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
+ T38FaxTranscodingJBIG. By simply being present, we should treat
+ these as a "true" value. The current code, however, was requiring
+ a 1 or 0 as the value of the attribute in order to parse it. This
+ is due to the fact that there are some T.38 endpoints and
+ gateways that also transmit this information incorrectly. This
+ patch follows the "be liberal in what you accept and strict in
+ what you send" philosophy by accepting both the correctly- and
+ incorrectly-formatted attributes, but only sending information as
+ it is supposed to be sent. It was also discovered that a
+ particular type of T.38 gateway sends some non-standard T.38 SDP
+ attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
+ it used T38MaxDatagram and T38FaxMaxRate respectively. We now
+ will properly accept these attributes as well. Note that there
+ are a lot of patches cited in the below commit message template.
+ This is because the person who submitted these patches is an
+ awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
+ issue #13976) Reported by: linulin Patches:
+ chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
+ by arcivanov (license 648)
+ chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) Tested by: arcivanov ........ ................
+
+2009-01-05 16:46 +0000 [r167178] Tilghman Lesher <tlesher@digium.com>
+
+ * /, UPGRADE-1.6.txt: Merged revisions 167176 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r167176 |
+ tilghman | 2009-01-05 10:44:47 -0600 (Mon, 05 Jan 2009) | 7 lines
+ More clearly explain that quote marks are no longer necessary.
+ (closes issue #13718) Reported by: davidw Patches:
+ 20081020__bug13718.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: blitzrage ........
+
+2008-12-31 19:38 +0000 [r166957] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 166954 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r166954 | tilghman | 2008-12-31 13:34:28 -0600
+ (Wed, 31 Dec 2008) | 12 lines Merged revisions 166953 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008)
+ | 5 lines Also inherit the musiconhold class. (Closes #14153)
+ Reported by: Jerry Geis, via the users list. Patch by: me
+ (license 14) ........ ................
+
+2008-12-30 20:57 +0000 [r166910] Terry Wilson <twilson@digium.com>
+
+ * phoneprov/polycom_line.xml, doc/realtimetext.txt, /,
+ res/res_phoneprov.c, doc/sip-retransmit.txt,
+ doc/tex/phoneprov.tex, res/res_http_post.c: Merged revisions
+ 166908 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r166908 |
+ twilson | 2008-12-30 14:50:05 -0600 (Tue, 30 Dec 2008) | 2 lines
+ Fix some svn:keywords ........
+
+2008-12-29 18:16 +0000 [r166863] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 166861 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon,
+ 29 Dec 2008) | 14 lines Update app_queue to deal with the removal
+ of AST_PBX_KEEPALIVE When placing a call to a queue which ran a
+ gosub on the member's channel, Asterisk would crash every time,
+ stemming from the fact that the member's channel was being hung
+ up unexpectedly when the Gosub completed. The necessary change
+ was pretty much copied and pasted from app_dial's similar changes
+ made last week. I also took the opportunity to change a LOG_DEBUG
+ message in app_dial to use ast_debug. I am guessing this was due
+ to a direct merge from 1.4 that was not corrected to use trunk's
+ preferred syntax. ........
+
+2008-12-29 14:52 +0000 [r166858] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Per kpfleming add a note describing why you
+ must never change the first element of peer_finding_info.
+
+2008-12-28 15:16 +0000 [r166775] Russell Bryant <russell@digium.com>
+
+ * channels/misdn_config.c, /: Merged revisions 166773 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r166773 | russell | 2008-12-28 09:15:14 -0600
+ (Sun, 28 Dec 2008) | 12 lines Merged revisions 166772 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008)
+ | 4 lines Use strncat() instead of an sprintf() in which source
+ and target buffers overlap
+ http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
+ ........ ................
+
+2008-12-24 01:15 +0000 [r166730] Steve Murphy <murf@digium.com>
+
+ * apps/app_queue.c, include/asterisk/features.h, apps/app_dial.c,
+ main/pbx.c, /, main/features.c, apps/app_macro.c,
+ include/asterisk/pbx.h: Merged revisions 166665 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk This merged from
+ trunk with no conflicts. I tested mostly the 'tired' cases, and
+ for the most part ignored the tests for reconnecting and dialing
+ in to fetch a parked call, after the first case. ................
+ r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) |
+ 153 lines Merged revisions 166093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to
+ merge this 1.4 patch into trunk, I had to resolve some conflicts
+ and wait for Russell to make some changes to res_agi. I re-ran
+ all the tests; 39 calls in all, and made fairly careful notes and
+ comparisons: I don't want this to blow up some aspect of
+ asterisk; I completely removed the KEEPALIVE from the pbx.h
+ decls. The first 3 scenarios involving feature park; feature xfer
+ to 700; hookflash park to Park() app call all behave the same,
+ don't appear to leave hung channels, and no crashes. ........
+ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) |
+ 131 lines This merges the masqpark branch into 1.4 These changes
+ eliminate the need for (and use of) the KEEPALIVE return code in
+ res_features.c; There are other places that use this result code
+ for similar purposes at a higher level, these appear to be left
+ alone in 1.4, but attacked in trunk. The reason these changes are
+ being made in 1.4, is that parking ends a channel's life, in some
+ situations, and the code in the bridge (and some other places),
+ was not checking the result code properly, and dereferencing the
+ channel pointer, which could lead to memory corruption and
+ crashes. Calling the masq_park function eliminates this danger in
+ higher levels. A series of previous commits have replaced some
+ parking calls with masq_park, but this patch puts them ALL to
+ rest, (except one, purposely left alone because a masquerade is
+ done anyway), and gets rid of the code that tests the KEEPALIVE
+ result, and the NOHANGUP_PEER result codes. While bug 13820
+ inspired this work, this patch does not solve all the problems
+ mentioned there. I have tested this patch (again) to make sure I
+ have not introduced regressions. Crashes that occurred when a
+ parked party hung up while the parking party was listening to the
+ numbers of the parking stall being assigned, is eliminated. These
+ are the cases where parking code may be activated: 1. Feature one
+ touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3.
+ Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi
+ hookflash xfer to 700) 4. Run Park via manager. The interesting
+ testing cases for parking are: I. A calls B, A parks B a. B hangs
+ up while A is getting the numbers announced. b. B hangs up after
+ A gets the announcement, but before the parking time expires c. B
+ waits, time expires, A is redialed, A answers, B and A are
+ connected, after which, B hangs up. d. C picks up B while still
+ in parking lot. II. A calls B, B parks A a. A hangs up while B is
+ getting the numbers announced. b. A hangs up after B gets the
+ announcement, but before the parking time expires c. A waits,
+ time expires, B is redialed, B answers, A and B are connected,
+ after which, A hangs up. d. C picks up A while still in parking
+ lot. Testing this throroughly involves acting all the
+ permutations of I and II, in situations 1,2,3, and 4. Since I
+ added a few more changes (ALL references to KEEPALIVE in the
+ bridge code eliimated (I missed one earlier), I retested most of
+ the above cases, and no crashes. H-extension weirdness. Current
+ h-extension execution is not completely correct for several of
+ the cases. For the case where A calls B, and A parks B, the 'h'
+ exten is run on A's channel as soon as the park is accomplished.
+ This is expected behavior. But when A calls B, and B parks A,
+ this will be current behavior: After B parks A, B is hung up by
+ the system, and the 'h' (hangup) exten gets run, but the channel
+ mentioned will be a derivative of A's... Thus, if A is DAHDI/1,
+ and B is DAHDI/2, the h-extension will be run on channel
+ Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be
+ those relating to Channel A. And, in the case where A is
+ reconnected to B after the park time expires, when both parties
+ hang up after the joyful reunion, no h-exten will be run at all.
+ In the case where C picks up A from the parking lot, when either
+ A or C hang up, the h-exten will be run for the C channel. CDR's
+ are a separate issue, and not addressed here. As to WHY this
+ strange behavior occurs, the answer lies in the procedure
+ followed to accomplish handing over the channel to the parking
+ manager thread. This procedure is called masquerading. In the
+ process, a duplicate copy of the channel is created, and most of
+ the active data is given to the new copy. The original channel
+ gets its name changed to XXX<ZOMBIE> and keeps the PBX
+ information for the sake of the original thread (preserving its
+ role as a call originator, if it had this role to begin with),
+ while the new channel is without this info and becomes a call
+ target (a "peer"). In this case, the parking lot manager thread
+ is handed the new (masqueraded) channel. It will not run an
+ h-exten on the channel if it hangs up while in the parking lot.
+ The h exten will be run on the original channel instead, in the
+ original thread, after the bridge completes. See bug 13820 for
+ our intentions as to how to clean up the h exten behavior.
+ Review: http://reviewboard.digium.com/r/29/ ........
+ ................
+
+2008-12-23 20:56 +0000 [r166698] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, /, channels/chan_sip.c, main/app.c:
+ Merged revisions 166696 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r166696 |
+ tilghman | 2008-12-23 14:47:08 -0600 (Tue, 23 Dec 2008) | 7 lines
+ Allow semicolons and extended characters in user-specified SIP
+ headers. (closes issue #14110) Reported by: gork Patches:
+ 20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: gork, putnopvut ........
+
+2008-12-23 15:20 +0000 [r166571] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 166569 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r166569 | mmichelson | 2008-12-23 09:17:54 -0600 (Tue, 23 Dec
+ 2008) | 20 lines Merged revisions 166568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec
+ 2008) | 12 lines Fix a crash resulting from a datastore with
+ inheritance but no duplicate callback The fix for this is to
+ simply set the newly created datastore's data pointer to NULL if
+ it is inherited but has no duplicate callback. (closes issue
+ #14113) Reported by: francesco_r Patches: 14113.patch uploaded by
+ putnopvut (license 60) Tested by: francesco_r ........
+ ................
+
+2008-12-23 04:34 +0000 [r166535] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 166533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r166533 | tilghman | 2008-12-22 22:32:15 -0600 (Mon, 22 Dec 2008)
+ | 11 lines Merged revisions 166509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008)
+ | 4 lines Use the integer form of condition for integer
+ comparisons. (closes issue #14127) Reported by: andrew ........
+ ................
+
+2008-12-22 23:27 +0000 [r166440-166472] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 166470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r166470 |
+ mmichelson | 2008-12-22 17:25:34 -0600 (Mon, 22 Dec 2008) | 11
+ lines Always use the value of the AGISIGHUP when running an AGI.
+ Prior to this patch, the value of AGISIGUP was not always honored
+ when set on a channel. (closes issue #13711) Reported by:
+ fmueller Patches: 13711.patch uploaded by putnopvut (license 60)
+ ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 166382 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r166382 | mmichelson | 2008-12-22 15:08:03 -0600
+ (Mon, 22 Dec 2008) | 44 lines Merged revisions 166380 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec
+ 2008) | 36 lines Fix a deadlock relating to channel locks and
+ autoservice It has been discovered that if a channel is locked
+ prior to a call to ast_autoservice_stop, then it is likely that a
+ deadlock will occur. The reason is that the call to
+ ast_autoservice_stop has a check built into it to be sure that
+ the thread running autoservice is not currently trying to
+ manipulate the channel we are about to pull out of autoservice.
+ The autoservice thread, however, cannot advance beyond where it
+ currently is, though, because it is trying to acquire the lock of
+ the channel for which autoservice is attempting to be stopped.
+ The gist of all this is that a channel MUST NOT be locked when
+ attempting to stop autoservice on the channel. In this particular
+ case, the channel was locked by a call to ast_read. A call to
+ ast_exists_extension led to autoservice being started and stopped
+ due to the existence of dialplan switches. It may be that there
+ are future commits which handle the same symptoms but in a
+ different location, but based on my looks through the code, it is
+ very rare to see a construct such as this one. (closes issue
+ #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
+ by putnopvut (license 60) Tested by: rtrauntvein Review:
+ http://reviewboard.digium.com/r/107/ ........ ................
+
+2008-12-22 21:46 +0000 [r166277-166438] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 166436 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r166436 | russell | 2008-12-22 15:45:28 -0600 (Mon, 22 Dec 2008)
+ | 2 lines Cosmetic change - don't mix struct initializer styles.
+ ........
+
+ * /, res/res_musiconhold.c: Merged revisions 166377 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r166377 | russell | 2008-12-22 14:26:48 -0600 (Mon, 22 Dec 2008)
+ | 2 lines Fix a bad typo. ........
+
+ * main/astobj2.c, /: Merged revisions 166342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r166342 |
+ russell | 2008-12-22 11:44:23 -0600 (Mon, 22 Dec 2008) | 2 lines
+ Remove some error messages. This is the default handler that is
+ valid to use. ........
+
+ * /, main/utils.c: Merged revisions 166317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r166317 | russell | 2008-12-22 11:29:10 -0600 (Mon, 22 Dec 2008)
+ | 10 lines Merged revisions 166297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008)
+ | 2 lines Fix up timeout handling in ast_carefulwrite(). ........
+ ................
+
+ * include/asterisk/utils.h, main/manager.c, /, main/utils.c: Merged
+ revisions 166282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r166282 |
+ russell | 2008-12-22 11:09:36 -0600 (Mon, 22 Dec 2008) | 12 lines
+ Introduce ast_careful_fwrite() and use in AMI to prevent partial
+ writes. This patch introduces a function to do careful writes on
+ a file stream which will handle timeouts and partial writes. It
+ is currently used in AMI to address the issue that has been
+ reported. However, there are probably a few other places where
+ this could be used. (closes issue #13546) Reported by: srt Tested
+ by: russell http://reviewboard.digium.com/r/104/ ........
+
+ * /, res/res_musiconhold.c: Merged revisions 166273 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r166273 | russell | 2008-12-22 10:10:40 -0600 (Mon, 22 Dec 2008)
+ | 7 lines Re-work ref count handling of MoH classes using astobj2
+ to resolve crashes. (closes issue #13566) Reported by:
+ igorcarneiro Tested by: russell Review:
+ http://reviewboard.digium.com/r/106/ ........
+
+2008-12-22 16:17 +0000 [r166275] Mark Michelson <mmichelson@digium.com>
+
+ * /, funcs/func_timeout.c, main/file.c: Merged revisions 166267 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r166267 | mmichelson | 2008-12-22 10:07:59 -0600 (Mon,
+ 22 Dec 2008) | 17 lines Fix a file playback crash and explicitly
+ initialize values in func_timeout.c A crash was brought up on the
+ bugtracker. The first run through valgrind was full of legitimate
+ complaints of uninitialized values in func_timeout when setting a
+ response timeout. These were fixed but the crash persisted. A
+ second run through showed the real problem. The reference
+ counting used for filestreams was incorrect because there were
+ some missing increments when a frame was read from a format
+ module. (closes issue #14118) Reported by: blitzrage Patches:
+ 14118v2.patch uploaded by putnopvut (license 60) Tested by:
+ blitzrage ........
+
+2008-12-22 16:10 +0000 [r166272] Joshua Colp <jcolp@digium.com>
+
+ * main/dnsmgr.c, /: Merged revisions 166268 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r166268 |
+ file | 2008-12-22 12:08:13 -0400 (Mon, 22 Dec 2008) | 7 lines
+ Record the previous port in the temporary address structure so
+ that the comparison does not treat the host as having changed
+ even if it did not. This would have been uninitialized before and
+ would have led to a baddddd port. (closes issue #13628) Reported
+ by: pananix Patches: bug13628.patch uploaded by jpeeler (license
+ 325) Tested by: file, blitzrage ........
+
+2008-12-22 14:19 +0000 [r166260] Russell Bryant <russell@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 166258 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r166258 |
+ russell | 2008-12-22 08:16:54 -0600 (Mon, 22 Dec 2008) | 26 lines
+ Remove AST_PBX_KEEPALIVE usage from res_agi. This patch removes
+ the usage of AST_PBX_KEEPALIVE from res_agi. The only usage was
+ for the AGI command, "asyncagi break". This patch removes this
+ feature. Normally, a feature would not be removed like this.
+ However, this code is broken and usage of it will result in a
+ memory leak. Usage of this feature will make the AGI code return
+ a result of AST_PBX_KEEPALIVE. The PBX handler assumes that
+ another thread has assumed ownership of the channel. The channel
+ thread will exit without destroying the channel. Unfortunately,
+ _no_ thread has ownership of the channel at this point. There are
+ a couple of serious problems here: 1) The only way to recover the
+ caller is to issue a channel redirect. This will work, but this
+ will be done with a masquerade, and the old ast_channel structure
+ will be lost. 2) Until the channel redirect happens, there is no
+ code servicing the channel. That means nothing is reading audio
+ or handling events coming from the channel. This is very bad. The
+ recommended way to get this same "break" functionality is to
+ issue the redirect while the channel is still being handled by
+ the AGI code. That way, there will be no memory leak, and there
+ will be no period of time that the channel is not being serviced.
+ ........
+
+2008-12-19 23:45 +0000 [r166098-166164] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/audiohook.c: Merged revisions 166162 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r166162 |
+ mmichelson | 2008-12-19 17:45:00 -0600 (Fri, 19 Dec 2008) | 6
+ lines Get rid of an extra space. I don't know how this crept back
+ in when I had already fixed it earlier ........
+
+ * funcs/func_audiohookinherit.c: Switch documentation formats for
+ func_audiohookinherit.c 1.6.1 does not have xml documentation, so
+ I reverted to the old way here.
+
+ * main/channel.c, funcs/func_audiohookinherit.c (added), /,
+ include/asterisk/audiohook.h, main/audiohook.c: Merged revisions
+ 166092,166095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r166092 |
+ mmichelson | 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) | 28
+ lines Adding a new dialplan function AUDIOHOOK_INHERIT This
+ function is being added as a method to allow for an audiohook to
+ move to a new channel during a channel masquerade. The most
+ obvious use for such a facility is for MixMonitor when a transfer
+ is performed. Prior to the addition of this functionality, if a
+ channel running MixMonitor was transferred by another party, then
+ the recording would stop once the transfer had completed. By
+ using AUDIOHOOK_INHERIT, you can make MixMonitor continue
+ recording the call even after the transfer has completed. It has
+ also been determined that since this is seen by most as a bug fix
+ and is not an invasive change, this functionality will also be
+ backported to 1.4 and merged into the 1.6.0 branches, even though
+ they are feature-frozen. (closes issue #13538) Reported by: mbit
+ Patches: 13538.patch uploaded by putnopvut (license 60) Tested
+ by: putnopvut Review: http://reviewboard.digium.com/r/102/
+ ........ r166095 | mmichelson | 2008-12-19 16:40:57 -0600 (Fri,
+ 19 Dec 2008) | 5 lines Remove the verbatim tag from the author
+ line I could have sworn I already did that before, though...
+ ........
+
+2008-12-19 15:08 +0000 [r165892] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 165890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r165890 | russell | 2008-12-19 09:05:09 -0600 (Fri, 19 Dec 2008)
+ | 17 lines Merged revisions 165889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008)
+ | 9 lines Ensure that the chanspy datastore is fully initialized.
+ This patch resolved some random crash issues observed by a user
+ on a BSD system (closes issue #14111) Reported by: ys Patches:
+ app_chanspy.c.diff uploaded by ys (license 281) ........
+ ................
+
+2008-12-18 Leif Madsen <leif@digium.com>
+
+ * Asterisk 1.6.1-beta4 released.
+
+2008-12-18 21:57 +0000 [r165808] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 165797 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r165797 | tilghman | 2008-12-18 15:41:02 -0600
+ (Thu, 18 Dec 2008) | 15 lines Merged revisions 165767 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008)
+ | 8 lines Add mutexes around accesses to the IMAP library
+ interface. This prevents certain crashes, especially when shared
+ mailboxes are used. (closes issue #13653) Reported by:
+ howardwilkinson Patches:
+ asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
+ howardwilkinson (license 590) Tested by: jpeeler ........
+ ................
+
+2008-12-18 21:47 +0000 [r165804] Russell Bryant <russell@digium.com>
+
+ * /, main/utils.c: Merged revisions 165801 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r165801 | russell | 2008-12-18 15:44:47 -0600 (Thu, 18 Dec 2008)
+ | 19 lines Merged revisions 165796 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008)
+ | 11 lines Make ast_carefulwrite() be more careful. This patch
+ handles some additional cases that could result in partial writes
+ to the file description. This was done to address complaints
+ about partial writes on AMI. (issue #13546) (more changes needed
+ to address potential problems in 1.6) Reported by: srt Tested by:
+ russell Review: http://reviewboard.digium.com/r/99/ ........
+ ................
+
+2008-12-18 21:24 +0000 [r165794] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_queue.c, channels/chan_oss.c, channels/chan_dahdi.c,
+ channels/chan_misdn.c, /, channels/chan_sip.c, pbx/pbx_ael.c:
+ Merged revisions 165792 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165792 |
+ file | 2008-12-18 17:21:44 -0400 (Thu, 18 Dec 2008) | 6 lines
+ Numerous documentation updates. (closes issue #13970) Reported
+ by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt
+ uploaded by blitzrage (license 10) ........
+
+2008-12-18 19:45 +0000 [r165728] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, /, include/asterisk/pbx.h: Merged
+ revisions 165723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165723 |
+ russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines
+ Remove the need for AST_PBX_KEEPALIVE with the GoSub option from
+ Dial. This is part of an effort to completely remove
+ AST_PBX_KEEPALIVE and other similar return codes from the source.
+ While this usage was perfectly safe, there are others that are
+ problematic. Since we know ahead of time that we do not want to
+ PBX to destroy the channel, the PBX API has been changed so that
+ information can be provided as an argument, instead, thus
+ removing the need for the KEEPALIVE return value. Further changes
+ to get rid of KEEPALIVE and related code is being done by murf.
+ There is a patch up for that on review 29. Review:
+ http://reviewboard.digium.com/r/98/ ........
+
+2008-12-18 19:36 +0000 [r165725] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_odbc.c, /: Merged revisions 165724 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165724 |
+ mmichelson | 2008-12-18 13:34:33 -0600 (Thu, 18 Dec 2008) | 8
+ lines Fix crashes in res_odbc. The variable "class" was being set
+ NULL just prior to being dereferenced in an ao2_link call. I have
+ moved the setting of the variable to NULL until after the
+ ao2_link call. ........
+
+2008-12-18 18:58 +0000 [r165664] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 165662 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r165662 | russell | 2008-12-18 12:54:47 -0600
+ (Thu, 18 Dec 2008) | 15 lines Merged revisions 165661 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008)
+ | 7 lines Set the process group ID on the MOH process so that all
+ children will get killed (closes issue #14099) Reported by: caspy
+ Patches: res_musiconhold.c.patch.killpg.try2 uploaded by caspy
+ (license 645) ........ ................
+
+2008-12-18 18:47 +0000 [r165660] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 165658 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r165658 | tilghman | 2008-12-18 12:36:48 -0600 (Thu, 18 Dec 2008)
+ | 2 lines Fix 2 resource leaks and fix another pipe-to-comma
+ conversion ........
+
+2008-12-18 17:59 +0000 [r165605-165606] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merge in changes to return chan_sip to
+ matching based on how it was previously done and how it is done
+ in trunk. It will do name based for users and friends and IP
+ based for peers. (closes issue #14107) Reported by: jsmith
+
+ * main/rtp.c, /: Merged revisions 165599 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r165599 | file | 2008-12-18 13:13:32 -0400 (Thu, 18 Dec 2008) |
+ 11 lines Merged revisions 165591 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4
+ lines Only care about a compatible codec for early bridging if we
+ are actually bridging to another channel. If we are not we
+ actually want to bring the audio back to us. (closes issue
+ #13545) Reported by: davidw ........ ................
+
+2008-12-18 16:48 +0000 [r165543] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c, /: Merged revisions 165541 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165541 |
+ tilghman | 2008-12-18 10:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines
+ Fix reference counts of the class and add an assertion to the
+ end. ........
+
+2008-12-17 21:48 +0000 [r165332] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_odbc.c, /: Merged revisions 165330 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165330 |
+ mmichelson | 2008-12-17 15:46:19 -0600 (Wed, 17 Dec 2008) | 3
+ lines Fix a refcount leak in res_odbc ........
+
+2008-12-17 21:31 +0000 [r165329] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_macro.c: Merged revisions 165325 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165325 |
+ tilghman | 2008-12-17 15:28:51 -0600 (Wed, 17 Dec 2008) | 2 lines
+ Oops, broke trunk ........
+
+2008-12-17 21:25 +0000 [r165324] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_directory.c, apps/app_queue.c, apps/app_voicemail.c, /,
+ res/res_realtime.c: Merged revisions 165318 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r165318 | mmichelson | 2008-12-17 15:17:20 -0600 (Wed, 17 Dec
+ 2008) | 15 lines Merged revisions 165255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec
+ 2008) | 7 lines Fix some memory leaks found while looking at how
+ realtime configs are handled. Also cleaned up some coding
+ guidelines violations in app_realtime.c, mostly related to
+ spacing ........ ................
+
+2008-12-17 21:22 +0000 [r165323] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_macro.c: Merged revisions 165319 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r165319 | tilghman | 2008-12-17 15:18:57 -0600 (Wed, 17 Dec 2008)
+ | 11 lines Merged revisions 165317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008)
+ | 4 lines Reverse the fix from issue #6176 and add proper
+ handling for that issue. (Closes issue #13962, closes issue
+ #13363) Fixed by myself (license 14) ........ ................
+
+2008-12-17 21:02 +0000 [r165279] Steve Murphy <murf@digium.com>
+
+ * /, utils/extconf.c: Merged revisions 165254 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165254 |
+ murf | 2008-12-17 13:50:19 -0700 (Wed, 17 Dec 2008) | 5 lines
+ This patch is here committed to satisfy the buildbot, who has a
+ problem with the const. ........
+
+2008-12-17 20:02 +0000 [r165242] Terry Wilson <twilson@digium.com>
+
+ * /, res/res_phoneprov.c: Merged revisions 165219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165219 |
+ twilson | 2008-12-17 13:55:10 -0600 (Wed, 17 Dec 2008) | 2 lines
+ Polycom phones close the connection after reading a little bit of
+ the firmware files, we should stop sending in that case. Also,
+ make that case print out a debug statement instead of a scary
+ WARNING. ........
+
+2008-12-17 19:54 +0000 [r165218] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 165216 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165216 |
+ file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines
+ Call proxy_update so that the IP address gets populated. Sending
+ stuff to 0.0.0.0 is silly! (closes issue #14055) Reported by:
+ chris-mac ........
+
+2008-12-17 17:56 +0000 [r165146] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 165142-165143 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r165142 | mmichelson | 2008-12-17 11:52:50 -0600 (Wed,
+ 17 Dec 2008) | 10 lines Use the create_vm_state_from_user
+ function in a place where it was not being used before. Also,
+ I've moved the urgent folder check in messagecount() up a bit so
+ that the flow is a bit better. This was something I noticed while
+ taking a look at issue #13973, although I don't think this is the
+ underlying cause of the issue. ........ r165143 | mmichelson |
+ 2008-12-17 11:53:37 -0600 (Wed, 17 Dec 2008) | 3 lines And
+ actually assign the function to a pointer... ........
+
+2008-12-17 05:53 +0000 [r165093] Steve Murphy <murf@digium.com>
+
+ * utils/conf2ael.c, pbx/ael/ael-test/ref.ael-vtest13,
+ utils/check_expr.c, utils/Makefile,
+ pbx/ael/ael-test/ref.ael-vtest17, /, pbx/pbx_ael.c,
+ utils/ael_main.c, utils/extconf.c: Merged revisions 165071 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk I
+ might add here that in I tested the merged fixes from trunk in
+ both 1.6.0 and 1.6.1 via both 'make' and ./runtests in the ael
+ regression tests for all but DEBUG_CHANNEL_LOCKS,
+ DEBUG_SCHEDULER, and CHANNEL_TRACE options. ........ r165071 |
+ murf | 2008-12-16 22:04:56 -0700 (Tue, 16 Dec 2008) | 31 lines A
+ possibly "horrible fix" for a "horribly broken" situation. As
+ stuff shifts around in the asterisk code, the miscellaneous
+ inclusions from the standalone stuff gets broken. There's no easy
+ fix for this situation. I made sure that everything in utils
+ builds without problem ***AND*** that aelparse runs the
+ regressions correctly with the following make menuselect options
+ both on and off: DONT_OPTIMIZE DEBUG_THREADS DEBUG_CHANNEL_LOCKS
+ MALLOC_DEBUG MTX_PROFILE DEBUG_SCHEDULER DEBUG_THREADLOCALS
+ DETECT_DEADLOCKS CHANNEL_TRACE I think from now on, I'm going to
+ #undef all these features in the various utils native files; I
+ guess I could do the same for the copied-in files, surrounded by
+ STANDALONE ifdef. A standalone isn't going to care about threads,
+ mutexes, etc. ........
+
+2008-12-16 23:07 +0000 [r164980] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 164978 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r164978 | mmichelson | 2008-12-16 17:06:04 -0600 (Tue, 16 Dec
+ 2008) | 15 lines Merged revisions 164977 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec
+ 2008) | 7 lines After looking through SIP registration code most
+ of the day, this is one of the few things I could find that was
+ just plain wrong. Even though it probably isn't possible for it
+ to happen, it seems weird to have code that checks if a pointer
+ is NULL and then immediately dereferences that pointer if it was
+ NULL. ........ ................
+
+2008-12-16 22:52 +0000 [r164960] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_record.c: Merged revisions 164942 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164942 |
+ jpeeler | 2008-12-16 16:45:39 -0600 (Tue, 16 Dec 2008) | 6 lines
+ (closes issue #13669) Reported by: pj Delete file recording if
+ recording terminated from a hangup. ........
+
+2008-12-16 21:40 +0000 [r164813-164884] Russell Bryant <russell@digium.com>
+
+ * /, main/utils.c: Merged revisions 164882 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r164882 | russell | 2008-12-16 15:39:15 -0600 (Tue, 16 Dec 2008)
+ | 17 lines Merged revisions 164881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008)
+ | 9 lines Fix an issue where DEBUG_THREADS may erroneously report
+ that a thread is exiting while holding a lock. If the last lock
+ attempt was a trylock, and it failed, it will still be in the
+ list of locks so that it can be reported. (closes issue #13219)
+ Reported by: pj ........ ................
+
+ * /, apps/app_macro.c: Merged revisions 164877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008)
+ | 14 lines Merged revisions 164876 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008)
+ | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has
+ been returned. This is a bug I noticed while looking at the code
+ for app_macro. This return code means that another thread has
+ assumed ownership of the channel and it can no longer be touched.
+ (I hate this return code with a passion, by the way.) ........
+ ................
+
+ * main/manager.c, /: Merged revisions 164807 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r164807 | russell | 2008-12-16 14:41:51 -0600 (Tue, 16 Dec 2008)
+ | 17 lines Merged revisions 164806 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008)
+ | 9 lines Add "restart gracefully" to the AMI blacklist of CLI
+ commands. "module unload" was already identified as a command
+ that can not be used from the AMI. "restart gracefully"
+ effectively unloads all modules, and will run in to the same
+ problems. (closes issue #13894) Reported by: kernelsensei
+ ........ ................
+
+2008-12-16 20:18 +0000 [r164805] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /: Merged revisions 164801 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164801 |
+ murf | 2008-12-16 13:04:46 -0700 (Tue, 16 Dec 2008) | 36 lines
+ (closes issue #14076) Reported by: toc Tested by: murf OK, Well
+ this issue has had its share of flip-flopping. I found the
+ following: 1. the code in question, in ext_cmp1 in pbx.c, would
+ not allow two extensions that vary only by any dashes contained
+ within them, to be defined in the same context. 2. for input
+ dialstrings, dashes are NOT ignored. So, skipping them when
+ sorting patterns seemed a bit silly. Thus, you might declare ext
+ 891 in a context, but if you try dialing 8-9-1, it will NOT match
+ 891. So, I proposed to remove the code from ext_cmp1 to skip the
+ spaces and dashes. Just kept us from declaring 891 and 8-9-1 in
+ the same context, forcing users to generate otherwise uselessly
+ obfuscated dialplan code to get the same effect. Then, I tried
+ out 1.4, and found that: 1. you can declare 891 and 8-9-1 in the
+ same context! 2. You can't define 891, and have 8-9-1 match it!
+ Nor can you define 8-9-1, and have 891 match it! So, it appears
+ that my proposal simply restores the pbx to behaving as it did in
+ 1.4. ........
+
+2008-12-16 19:54 +0000 [r164799] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/safe_asterisk, /: Merged revisions 164798 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r164798 | tilghman | 2008-12-16 13:54:11 -0600 (Tue, 16
+ Dec 2008) | 4 lines Set up umask as a possible configuration
+ option. (closes issue #13753) Reported by: irroot ........
+
+2008-12-16 17:18 +0000 [r164677-164739] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/threadstorage.h, /, main/threadstorage.c: Merged
+ revisions 164737 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r164737 | russell | 2008-12-16 11:14:01 -0600 (Tue, 16 Dec 2008)
+ | 22 lines Merged revisions 164736 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008)
+ | 14 lines Fix memory leak and invalid reporting issues with
+ DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was
+ being used within the context of the thread local data
+ destructors. We would go off and allocate more thread local data
+ while the pthread lib was in the middle of destroying it all.
+ This led to a memory leak. Another issue was an invalid argument
+ being provided to the the object_add API call. (closes issue
+ #13678) Reported by: ys Tested by: Russell ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 164675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008)
+ | 19 lines Merged revisions 164672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008)
+ | 11 lines Fix a memory leak related to the use of the "setvar"
+ configuration option. The problem was that these variables were
+ being appended to the list of vars on the sip_pvt every time a
+ re-registration or re-subscription came in. Since it's just a
+ waste of memory to put them there unless the request was an
+ INVITE, then the fix is to check the request type before copying
+ the vars. (closes issue #14037) Reported by: marvinek Tested by:
+ russell ........ ................
+
+2008-12-16 15:47 +0000 [r164662] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 164659 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164659 |
+ file | 2008-12-16 11:44:28 -0400 (Tue, 16 Dec 2008) | 4 lines
+ When using externhost make sure the port gets set to the bindaddr
+ port if one was not specified in the externhost value itself.
+ (closes issue #13634) Reported by: performer ........
+
+2008-12-16 15:42 +0000 [r164658] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /: Merged revisions 164648 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r164648 | murf | 2008-12-16 08:31:54 -0700 (Tue, 16 Dec 2008) |
+ 13 lines Merged revisions 164634 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5
+ lines I added a sentence to clarify why - and ' ' are ignored in
+ patterns as per bug 14076. Leif says he'll put some stuff about
+ it in the extensions.conf sample, etc. ........ ................
+
+2008-12-16 15:02 +0000 [r164521-164625] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_minivm.c: Merged revisions 164623 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164623 |
+ russell | 2008-12-16 09:00:27 -0600 (Tue, 16 Dec 2008) | 5 lines
+ Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable
+ that was not needed. (closes issue #14081) Reported by: pkempgen
+ ........
+
+ * /, res/res_musiconhold.c: Merged revisions 164606 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r164606 | russell | 2008-12-16 08:31:02 -0600
+ (Tue, 16 Dec 2008) | 13 lines Merged revisions 164605 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16 Dec 2008)
+ | 5 lines Don't try to change working directory if a directory
+ was not configured. (closes issue #14089) Reported by: caspy
+ ........ ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 164602 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r164602 | russell | 2008-12-16 08:17:45 -0600 (Tue, 16 Dec 2008)
+ | 7 lines Fix usage of the DAHDI_VMWI ioctl. (closes issue
+ #14090) Reported by: alecdavis Patches:
+ chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license
+ 585) ........
+
+ * channels/chan_iax2.c, /: Merged revisions 164525 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r164525 | russell | 2008-12-15 16:25:46 -0600 (Mon, 15 Dec 2008)
+ | 6 lines Open a timer before loading configuration so that the
+ trunking configuration option will take effect. (closes issue
+ #14082) Reported by: seandarcy ........
+
+ * channels/chan_iax2.c, /: Merged revisions 164522 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r164522 | russell | 2008-12-15 16:22:43 -0600 (Mon, 15 Dec 2008)
+ | 4 lines Fix log message to refer to the generic timing
+ interface, not DAHDI specifically (inspired by issue #14082)
+ ........
+
+ * main/frame.c, /: Merged revisions 164519 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164519 |
+ russell | 2008-12-15 15:53:30 -0600 (Mon, 15 Dec 2008) | 7 lines
+ Make sure we handle a uint32_t payload in ast_frdup() (closes
+ issue #14080) Reported by: fnordian Patches: frame.patch uploaded
+ by fnordian (license 110) ........
+
+2008-12-15 19:54 +0000 [r164421-164425] Mark Michelson <mmichelson@digium.com>
+
+ * /, include/asterisk/pbx.h: Merged revisions 164423 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r164423 | mmichelson | 2008-12-15 13:53:29 -0600
+ (Mon, 15 Dec 2008) | 11 lines Merged revisions 164422 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec
+ 2008) | 3 lines Add the deadlock note to ast_spawn_extension as
+ well ........ ................
+
+ * /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged
+ revisions 164419 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r164419 | mmichelson | 2008-12-15 13:51:24 -0600 (Mon, 15 Dec
+ 2008) | 12 lines Merged revisions 164416 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec
+ 2008) | 4 lines Add notes to autoservice and pbx doxygen
+ regarding a potential deadlock scenario so that it is avoided in
+ the future ........ ................
+
+2008-12-15 18:27 +0000 [r164355] Tilghman Lesher <tlesher@digium.com>
+
+ * /, cdr/cdr_pgsql.c: Merged revisions 164349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164349 |
+ tilghman | 2008-12-15 12:09:58 -0600 (Mon, 15 Dec 2008) | 4 lines
+ When querying for the structure of the CDR table, remove the
+ schema, if it exists. (Closes issue #14058) ........
+
+2008-12-15 18:14 +0000 [r164314-164353] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 164351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r164351 | file | 2008-12-15 14:12:24 -0400 (Mon, 15 Dec 2008) |
+ 13 lines Merged revisions 164350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6
+ lines Do not try to unlock a non-existant channel if the transfer
+ fails. (closes issue #13800) Reported by: dwagner Patches:
+ asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license
+ 608) ........ ................
+
+ * /, main/file.c: Merged revisions 164312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164312 |
+ file | 2008-12-15 13:24:28 -0400 (Mon, 15 Dec 2008) | 4 lines Use
+ ast_seekstream to return the file stream back to the beginning
+ instead of directly seeking to zero. This is because some audio
+ formats have headers at the front that need to be skipped, which
+ will be done by the format module. (closes issue #14079) Reported
+ by: elguero ........
+
+2008-12-15 16:32 +0000 [r164276-164300] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /, main/features.c: Merged revisions 164203 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r164203 | russell | 2008-12-15 08:40:24 -0600
+ (Mon, 15 Dec 2008) | 39 lines Merged revisions 164201 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008)
+ | 31 lines Handle a case where a call can be bridged to a channel
+ that is still ringing. The issue that was reported was about a
+ case where a RINGING channel got redirected to an extension to
+ pick up a call from parking. Once the parked call got taken out
+ of parking, it heard silence until the other side answered.
+ Ideally, the caller that was parked would get a ringing
+ indication. This patch fixes this case so that the caller
+ receives ringback once it comes out of parking until the other
+ side answers. The fixes are: - Make sure we remember that a
+ channel was an outgoing channel when doing a masquerade. This
+ prevents an erroneous ast_answer() call on the channel, which
+ causes a bogus 200 OK to be sent in the case of SIP. - Add some
+ additional comments to explain related parts of code. - Update
+ the handling of the ast_channel visible_indication field. Storing
+ values that are not stateful is pointless. Control frames that
+ are events or commands should be ignored. - When a bridge first
+ starts, check to see if the peer channel needs to be given
+ ringing indication because the calling side is still ringing. -
+ Rework ast_indicate_data() a bit for the sake of readability.
+ (closes issue #13747) Reported by: davidw Tested by: russell
+ Review: http://reviewboard.digium.com/r/90/ ........
+ ................
+
+ * /, pbx/pbx_dundi.c: Merged revisions 164272 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164272 |
+ russell | 2008-12-15 10:17:55 -0600 (Mon, 15 Dec 2008) | 8 lines
+ When a reload is issued, always process the configuration for
+ dundi.conf. The reason is that a reload can be used to refresh
+ DNS lookups for defined peers. Even if the config file hasn't
+ changed, we want to process it for that purpose. (closes issue
+ #13776) Reported by: kombjuder ........
+
+2008-12-15 16:18 +0000 [r164273-164274] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 164270 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164270 |
+ mmichelson | 2008-12-15 10:16:47 -0600 (Mon, 15 Dec 2008) | 4
+ lines Fix a compile warning and a logic error that could have
+ been bad for non-realtime queues ........
+
+ * apps/app_queue.c, /: Merged revisions 164268 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164268 |
+ mmichelson | 2008-12-15 10:10:43 -0600 (Mon, 15 Dec 2008) | 17
+ lines Fix up a few issues with regards to queues * Fix reference
+ counting used in the __queues_show function * Add code to be sure
+ that the "queue show" command does not print information for a
+ realtime queue which has been deleted from the backend * Add a
+ missing unref to the realtime queue loading function for the case
+ where a queue is in the module's container but has been deleted
+ from the realtime backend (closes issue #14033) Reported by:
+ cristiandimache Patches: 14033.patch uploaded by putnopvut
+ (license 60) Tested by: cristiandimache ........
+
+2008-12-15 15:50 +0000 [r164266] Joshua Colp <jcolp@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, apps/app_fax.c,
+ configure.ac: Merged revisions 164257 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r164257 |
+ file | 2008-12-15 11:41:22 -0400 (Mon, 15 Dec 2008) | 4 lines
+ Make app_fax compatible with newer versions of spandsp. This
+ remains backwards compatible with earlier versions though so do
+ not fret. (closes issue #14073) Reported by: seandarcy ........
+
+2008-12-13 01:01 +0000 [r163914] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 163912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r163912 |
+ file | 2008-12-12 20:59:24 -0400 (Fri, 12 Dec 2008) | 2 lines
+ Only detach and destroy the whisper audiohooks if they are
+ actually in use. ........
+
+2008-12-13 00:08 +0000 [r163875] Terry Wilson <twilson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 163873 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r163873 |
+ twilson | 2008-12-12 17:48:26 -0600 (Fri, 12 Dec 2008) | 6 lines
+ When using realtime queues, app_queue wasn't updating the
+ strategy if it was changed in the realtime backend. This patch
+ resolves the issue for almost all situations. It is currently not
+ supported to switch to the linear strategy via realtime since the
+ ao2_container for members will have been set to have multiple
+ buckets and therefore the members would be unordered. (closes
+ issue #14034) Reported by: cristiandimache Tested by:
+ otherwiseguy, cristiandimache ........
+
+2008-12-12 23:08 +0000 [r163830] Russell Bryant <russell@digium.com>
+
+ * /: Merged revisions 163829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ ........
+
+2008-12-12 22:05 +0000 [r163764] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, main/editline/read.c, /: Merged revisions 163762
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r163762 | tilghman | 2008-12-12 16:04:26 -0600
+ (Fri, 12 Dec 2008) | 14 lines Merged revisions 163761 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008)
+ | 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk,
+ but also add a pointer inside editline to look back to
+ asterisk.c, so others don't spend as much time as I did looking
+ (in the wrong place) for the appropriate function. Reported by:
+ ZX81, via the #asterisk-users channel Fixed by: me (license 14)
+ ........ ................
+
+2008-12-12 19:58 +0000 [r163715] Steve Murphy <murf@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 163675 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r163675 | murf | 2008-12-12 12:16:32 -0700 (Fri, 12 Dec 2008) | 1
+ line demote always-appearing debug message (for certain boards)
+ to ast_debug lev 3 msg instead ........
+
+2008-12-12 18:53 +0000 [r163656-163672] Russell Bryant <russell@digium.com>
+
+ * main/tcptls.c, /, channels/chan_sip.c: Merged revisions 163670
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12
+ Dec 2008) | 6 lines Rename a number of tcptls_session variables.
+ There are no functional changes here. The name "ser" was used in
+ a lot of places. However, it is a relic from when the struct was
+ a server_instance, not a session_instance. It was renamed since
+ it represents both a server or client connection. ........
+
+ * /, channels/chan_sip.c: Merged revisions 163667 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r163667 |
+ russell | 2008-12-12 12:33:27 -0600 (Fri, 12 Dec 2008) | 5 lines
+ Fix a small race condition in sip_tcp_locate(). We must increase
+ the reference count on the tcptls_session _before_ unlocking the
+ thread list. ........
+
+ * /, channels/chan_sip.c: Merged revisions 163642 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r163642 |
+ russell | 2008-12-12 12:19:47 -0600 (Fri, 12 Dec 2008) | 7 lines
+ Resolve crashes when using SIP TCP/TLS with qualify. The problem
+ was a reference count error on the tcptls_session structure.
+ (closes issue #13989) Reported by: Nugget ........
+
+2008-12-12 18:19 +0000 [r163640] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 163629 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r163629 |
+ file | 2008-12-12 14:17:12 -0400 (Fri, 12 Dec 2008) | 4 lines
+ When a device registers we need to unlink them (if linked) from
+ the peers_by_ip container and link them back in since their IP
+ address has changed. This would have manifested itself if you
+ configured a new device (as type=peer), registered, and then
+ tried to place a call from the device. Since the peer was not
+ linked into the peers_by_ip container it would have never been
+ found. (closes issue #13811) Reported by: pj ........
+
+2008-12-12 17:26 +0000 [r163624] Michiel van Baak <michiel@vanbaak.info>
+
+ * res/res_monitor.c, /: Merged revisions 163612 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r163612 |
+ mvanbaak | 2008-12-12 18:22:47 +0100 (Fri, 12 Dec 2008) | 7 lines
+ Document default Monitor file location. (closes issue #14065)
+ Reported by: kshumard Patches:
+ res_monitor.documentation.patch.txt uploaded by kshumard (license
+ 92) ........
+
+2008-12-12 16:57 +0000 [r163581] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /, channels/chan_sip.c: Merged revisions 163579
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec
+ 2008) | 4 lines Since chan_sip is callback devicestate driven do
+ not pass in actual states, pass in unknown so we get asked.
+ Additionally do not pass in an actual device state value in
+ ast_setstate since the channel may be callback driven. (closes
+ issue #13525) Reported by: pj ........
+
+2008-12-12 14:48 +0000 [r163514-163515] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, main/autoservice.c, /,
+ include/asterisk/channel.h: Merged revisions 163449 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r163449 | russell | 2008-12-12 07:55:30 -0600
+ (Fri, 12 Dec 2008) | 34 lines Merged revisions 163448 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008)
+ | 26 lines Resolve issues that could cause DTMF to be processed
+ out of order. These changes come from team/russell/issue_12658 1)
+ Change autoservice to put digits on the head of the channel's
+ frame readq instead of the tail. If there were frames on the
+ readq that autoservice had not yet read, the previous code would
+ have resulted in out of order processing. This required a new API
+ call to queue a frame to the head of the queue instead of the
+ tail. 2) Change up the processing of DTMF in ast_read(). Some of
+ the problems were the result of having two sources of pending
+ DTMF frames. There was the dtmfq and the more generic readq. Both
+ were used for pending DTMF in various scenarios. Simplifying
+ things to only use the frame readq avoids some of the problems.
+ 3) Fix a bug where a DTMF END frame could get passed through when
+ it shouldn't have. If code set END_DTMF_ONLY in the middle of
+ digit emulation, and a digit arrived before emulation was
+ complete, digits would get processed out of order. (closes issue
+ #12658) Reported by: dimas Tested by: russell, file Review:
+ http://reviewboard.digium.com/r/85/ ........ ................
+
+ * /, pbx/pbx_dundi.c: Merged revisions 163512 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r163512 | russell | 2008-12-12 08:44:06 -0600 (Fri, 12 Dec 2008)
+ | 13 lines Merged revisions 163511 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008)
+ | 5 lines Specify uint32_t for variables storing a CRC32 so that
+ it is actually 32 bits on 64-bit machines, as well. (inspired by
+ issue #13879) ........ ................
+
+2008-12-11 23:48 +0000 [r163386] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 163384 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r163384 | tilghman | 2008-12-11 17:38:56 -0600 (Thu, 11 Dec 2008)
+ | 16 lines Merged revisions 163383 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008)
+ | 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on
+ certain shells, the terminal is messed up. By intercepting those
+ events with a signal handler in the remote console, we can avoid
+ those issues. (closes issue #13464) Reported by: tzafrir Patches:
+ 20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: blitzrage ........ ................
+
+2008-12-11 22:52 +0000 [r163319] Matt Nicholson <mnicholson@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 163317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r163317 | mnicholson | 2008-12-11 16:49:59 -0600 (Thu, 11 Dec
+ 2008) | 16 lines Merged revisions 163316 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec
+ 2008) | 9 lines Clean up the dundi cache every 5 minutes. (closes
+ issue #13819) Reported by: adomjan Patches:
+ pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
+ dundi_clearecache3.diff uploaded by mnicholson (license 96)
+ Tested by: adomjan ........ ................
+
+2008-12-11 21:50 +0000 [r163252-163256] Russell Bryant <russell@digium.com>
+
+ * /, funcs/func_strings.c, funcs/func_cut.c: Merged revisions
+ 163254 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r163254 | russell | 2008-12-11 15:48:08 -0600 (Thu, 11 Dec 2008)
+ | 16 lines Merged revisions 163253 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008)
+ | 8 lines Fix some observed slowdowns in dialplan processing. The
+ change is to remove autoservice usage from dialplan functions
+ that do not need it because they do not perform operations that
+ potentially block. (closes issue #13940) Reported by: tbelder
+ ........ ................
+
+ * /, res/res_timing_pthread.c: Merged revisions 163241 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r163241 | russell | 2008-12-11 15:21:31 -0600 (Thu, 11 Dec 2008)
+ | 8 lines Fix a problem where continuous mode will get
+ inadvertently get turned off if set_rate() is used while
+ continuous mode was already turned on. (closes issue #13738)
+ Reported by: smurfix Patches: res.patch.fixed uploaded by smurfix
+ (license 547) ........
+
+2008-12-11 21:00 +0000 [r163214] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
+ revisions 163213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r163213 |
+ mmichelson | 2008-12-11 14:57:44 -0600 (Thu, 11 Dec 2008) | 9
+ lines Add an option to voicemail.conf to allow urgent messages to
+ be forwarded as not urgent. (closes issue #14063) Reported by:
+ jaroth Patches: urgfwd_v2.patch uploaded by jaroth (license 50)
+ ........
+
+2008-12-11 20:10 +0000 [r163173] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 163171 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r163171 |
+ russell | 2008-12-11 14:07:47 -0600 (Thu, 11 Dec 2008) | 16 lines
+ Fix the "failed" extension for outgoing calls. The conversion to
+ use ast_check_hangup() everywhere instead of checking the
+ softhangup flag directly introduced this problem. The issue is
+ that ast_check_hangup() checked for tech_pvt to be NULL.
+ Unfortunately, this will be NULL is some valid circumstances,
+ such as with a dummy channel. The fix is simple. Don't check
+ tech_pvt. It's pointless, because the code path that sets this to
+ NULL is when the channel hangup callback gets called. This
+ happens inside of ast_hangup(), which is the same function
+ responsible for freeing the channel. Any code calling
+ ast_check_hangup() better not be calling it after that point, and
+ if so, we have a bigger problem at hand. (closes issue #14035)
+ Reported by: erogoza ........
+
+2008-12-11 20:05 +0000 [r163170] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 163168 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r163168 | tilghman | 2008-12-11 14:02:35 -0600 (Thu, 11 Dec 2008)
+ | 5 lines Sometimes even Linux needs -lm to link libtonezone,
+ such as when libtonezone is compiled statically. (closes issue
+ #13887) Reported by: tzafrir ........
+
+2008-12-11 17:16 +0000 [r163100] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 163094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r163094 | russell | 2008-12-11 11:06:16 -0600 (Thu, 11 Dec 2008)
+ | 19 lines Merged revisions 163092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008)
+ | 11 lines Fix an issue that made it so you could only have a
+ single caller executing a custom feature at a time. This was
+ especially problematic when custom features ran for any
+ appreciable amount of time. The fix turned out to be quite
+ simple. The dynamic features are now stored in a read/write list
+ instead of a list using a mutex. (closes issue #13478) Reported
+ by: neutrino88 Fix suggested by file ........ ................
+
+2008-12-11 16:54 +0000 [r163091] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 163089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r163089 | tilghman | 2008-12-11 10:52:24 -0600 (Thu, 11 Dec 2008)
+ | 13 lines Merged revisions 163088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008)
+ | 6 lines Don't wait forever, if there's a specified recording
+ timeout. (closes issue #13885) Reported by: bamby Patches:
+ res_agi.c.patch uploaded by bamby (license 430) ........
+ ................
+
+2008-12-11 16:49 +0000 [r163083-163087] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 163085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r163085 | mmichelson | 2008-12-11 10:47:34 -0600 (Thu, 11 Dec
+ 2008) | 12 lines Merged revisions 163084 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec
+ 2008) | 4 lines Revert this cast to long. Using time_t here
+ causes build failures on a FreeBSD 32-bit build. ........
+ ................
+
+ * apps/app_queue.c, /: Merged revisions 163081 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r163081 | mmichelson | 2008-12-11 10:33:16 -0600 (Thu, 11 Dec
+ 2008) | 22 lines Merged revisions 163080 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec
+ 2008) | 14 lines Fix a potential crash due to unsafe datastore
+ handling. This patch also contains a conversion from using long
+ to time_t for representing times for a queue, as well as some
+ whitespace fixes. (closes issue #14060) Reported by: nivek
+ Patches: datastore_fixup.patch.corrected uploaded by nivek
+ (license 636) with slight modification from me Tested by: nivek
+ ........ ................
+
+2008-12-11 15:07 +0000 [r163006] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 162997 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r162997 |
+ file | 2008-12-11 11:05:49 -0400 (Thu, 11 Dec 2008) | 4 lines
+ When a device registers to use it is entirely possible that they
+ may be in use, so tell the core that we don't know the devstate
+ and have it ask us for it. (closes issue #13525) Reported by: pj
+ ........
+
+2008-12-10 23:13 +0000 [r162949] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 162922,162930 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r162922 |
+ tilghman | 2008-12-10 16:48:09 -0600 (Wed, 10 Dec 2008) | 7 lines
+ Checking global variables here actually overwrote the previous
+ substitution by channel variables, and in any case, was
+ redundant; pbx_substitute_variables_helper ALREADY does
+ substitution for global variables. (closes issue #13327) Reported
+ by: pj ........ r162930 | tilghman | 2008-12-10 17:01:14 -0600
+ (Wed, 10 Dec 2008) | 2 lines Previously missing line, now the
+ substitution works correctly ........
+
+2008-12-10 22:54 +0000 [r162896-162929] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 162927 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r162927 | jpeeler | 2008-12-10 16:53:34 -0600
+ (Wed, 10 Dec 2008) | 11 lines Merged revisions 162926 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10 Dec 2008)
+ | 3 lines Oops, inverted logic for a strcasecmp check. Pointed
+ out by mmichelson, thanks! ........ ................
+
+ * /, res/res_musiconhold.c: Merged revisions 162891 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r162891 | jpeeler | 2008-12-10 16:11:46 -0600
+ (Wed, 10 Dec 2008) | 13 lines Merged revisions 162874 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10 Dec 2008)
+ | 5 lines (closes issue #13229) Reported by:
+ clegall_proformatique Ensure that moh_generate does not return
+ prematurely before local_ast_moh_stop is called. Also, the sleep
+ in mp3_spawn now only occurs for http locations since it seems to
+ have been added originally only for failing media streams.
+ ........ ................
+
+2008-12-10 19:05 +0000 [r162741-162807] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 162805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162805 | file | 2008-12-10 15:02:57 -0400 (Wed, 10 Dec 2008) |
+ 13 lines Merged revisions 162804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6
+ lines Fix subscription based MWI up a bit. We only want to put
+ sip: at the beginning of the URI if it is not already there and
+ revert code to ignore destination check if subscribing for MWI.
+ (closes issue #12560) Reported by: vsauer Patches: patch001.diff
+ uploaded by ramonpeek (license 266) ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 162739 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162739 | file | 2008-12-10 13:53:09 -0400 (Wed, 10 Dec 2008) |
+ 13 lines Merged revisions 162738 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6
+ lines When a SIP peer unregisters set the expiry time back to 0
+ so that the 200 OK contains an expires of 0. (closes issue
+ #13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded
+ by hjourdain (license 583) ........ ................
+
+2008-12-10 16:39 +0000 [r162666-162669] Mark Michelson <mmichelson@digium.com>
+
+ * doc/tex/misdn.tex, /: Merged revisions 162667 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162667 | mmichelson | 2008-12-10 10:39:10 -0600 (Wed, 10 Dec
+ 2008) | 16 lines Merged revisions 162659 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec
+ 2008) | 8 lines Add missing documentation to misdn.txt (closes
+ issue #14052) Reported by: festr Patches: misdn.txt.patch
+ uploaded by festr (license 443) ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 162664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162664 | mmichelson | 2008-12-10 10:34:35 -0600 (Wed, 10 Dec
+ 2008) | 19 lines Merged revisions 162663 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec
+ 2008) | 11 lines Revert fix for issue 13570. It has caused more
+ problems than it helped to fix. (closes issue #13783) Reported
+ by: navkumar (closes issue #14025) Reported by: ffs ........
+ ................
+
+2008-12-10 16:08 +0000 [r162622-162658] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 162656 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162656 | file | 2008-12-10 12:06:59 -0400 (Wed, 10 Dec 2008) |
+ 13 lines Merged revisions 162653 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6
+ lines Increment the sequence number on the end packets for
+ RFC2833. After reading the RFC some more and doing some testing I
+ agree with this change. (closes issue #12983) Reported by: vt
+ Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
+ 520) ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 162619 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r162619 |
+ file | 2008-12-10 11:22:26 -0400 (Wed, 10 Dec 2008) | 4 lines
+ When transmitting a register set the socket port to the local one
+ for the transport being used, not the port for the remote server.
+ (closes issue #13633) Reported by: performer ........
+
+2008-12-10 11:37 +0000 [r162585] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, res/snmp/agent.c: Merged revisions 162583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r162583 |
+ mvanbaak | 2008-12-10 12:34:09 +0100 (Wed, 10 Dec 2008) | 5 lines
+ Make res_snmp.so compile on OpenBSD. OpenBSD uses an old version
+ of gcc which throws an error if you use a macro that's not
+ #defined ........
+
+2008-12-09 23:45 +0000 [r162490] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/stringfields.h, /: Merged revisions 162488 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r162488 | kpfleming | 2008-12-09 17:41:02 -0600 (Tue, 09
+ Dec 2008) | 1 line it does help if the compiler attribute syntax
+ is correct ........
+
+2008-12-09 23:12 +0000 [r162472] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 162466 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r162466 | tilghman | 2008-12-09 17:10:34 -0600
+ (Tue, 09 Dec 2008) | 9 lines Merged revisions 162463 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09
+ Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........
+ ................
+
+2008-12-09 22:34 +0000 [r162416] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, include/asterisk/utils.h, /, main/utils.c:
+ Merged revisions 162414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162414 | russell | 2008-12-09 16:25:06 -0600 (Tue, 09 Dec 2008)
+ | 16 lines Merged revisions 162413 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008)
+ | 8 lines Remove the test_for_thread_safety() function
+ completely. The test is not valid. Besides, if we actually
+ suspected that recursive mutexes were not working, we would get a
+ ton of LOG_ERROR messages when DEBUG_THREADS is turned on.
+ (inspired by a discussion on the asterisk-dev list) ........
+ ................
+
+2008-12-09 22:02 +0000 [r162372] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 162355 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r162355 | tilghman | 2008-12-09 15:57:09 -0600
+ (Tue, 09 Dec 2008) | 11 lines Merged revisions 162348 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008)
+ | 4 lines We appear to have documented tz= in the [general]
+ section of voicemail.conf, without actually having implemented
+ it. Oops. (Reported by Olivier on the -users list) ........
+ ................
+
+2008-12-09 21:18 +0000 [r162344] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 162342 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r162342 | file | 2008-12-09 17:16:37 -0400 (Tue,
+ 09 Dec 2008) | 11 lines Merged revisions 162341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4
+ lines Add 'down' as a valid state for directed call pickup. This
+ creeps up when we receive session progress when dialing a device
+ and not ringing. (closes issue #14005) Reported by: ddl ........
+ ................
+
+2008-12-09 21:03 +0000 [r162302] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 162291 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162291 | russell | 2008-12-09 14:59:54 -0600 (Tue, 09 Dec 2008)
+ | 17 lines Merged revisions 162286 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008)
+ | 9 lines Fix an issue where callers on an incoming call on an
+ SLA trunk would not hear ringback. We need to make sure that we
+ don't start writing audio to the trunk channel until we're
+ actually ready to answer it. Otherwise, the channel driver will
+ treat it as inband progress, even though all they are getting is
+ silence. (closes issue #12471) Reported by: mthomasslo ........
+ ................
+
+2008-12-09 20:48 +0000 [r162278] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 162275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162275 | file | 2008-12-09 16:46:11 -0400 (Tue, 09 Dec 2008) |
+ 11 lines Merged revisions 162273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4
+ lines Fix double declaration of 'x' on the PPC platform. (closes
+ issue #14038) Reported by: ffloimair ........ ................
+
+2008-12-09 20:47 +0000 [r162277] Steve Murphy <murf@digium.com>
+
+ * res/ael/ael.flex, /, res/ael/ael_lex.c: Merged revisions 162271
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r162271 | murf | 2008-12-09 13:40:31 -0700 (Tue,
+ 09 Dec 2008) | 9 lines Merged revisions 162264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1
+ line In discussion with seanbright on #asterisk-dev, I have added
+ a default rule, and an option to suppress the default rule from
+ being generated in the flex output, for the sake of those OS's
+ where they didn't tweak flex's ECHO macro, and the compiler
+ doesn't like it. The regressions are OK with this. ........
+ ................
+
+2008-12-09 20:31 +0000 [r162269] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c, /: Merged revisions 162266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162266 | mmichelson | 2008-12-09 14:30:07 -0600 (Tue, 09 Dec
+ 2008) | 14 lines Merged revisions 162265 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec
+ 2008) | 6 lines If we fail to start a thread for the pbx to run
+ in, we need to be sure to decrease the number of active calls on
+ the system. This fix may relate to ABE-1713, but it is not
+ certain yet. ........ ................
+
+2008-12-09 19:52 +0000 [r162202-162207] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 162205 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162205 | file | 2008-12-09 15:48:35 -0400 (Tue, 09 Dec 2008) |
+ 14 lines Merged revisions 162204 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7
+ lines Make sure that the timestamp for DTMF is not the same as
+ the previous voice frame and do not send audio when transmitting
+ DTMF as this confuses some equipment. (closes issue #13209)
+ Reported by: ip-rob Patches: 13209.diff uploaded by file (license
+ 11) Tested by: ip-rob, bujones ........ ................
+
+ * main/rtp.c, /: Merged revisions 162197 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162197 | file | 2008-12-09 15:08:39 -0400 (Tue, 09 Dec 2008) |
+ 11 lines Merged revisions 162188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4
+ lines Take video into account when early bridging RTP. (closes
+ issue #13535) Reported by: davidw ........ ................
+
+2008-12-09 18:49 +0000 [r162082-162142] Steve Murphy <murf@digium.com>
+
+ * res/ael/ael.flex, /, res/ael/ael_lex.c: Merged revisions 162140
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r162140 | murf | 2008-12-09 11:35:35 -0700 (Tue,
+ 09 Dec 2008) | 9 lines Merged revisions 162136 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1
+ line Previous fix used ast_malloc and ast_copy_string and messed
+ up the standalone stuff. Fixed. ........ ................
+
+ * res/ael/ael.flex, res/ael/pval.c, /, include/asterisk/pval.h,
+ res/ael/ael_lex.c: Merged revisions 162079 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162079 | murf | 2008-12-09 10:18:03 -0700 (Tue, 09 Dec 2008) |
+ 53 lines Merged revisions 162013 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) |
+ 45 lines (closes issue #14019) Reported by: ckjohnsonme Patches:
+ 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme,
+ murf This crash was the result of a few small errors that would
+ combine in 64-bit land to result in a crash. 32-bit land might
+ have seen these combine to mysteriously drop the args to an
+ application call, in certain circumstances. Also, in trying to
+ find this bug, I spotted a situation in the flex input, where, in
+ passing back a 'word' to the parser, it would allocate a buffer
+ larger than necessary. I changed the usage in such situations, so
+ that strdup was not used, but rather, an ast_malloc, followed by
+ ast_copy_string. I removed a field from the pval struct, in u2,
+ that was never getting used, and set in one spot in the code. I
+ believe it was an artifact of a previous fix to make switch cases
+ work invisibly with extens. And, for goto's I removed a '!' from
+ before a strcmp, that has been there since the initial merging of
+ AEL2, that might prevent the proper target of a goto from being
+ found. This was pretty harmless on its own, as it would just
+ louse up a consistency check for users. Many thanks to
+ ckjohnsonme for providing a simplified and complete set of
+ information about the bug, that helped considerably in finding
+ and fixing the problem. Now, to get aelparse up and running again
+ in trunk, and out of its "horribly broken" state, so I can run
+ the regression suite! ........ ................
+
+2008-12-09 16:50 +0000 [r161963-162018] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_disa.c: Merged revisions 162016 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r162016 | russell | 2008-12-09 10:47:39 -0600 (Tue, 09 Dec 2008)
+ | 13 lines Merged revisions 162014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008)
+ | 5 lines Allow DISA to handle extensions that start with #.
+ (closes issue #13330) Reported by: jcovert ........
+ ................
+
+ * /, main/app.c: Merged revisions 161951 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r161951 | russell | 2008-12-09 08:57:39 -0600 (Tue, 09 Dec 2008)
+ | 23 lines Merged revisions 161948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008)
+ | 15 lines Fix a problem with GROUP() settings on a masquerade.
+ The previous code carried over group settings from the old
+ channel to the new one. However, it did nothing with the group
+ settings that were already on the new channel. This patch removes
+ all group settings that already existed on the new channel. I
+ have a more complicated version of this patch which addresses
+ only the most blatant problem with this, which is that a channel
+ can end up with multiple group settings in the same category.
+ However, I could not think of a use case for keeping any of the
+ group settings from the old channel, so I went this route for
+ now. (closes AST-152) ........ ................
+
+2008-12-08 20:55 +0000 [r161835] Joshua Colp <jcolp@digium.com>
+
+ * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8:
+ Merged revisions 161830 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r161830 |
+ file | 2008-12-08 16:53:50 -0400 (Mon, 08 Dec 2008) | 2 lines
+ Update autosupport script with a few changes. ........
+
+2008-12-08 18:52 +0000 [r161792] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 161790 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r161790 |
+ tilghman | 2008-12-08 12:49:50 -0600 (Mon, 08 Dec 2008) | 6 lines
+ Allocate enough space initially for the message. (closes issue
+ #14027) Reported by: junky Patches: M14027.diff uploaded by junky
+ (license 177) ........
+
+2008-12-08 18:49 +0000 [r161729-161789] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, /: Merged revisions 161787 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r161787 |
+ file | 2008-12-08 14:47:32 -0400 (Mon, 08 Dec 2008) | 6 lines Fix
+ a regression introduced when the PBX timeouts were converted to
+ milliseconds. collect_digits now gets milliseconds fed to it, not
+ seconds. (closes issue #14012) Reported by: dveiga Patches:
+ 14012.patch uploaded by bkruse (license 132) ........
+
+ * /, channels/chan_sip.c: Merged revisions 161726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r161726 | file | 2008-12-08 13:53:32 -0400 (Mon, 08 Dec 2008) |
+ 13 lines Merged revisions 161725 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6
+ lines Make the usereqphone option work again. (closes issue
+ #13474) Reported by: mmaguire Patches: 20080912_bug13474.diff
+ uploaded by mmaguire (license 571) ........ ................
+
+2008-12-08 17:24 +0000 [r161722] Matt Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 161721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r161721 |
+ mnicholson | 2008-12-08 11:23:41 -0600 (Mon, 08 Dec 2008) | 7
+ lines Fix a crash that can occur on a transfer in chan_sip when
+ attempting to collect rtp stats. (closes issue #13956) Reported
+ by: chris-mac Tested by: chris-mac ........
+
+2008-12-05 23:29 +0000 [r161496] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 161493 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r161493 |
+ mmichelson | 2008-12-05 17:24:38 -0600 (Fri, 05 Dec 2008) | 8
+ lines If the autoloop flag is set on a channel, then we need to
+ add 1 to the priority when checking if the extension exists.
+ Otherwise, gosubs will fail. This was discovered when
+ investigating an asterisk-users mailing list post made by Gary
+ Hawkins. ........
+
+2008-12-05 21:16 +0000 [r161352-161429] Sean Bright <sean.bright@gmail.com>
+
+ * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
+ 161427 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r161427 | seanbright | 2008-12-05 16:08:43 -0500 (Fri, 05 Dec
+ 2008) | 22 lines Merged revisions 161426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r161426 | seanbright | 2008-12-05 16:02:20 -0500
+ (Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec
+ 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned
+ int). (closes issue #14006) Reported by: alphaque Patches:
+ astobj2.h-patch uploaded by alphaque (license 259) (Slightly
+ modified by seanbright) ........ ................
+ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 161349-161350 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r161349 | seanbright | 2008-12-05 10:56:15 -0500 (Fri,
+ 05 Dec 2008) | 5 lines When using IMAP_STORAGE, it's important to
+ convert bare newlines (\n) in emailbody and pagerbody to CR-LF so
+ that the IMAP server doesn't spit out an error. This was
+ informally reported on #asterisk-dev a few weeks ago. Reviewed by
+ Mark M. on IRC. ........ r161350 | seanbright | 2008-12-05
+ 11:04:36 -0500 (Fri, 05 Dec 2008) | 2 lines Use ast_free()
+ instead of free(), pointed out by eliel on IRC. ........
+
+2008-12-05 14:18 +0000 [r161285-161290] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 161288 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r161288 | russell | 2008-12-05 08:16:24 -0600 (Fri, 05 Dec 2008)
+ | 10 lines Merged revisions 161287 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008)
+ | 2 lines Fix a NULL format string warning found by buildbot.
+ ........ ................
+
+ * /, apps/app_minivm.c: Merged revisions 161252 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r161252 |
+ russell | 2008-12-05 07:46:01 -0600 (Fri, 05 Dec 2008) | 2 lines
+ Resolve a compiler warning from buildbot about a NULL format
+ string. ........
+
+2008-12-05 05:42 +0000 [r161182] Tilghman Lesher <tlesher@digium.com>
+
+ * main/config.c, /: Merged revisions 161181 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r161181 |
+ tilghman | 2008-12-04 23:41:41 -0600 (Thu, 04 Dec 2008) | 11
+ lines The first file should have a blank config filename in the
+ structure, so that when a save occurs to a different filename,
+ everything goes to the alternate filename, instead of appending
+ to the original. This is important for the AMI command
+ UpdateConfig. (closes issue #13301) Reported by: trevo Patches:
+ 20081113__bug13301.diff.txt uploaded by Corydon76 (license 14)
+ 20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license
+ 14) Tested by: Corydon76, blitzrage ........
+
+2008-12-05 02:52 +0000 [r161149] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 161147 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r161147 | seanbright | 2008-12-04 21:47:54 -0500 (Thu, 04 Dec
+ 2008) | 3 lines Check the return value of fread/fwrite so the
+ compiler doesn't complain. Only a problem when IMAP_STORAGE is
+ enabled. ........
+
+2008-12-04 18:37 +0000 [r161016] Jeff Peeler <jpeeler@digium.com>
+
+ * main/rtp.c, /: Merged revisions 161014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r161014 | jpeeler | 2008-12-04 12:32:20 -0600 (Thu, 04 Dec 2008)
+ | 17 lines Merged revisions 161013 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008)
+ | 9 lines (closes issue #13835) Reported by: matt_b Tested by:
+ jpeeler This mirrors a check that was present in ast_rtp_read to
+ also be in ast_rtp_raw_write to not schedule sending the receiver
+ report if the remote RTCP endpoint address isn't present in the
+ RTCP structure. Closes AST-142. ........ ................
+
+2008-12-04 16:48 +0000 [r160947] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/callerid.c: Merged revisions 160945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r160945 | mmichelson | 2008-12-04 10:45:06 -0600 (Thu, 04 Dec
+ 2008) | 23 lines Merged revisions 160943 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec
+ 2008) | 15 lines Fix a callerid parsing issue. If someone
+ formatted callerid like the following: "name <number>" (including
+ the quotation marks), then the parts would be parsed as name:
+ "name number: number This is because the closing quotation mark
+ was not discovered since the number and everything after was
+ parsed out of the string earlier. Now, there is a check to see if
+ the closing quote occurs after the number, so that we can know if
+ we should strip off the opening quote on the name. Closes AST-158
+ ........ ................
+
+2008-12-04 01:41 +0000 [r160858-160859] Richard Mudgett <rmudgett@digium.com>
+
+ * funcs/func_callerid.c, /: Merged revisions 160856 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r160856 | rmudgett | 2008-12-03 19:36:39 -0600 (Wed, 03 Dec 2008)
+ | 1 line Jcolp pointed out that num will also match number
+ ........
+
+ * funcs/func_callerid.c, /: Merged revisions 160854 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r160854 | rmudgett | 2008-12-03 19:14:22 -0600 (Wed, 03 Dec 2008)
+ | 4 lines * Found a couple more places where num/number needed to
+ be done so 1.4 upgraders will not have problems. * Added curly
+ braces and minor tweaks. ........
+
+2008-12-03 22:02 +0000 [r160811] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 160791 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r160791 | tilghman | 2008-12-03 15:58:21 -0600
+ (Wed, 03 Dec 2008) | 9 lines Merged revisions 160770 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03
+ Dec 2008) | 2 lines Some compilers warn on null format strings;
+ some don't (caught by buildbot) ........ ................
+
+2008-12-03 21:40 +0000 [r160766] Steve Murphy <murf@digium.com>
+
+ * funcs/func_callerid.c, /: Merged revisions 160760 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r160760 | murf | 2008-12-03 14:09:15 -0700 (Wed,
+ 03 Dec 2008) | 23 lines Merged revisions 160703 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) |
+ 11 lines (closes issue #13597) Reported by: john8675309 Patches:
+ patch.13597 uploaded by murf (license 17) Tested by: murf,
+ john8675309 This patch causes the setcid func to update the CDR
+ clid after setting the channel field. I also notice that in
+ trunk, the num/number of 1.4 is left out; I decided to include
+ the option to use either in trunk, so as not to have 1.4
+ upgraders not to have problems. ........ ................
+
+2008-12-03 20:36 +0000 [r160702] Jason Parker <jparker@digium.com>
+
+ * main/manager.c, /: Merged revisions 160699-160700 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r160699 | qwell | 2008-12-03 14:32:20 -0600 (Wed, 03 Dec 2008) |
+ 7 lines Fix typo when ListCategories returns none. (closes issue
+ #13994) Reported by: mika Patches: ListCategoriesActionPatch.diff
+ uploaded by mika (license 624) ........ r160700 | qwell |
+ 2008-12-03 14:35:36 -0600 (Wed, 03 Dec 2008) | 1 line Another
+ place this is missing ........
+
+2008-12-03 19:49 +0000 [r160665] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, channels/iax2-provision.c: Merged revisions 160663 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r160663 | eliel | 2008-12-03 17:25:30 -0200 (Wed, 03 Dec
+ 2008) | 13 lines - iax2-provision was not freeing iax_templates
+ structure when unloading the chan_iax2.so module. - Move the code
+ to start using the LIST macros. Review:
+ http://reviewboard.digium.com/r/72 (closes issue #13232) Reported
+ by: eliel Patches: iax2-provision.patch.txt uploaded by eliel
+ (license 64) (with minor changes pointed by Mark Michelson on
+ review board) Tested by: eliel ........
+
+2008-12-03 18:42 +0000 [r160628] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, apps/app_stack.c, apps/app_dial.c, /: Merged
+ revisions 160626 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r160626 |
+ mmichelson | 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) | 16
+ lines Add some safety measures when using gosub, especially when
+ using the options for app_dial and app_queue to run a gosub when
+ the call is answered. * Check for the existence of the gosub
+ target in gosub_exec. If it is nonexistent, then this will cause
+ errors when we attempt to actually run the gosub, including a
+ definite memory leak and potential crashes. Return an error in
+ this situation * Check the return value of pbx_exec in app_dial
+ and app_queue before attempting to actually run the gosub
+ routine. If there was an error, we should not attempt to run the
+ gosub. * Change a '|' to a ',' in app_queue. * Add some extra
+ curly braces where they had been missing previously. (closes
+ issue #13548) Reported by: fiddur ........
+
+2008-12-03 17:41 +0000 [r160561] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c, /: Merged revisions 160559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r160559 | tilghman | 2008-12-03 11:38:59 -0600 (Wed, 03 Dec 2008)
+ | 14 lines Merged revisions 160558 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008)
+ | 7 lines If an entry is added to the directory during a scan
+ when another entry expires, then that new entry will not be
+ processed promptly, but must wait for either a future entry to
+ start or a current entry's retry to occur. If no other entries
+ exist in the directory (other than the new entries) when a bunch
+ expire, then the new entries must wait until another new entry is
+ added to be processed. This was a rather weird race condition,
+ really. Fixes AST-147. ........ ................
+
+2008-12-03 17:10 +0000 [r160557] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 160555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r160555 |
+ mmichelson | 2008-12-03 11:07:09 -0600 (Wed, 03 Dec 2008) | 11
+ lines When investigating issue #13548, I found that gosub
+ handling in app_queue was just completely wrong, mostly because
+ the channel operations being performed were being done on the
+ incorrect channel. With this set of changes, a gosub will
+ correctly run on the answering queue member's channel. There are
+ still crash issues which occur if there are dialplan syntax
+ errors, so I cannot yet close the referenced issue. ........
+
+2008-12-03 17:02 +0000 [r160483-160554] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c, /: Merged revisions 160552 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r160552 | tilghman | 2008-12-03 11:01:03 -0600 (Wed, 03 Dec 2008)
+ | 11 lines Merged revisions 160551 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008)
+ | 4 lines Don't start scanning the directory until all modules
+ are loaded, because some required modules (channels, apps,
+ functions) may not yet be in memory yet. Fixes AST-149. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 160481 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008)
+ | 14 lines Merged revisions 160480 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008)
+ | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I
+ guess that having only ip-phones in mind is not a good approach.
+ Since it is possible to have a sip proxy connected to asterisk we
+ could receive a 407 (unauthorized) or 483 (too many hops) as
+ response and dialog ending would not be a good behavior." So
+ modified. ........ ................
+
+2008-12-02 18:05 +0000 [r160329-160339] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 160333 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r160333 | jpeeler | 2008-12-02 12:04:51 -0600 (Tue, 02 Dec 2008)
+ | 1 line remove duplicate comment that I accidentally merged
+ ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 160319 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r160319 | jpeeler | 2008-12-02 12:00:24 -0600 (Tue, 02 Dec 2008)
+ | 7 lines (closes issue #13786) Reported by: tzafrir Readding
+ DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which
+ fixes not being able to make outgoing calls on some FXO adapters:
+ http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553
+ ........
+
+2008-12-02 18:03 +0000 [r160234-160325] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 160308 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008)
+ | 17 lines Merged revisions 160297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008)
+ | 10 lines When the text does not match exactly (e.g. RTP/SAVP),
+ then the %n conversion fails, and the resulting integer is
+ garbage. Thus, we must initialize the integer and check it
+ afterwards for success. (closes issue #14000) Reported by: folke
+ Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke
+ (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by
+ folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
+ uploaded by folke (license 626) ........ ................
+
+ * include/asterisk/stringfields.h, apps/app_voicemail.c,
+ main/cli.c, main/pbx.c, main/frame.c, /,
+ channels/chan_features.c: Merged revisions 160208 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600
+ (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008)
+ | 3 lines Ensure that Asterisk builds with --enable-dev-mode,
+ even on the latest gcc and glibc. ........ ................
+
+2008-12-01 23:53 +0000 [r160175] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_phone.c, main/manager.c, /, utils/smsq.c: Merged
+ revisions 160170-160172 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec
+ 2008) | 1 line Pay attention to the return value of system(),
+ even if we basically ignore it. ................ r160171 |
+ seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1
+ line Silence a build warning. (chan_phone.c:810: warning: value
+ computed is not used) ................ r160172 | seanbright |
+ 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines Merged
+ revisions 159976 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008)
+ | 3 lines Get rid of the useless format string and argument in
+ the Bogus/ manager channelname. Noted by kpfleming and name
+ Bogus/manager suggested by eliel ........ ................
+
+2008-12-01 Tilghman Lesher <tlesher@digium.com>
+
+ * Released 1.6.1-beta3
+
+2008-12-01 21:46 +0000 [r160101] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 160097 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r160097 | tilghman | 2008-12-01 15:23:37 -0600 (Mon, 01 Dec 2008)
+ | 2 lines Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or
+ bad things happen. ........
+
+2008-12-01 17:45 +0000 [r160006] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 160004 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r160004 | russell | 2008-12-01 11:34:31 -0600
+ (Mon, 01 Dec 2008) | 14 lines Merged revisions 160003 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008)
+ | 6 lines Apply some logic used in iax2_indicate() to
+ iax2_setoption(), as well, since they both have the potential to
+ send control frames in the middle of call setup. We have to wait
+ until we have received a message back from the remote end before
+ we try to send any more frames. Otherwise, the remote end will
+ consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
+ ........ ................
+
+2008-12-01 16:06 +0000 [r159975] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/manager.c, /: Merged revisions 159898 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r159898 | mvanbaak | 2008-12-01 15:09:59 +0100 (Mon, 01 Dec 2008)
+ | 11 lines Merged revisions 159897 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008)
+ | 4 lines make manager compile on OpenBSD. The last (10th)
+ argument to ast_channel_alloc here should be a pointer and NULL
+ is not really a pointer. ........ ................
+
+2008-12-01 14:57 +0000 [r159920] Russell Bryant <russell@digium.com>
+
+ * .cleancount, /: Merged revisions 159911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r159911 | russell | 2008-12-01 08:56:10 -0600 (Mon, 01 Dec 2008)
+ | 10 lines Merged revisions 159900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008)
+ | 2 lines Force a "make clean" to avoid a bizarre build issue ...
+ ........ ................
+
+2008-11-29 18:34 +0000 [r159854] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_readexten.c: Merged revisions 159853 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r159853 | tilghman | 2008-11-29 12:33:18 -0600 (Sat, 29 Nov 2008)
+ | 2 lines Allow the '#' sign to exist within an extension
+ (inspired by issue #13330) ........
+
+2008-11-29 18:16 +0000 [r159851] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c, cdr/cdr_tds.c, include/asterisk/logger.h,
+ include/asterisk/res_odbc.h, channels/chan_misdn.c,
+ include/asterisk/astmm.h, include/asterisk/lock.h,
+ utils/extconf.c, makeopts.in, main/dns.c, funcs/Makefile,
+ include/asterisk/stringfields.h, include/asterisk/utils.h,
+ include/asterisk/devicestate.h, /, include/asterisk/dundi.h,
+ configure.ac, utils/astman.c, include/asterisk/cli.h,
+ include/asterisk/channel.h, include/asterisk/manager.h,
+ res/res_config_sqlite.c, utils/conf2ael.c, utils/frame.c,
+ channels/misdn_config.c, main/ast_expr2.c, Makefile, main/srv.c,
+ include/asterisk/compat.h, configure, channels/misdn/ie.c,
+ include/asterisk/module.h, main/features.c,
+ include/asterisk/linkedlists.h, main/logger.c, main/event.c,
+ include/asterisk/dlinkedlists.h, include/asterisk/strings.h,
+ utils/check_expr.c, channels/chan_vpb.cc, channels/chan_sip.c,
+ main/Makefile, include/asterisk/enum.h, channels/chan_agent.c,
+ main/utils.c, include/jitterbuf.h: Merged revisions 159818 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29
+ Nov 2008) | 18 lines incorporates r159808 from branches/1.4:
+ ------------------------------------------------------------------------
+ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov
+ 2008) | 7 lines update dev-mode compiler flags to match the ones
+ used by default on Ubuntu Intrepid, so all developers will see
+ the same warnings and errors since this branch already had some
+ printf format attributes, enable checking for them and tag
+ functions that didn't have them format attributes in a consistent
+ way
+ ------------------------------------------------------------------------
+ in addition: move some format attributes from main/utils.c to the
+ header files they belong in, and fix up references to the
+ relevant functions based on new compiler warnings ........
+
+2008-11-26 19:58 +0000 [r159561] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 159554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 |
+ mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19
+ lines Add some necessary hangup commands in the case that
+ forwarding a call fails 1) Hang up the original destination if
+ the local channel cannot be requested. 2) Hang up the local
+ channel (in addition to the original destination) if ast_call
+ fails when calling the newly created local channel. This prevents
+ channels from sticking around forever in the case of a botched
+ call forward (e.g. to an extension which does not exist). (closes
+ issue #13764) Reported by: davidw Patches: 13764_v2.patch
+ uploaded by putnopvut (license 60) Tested by: putnopvut, davidw
+ ........
+
+2008-11-26 19:17 +0000 [r159535] Kevin P. Fleming <kpfleming@digium.com>
+
+ * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules,
+ Makefile.rules: Merged revisions 159534 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r159534 | kpfleming | 2008-11-26 13:08:56 -0600 (Wed, 26 Nov
+ 2008) | 11 lines Merged revisions 159476 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov
+ 2008) | 7 lines simplify (and slightly bug-fix) the recent
+ developer-oriented COMPILE_DOUBLE mode ensure that 'make clean'
+ removes dependency files for .i files that are created in
+ COMPILE_DOUBLE mode ........ ................
+
+2008-11-26 18:38 +0000 [r159477] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c, /: Merged revisions 159475 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r159475 |
+ tilghman | 2008-11-26 12:33:04 -0600 (Wed, 26 Nov 2008) | 7 lines
+ If the config file does not exist, then the first use crashes
+ Asterisk. (closes issue #13848) Reported by: klaus3000 Patches:
+ udptl.c.patch uploaded by eliel (license 64) Tested by: blitzrage
+ ........
+
+2008-11-26 14:59 +0000 [r159438] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 159437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r159437 | mmichelson | 2008-11-26 08:58:17 -0600 (Wed, 26 Nov
+ 2008) | 10 lines Don't allow for configuration options to
+ overwrite options set via channel variables on a reload. (closes
+ issue #13921) Reported by: davidw Patches: 13921.patch uploaded
+ by putnopvut (license 60) Tested by: davidw ........
+
+2008-11-26 03:19 +0000 [r159403] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: Merged revisions 159402 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r159402 |
+ jpeeler | 2008-11-25 21:18:01 -0600 (Tue, 25 Nov 2008) | 3 lines
+ Always parse arguments in park_call_exec so that app_args is
+ valid. This prevents a crash when executing Park from the
+ dialplan with no arguments. ........
+
+2008-11-25 23:27 +0000 [r159375] Steve Murphy <murf@digium.com>
+
+ * channels/chan_iax2.c, main/cdr.c, /: Merged revisions 159360 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue,
+ 25 Nov 2008) | 23 lines Merged revisions 159316 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) |
+ 15 lines (closes issue #12694) Reported by: yraber Patches:
+ 12694.2nd.diff uploaded by murf (license 17) Tested by: murf,
+ laurav Thanks to file (Joshua Colp) for his IAX fix. the change
+ to cdr.c allows no-answer to percolate up into CDR's, and feels
+ like the right place to locate this fix; if BUSY is done here,
+ no-answer should be, too. ........ ................
+
+2008-11-25 21:58 +0000 [r159249-159280] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 159276 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r159276 | tilghman | 2008-11-25 15:57:59 -0600
+ (Tue, 25 Nov 2008) | 14 lines Merged revisions 159269 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008)
+ | 7 lines Don't try to send a response on a NULL pvt. (closes
+ issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch
+ uploaded by eliel (license 64) Tested by: barthpbx ........
+ ................
+
+ * channels/chan_iax2.c, /: Merged revisions 159247 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r159247 | tilghman | 2008-11-25 15:42:42 -0600
+ (Tue, 25 Nov 2008) | 21 lines Merged revisions 159246 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600
+ (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008)
+ | 7 lines Regression fix for last security fix. Set the iseqno
+ correctly. (closes issue #13918) Reported by: ffloimair Patches:
+ 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: ffloimair ........ ................ ................
+
+2008-11-25 16:21 +0000 [r159095] Terry Wilson <twilson@digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 159093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r159093 |
+ twilson | 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) | 2 lines
+ Add missing variable declaration for PPC code ........
+
+2008-11-25 05:05 +0000 [r159053] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/xpmr/xpmr.c, apps/app_rpt.c, channels/chan_usbradio.c,
+ /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 159050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r159050 | tilghman | 2008-11-24 23:02:11 -0600 (Mon, 24 Nov 2008)
+ | 10 lines Merged revisions 159025 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008)
+ | 3 lines System call ioperm is non-portable, so check for its
+ existence in autoconf. (Closes issue #13863) ........
+ ................
+
+2008-11-25 03:51 +0000 [r158993] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_usbradio.c, /: Merged revisions 158992 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r158992 | twilson | 2008-11-24 21:49:30 -0600 (Mon, 24 Nov 2008)
+ | 2 lines Make chan_usbradio compile under dev mode ........
+
+2008-11-25 00:41 +0000 [r158894-158927] Matt Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c, /, UPGRADE.txt: Merged revisions 158924 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r158924 | mnicholson | 2008-11-24 18:05:41 -0600 (Mon,
+ 24 Nov 2008) | 6 lines Make the Join event from app_queue use
+ CallerIDNum insead of CallerID for indicating the callerid number
+ just like the rest of asterisk. (closes issue #13883) Reported
+ by: davidw ........
+
+ * /, main/file.c: Merged revisions 158925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158925 |
+ mnicholson | 2008-11-24 18:19:55 -0600 (Mon, 24 Nov 2008) | 2
+ lines Fix compiling in dev mode. ........
+
+ * include/asterisk/manager.h, main/manager.c, /, res/res_agi.c:
+ Merged revisions 158876 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158876 |
+ mnicholson | 2008-11-24 15:56:22 -0600 (Mon, 24 Nov 2008) | 7
+ lines Added EVENT_FLAG_AGI and used it for manager calls in
+ res_agi.c (closes issue #13873) Reported by: fnordian Patches:
+ ami_agievent.patch uploaded by fnordian (license 110) ........
+
+2008-11-24 21:53 +0000 [r158861] Tilghman Lesher <tlesher@digium.com>
+
+ * main/dsp.c, /: Merged revisions 158857 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158857 |
+ tilghman | 2008-11-24 15:52:34 -0600 (Mon, 24 Nov 2008) | 3 lines
+ Add a bit of documentation (thanks, I-MOD) on what the silence
+ threshold constant actually does and what values are valid for
+ it. ........
+
+2008-11-24 21:44 +0000 [r158855] Matt Nicholson <mnicholson@digium.com>
+
+ * /, main/file.c: Merged revisions 158851 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158851 |
+ mnicholson | 2008-11-24 15:27:26 -0600 (Mon, 24 Nov 2008) | 6
+ lines Make ast_streamfile() check the result of ast_openstream()
+ before doing anything with it. (closes issue #13955) Reported by:
+ chris-mac ........
+
+2008-11-22 17:00 +0000 [r158689-158701] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_skinny.c: Merged revisions 158694 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r158694 | mvanbaak | 2008-11-22 17:57:11 +0100 (Sat, 22 Nov 2008)
+ | 8 lines dont send reorder tone after a device is hungup if a
+ dialout is abandoned or failed. Without this reorder tone will
+ play after hangup and both wedhorn's and my wife have threatened
+ to use an axe on our asterisk box (closes issue #13948) Reported
+ by: wedhorn Patches: switch.diff uploaded by wedhorn (license 30)
+ ........
+
+ * /, channels/chan_skinny.c: Merged revisions 158688 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r158688 | mvanbaak | 2008-11-22 17:06:38 +0100 (Sat, 22 Nov 2008)
+ | 4 lines fix a very occasional core dump in chan_skinny found by
+ wedhorn. (issue #13948) ........
+
+2008-11-21 23:45 +0000 [r158607] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 158606 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r158606 | murf | 2008-11-21 16:40:46 -0700 (Fri, 21 Nov 2008) |
+ 19 lines Merged revisions 158603 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) |
+ 11 lines In reference to the fix made for 13871, I was merging
+ the fix into 1.6.0 and realized I missed the code in the h-exten
+ block, and didn't catch it because my test case had the h-exten
+ commented out. So, this corrects the code I missed, as a
+ preventative against another crash report. Tested with the
+ h-exten defined, all is well. ........ ................
+
+2008-11-21 23:15 +0000 [r158604] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 158602 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008)
+ | 12 lines Merged revisions 158600 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008)
+ | 5 lines The passed extension may not be the same in the list as
+ the current entry, because we strip spaces when copying the
+ extension into the structure. Therefore, use the copied item to
+ place the item into the list. (found by lmadsen on -dev, fixed by
+ me) ........ ................
+
+2008-11-21 22:57 +0000 [r158572] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 158484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) |
+ 19 lines Merged revisions 158483 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) |
+ 11 lines (closes issue #13871) Reported by: mdu113 This one is
+ totally my fault. The code doesn't even create a bridge CDR if
+ the channel CDR has POST_DISABLED. I didn't check for that at the
+ end of the bridge. Fixed with a few small insertions. Tested.
+ Looks good. No cdr generated, no crash, no unnecc. data objects
+ created either. ........ ................
+
+2008-11-21 22:13 +0000 [r158541] Russell Bryant <russell@digium.com>
+
+ * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
+ 158540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008)
+ | 10 lines Merged revisions 158539 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008)
+ | 2 lines When compiling with DEBUG_THREADS, report the real
+ file/func/line for ao2_lock/ao2_unlock ........ ................
+
+2008-11-21 20:43 +0000 [r158450] Kevin P. Fleming <kpfleming@digium.com>
+
+ * CHANGES, /, UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt,
+ UPGRADE-1.6.txt: Merged revisions 158449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158449 |
+ kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov 2008) | 3
+ lines as suggested by jtodd, document the purposes of the CHANGES
+ and UPGRADE files ........
+
+2008-11-21 19:42 +0000 [r158415] Jason Parker <jparker@digium.com>
+
+ * main/manager.c, /: Merged revisions 158414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158414 |
+ qwell | 2008-11-21 13:40:57 -0600 (Fri, 21 Nov 2008) | 7 lines
+ Make sure we add the Event header for CoreShowChannels. (closes
+ issue #13334) Reported by: srt Patches:
+ 13334_missing_event_header_in_core_show_channel.diff uploaded by
+ srt (license 378) ........
+
+2008-11-21 17:17 +0000 [r158377] Terry Wilson <twilson@digium.com>
+
+ * cdr/cdr_csv.c, /: Merged revisions 158374 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158374 |
+ twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines
+ Reloading the config and having no changes still initialized some
+ settings to 0. Initialize settings after doing all of the cfg
+ checks. (closes issue #13942) Reported by: davidw Patches:
+ cdr_diff.txt uploaded by otherwiseguy (license 396) Tested by:
+ davidw ........
+
+2008-11-21 01:23 +0000 [r158223-158268] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 158265-158266 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu,
+ 20 Nov 2008) | 4 lines Use some magic constants to get the right
+ size for this sscanf statement. Thanks Richard! ........ r158266
+ | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3
+ lines Use a more expressive constant for a 64-bit scanned int
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 158262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 |
+ mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6
+ lines Fix the build for 32-bit systems. %lu is only 32-bits on
+ 32-bit systems, so we need to use %llu instead. Of course %llu is
+ 128-bits on 64-bit systems, so we have to cast to unsigned long
+ long. No harm, but it's sure annoying. ........
+
+ * /, channels/chan_sip.c: Merged revisions 158230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 |
+ mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20
+ lines Change the remote user agent session version variable from
+ an int to a uint64_t. This prevents potential comparison problems
+ from happening if the version string exceeds INT_MAX. This was an
+ apparent problem for one user who could not properly place a call
+ on hold since the version in the SDP of the re-INVITE to place
+ the call on hold greatly exceeded INT_MAX. This also aligns with
+ RFC 2327 better since it recommends using an NTP timestamp for
+ the version (which is a 64-bit number). (closes issue #13531)
+ Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut
+ (license 60) Tested by: sgofferj ........
+
+ * channels/chan_sip.c: Change this so it actually compiles. Thanks,
+ Terry!
+
+2008-11-20 19:43 +0000 [r158191] Sean Bright <sean.bright@gmail.com>
+
+ * res/ael/pval.c, /: Merged revisions 158188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158188 |
+ seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10
+ lines Fix one case where the application argument was not
+ converted from a pipe to a comma. This was causing problems with
+ switch statements with empty expressions. (closes issue #13901)
+ Reported by: smurfix Patches: 20081118_bug13901.diff uploaded by
+ seanbright (license 71) Tested by: seanbright Reviewed by: murf
+ ........
+
+2008-11-20 18:23 +0000 [r158135] Terry Wilson <twilson@digium.com>
+
+ * cdr/cdr_odbc.c, cdr/cdr_radius.c, cdr/cdr_custom.c,
+ cdr/cdr_manager.c, cdr/cdr_csv.c, cdr/cdr_sqlite3_custom.c, /,
+ cdr/cdr_sqlite.c, cdr/Makefile, cdr/cdr_adaptive_odbc.c,
+ cdr/cdr_pgsql.c: Merged revisions 158072 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 |
+ twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines
+ Begin on a crusade to end trailing whitespace! ........
+
+2008-11-20 18:20 +0000 [r158084-158134] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c,
+ /, channels/chan_sip.c, main/file.c: Merged revisions 158133 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r158133 | mmichelson | 2008-11-20 12:20:00 -0600
+ (Thu, 20 Nov 2008) | 10 lines Merged revisions 158072 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20
+ Nov 2008) | 2 lines Begin on a crusade to end trailing
+ whitespace! ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 158082 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov
+ 2008) | 24 lines Merged revisions 158071 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov
+ 2008) | 16 lines We don't handle 4XX responses to BYE well.
+ According to section 15 of RFC 3261, we should terminate a dialog
+ if we receive a 481 or 408 in response to our BYE. Since I am
+ aware of at least one phone manufacturer who may sometimes send a
+ 404 as well, I am being liberal and saying that any 4XX response
+ to a BYE should result in a terminated dialog. (closes issue
+ #12994) Reported by: pabelanger Patches: 12994.patch uploaded by
+ putnopvut (license 60) Closes AST-129 ........ ................
+
+2008-11-20 17:42 +0000 [r158069] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/file.c: Merged revisions 158062 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r158062 |
+ jpeeler | 2008-11-20 11:37:31 -0600 (Thu, 20 Nov 2008) | 6 lines
+ (closes issue #12929) Reported by: snyfer This handles the case
+ for a zero length file to attempt to be streamed. Instead of
+ failing from not playing any data, go ahead and return success as
+ ast_streamfile should consider playing nothing a success when
+ there is nothing to play. ........
+
+2008-11-20 17:40 +0000 [r158067] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 158066
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r158066 | mmichelson | 2008-11-20 11:39:06 -0600
+ (Thu, 20 Nov 2008) | 20 lines Merged revisions 158053 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov
+ 2008) | 12 lines Make sure to set the hangup cause on the calling
+ channel in the case that ast_call() fails. For incoming SIP
+ channels, this was causing us to send a 603 instead of a 486 when
+ the call-limit was reached on the destination channel. (closes
+ issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded
+ by putnopvut (license 60) Tested by: blitzrage ........
+ ................
+
+2008-11-20 00:10 +0000 [r157975] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree,
+ channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, res/ael,
+ channels, main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash,
+ codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules,
+ main/db1-ast/mpool, res/ais, channels/misdn, res/snmp,
+ Makefile.rules, pbx/Makefile, res/Makefile: Merged revisions
+ 157974 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r157974 | kpfleming | 2008-11-19 18:08:12 -0600 (Wed, 19 Nov
+ 2008) | 13 lines Merged revisions 157859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov
+ 2008) | 7 lines the gcc optimizer frequently finds broken code
+ (use of uninitalized variables, unreachable code, etc.), which is
+ good. however, developers usually compile with the optimizer
+ turned off, because if they need to debug the resulting code,
+ optimized code makes that process very difficult. this means that
+ we get code changes committed that weren't adequately checked
+ over for these sorts of problems. with this build system change,
+ if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is
+ turned on, when a source file is compiled it will actually be
+ preprocessed (into a .i or .ii file), then compiled once with
+ optimization (with the result sent to /dev/null) and again
+ without optimization (but only if the first compile succeeded, of
+ course). while making these changes, i did some cleanup work in
+ Makefile.rules to move commonly-used combinations of flag
+ variables into their own variables, to make the file easier to
+ read and maintain ........ ................
+
+2008-11-19 18:29 +0000 [r157785] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 157784 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r157784 | tilghman | 2008-11-19 12:28:14 -0600 (Wed, 19 Nov 2008)
+ | 6 lines Add check for t38_terminal_init in spandsp (not found
+ in 0.0.6, so it should fail reasonably) (closes issue #13473)
+ Reported by: genie Patches: 20080916__bug13473.diff.txt uploaded
+ by Corydon76 (license 14) ........
+
+2008-11-19 13:47 +0000 [r157719-157744] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 157743 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r157743 |
+ kpfleming | 2008-11-19 07:45:48 -0600 (Wed, 19 Nov 2008) | 1 line
+ correct small bug introduced during API conversion ........
+
+ * CHANGES, apps/app_stack.c, include/asterisk/agi.h, /,
+ res/res_agi.c, UPGRADE.txt, UPGRADE-1.6.txt (added): Merged
+ revisions 157706 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 |
+ kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5
+ lines make some corrections to the ast_agi_register_multiple(),
+ ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to
+ be consistent with API guidelines also, move UPGRADE.txt to
+ UPGRADE-1.6.txt and make the new UPGRADE.txt contain information
+ about upgrading between Asterisk 1.6 releases ........
+
+2008-11-19 01:08 +0000 [r157641] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/logger.h, /, main/logger.c, main/utils.c,
+ include/asterisk/strings.h: Merged revisions 157639 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r157639 | tilghman | 2008-11-18 19:02:45 -0600 (Tue, 18 Nov 2008)
+ | 7 lines Starting with a change to ensure that ast_verbose()
+ preserves ABI compatibility in 1.6.1 (as compared to 1.6.0 and
+ versions of 1.4), this change also deprecates the use of Asterisk
+ with FreeBSD 4, given the central use of va_copy in core
+ functions. va_copy() is C99, anyway, and we already require C99
+ for other purposes, so this isn't really a big change anyway.
+ This change also simplifies some of the core ast_str_* functions.
+ ........
+
+2008-11-19 01:00 +0000 [r157636] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/astmm.c: Merged revisions 157632 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r157632 |
+ mmichelson | 2008-11-18 18:59:48 -0600 (Tue, 18 Nov 2008) | 10
+ lines If malloc returns NULL, we need to return NULL immediately
+ or else Asterisk will crash when attempting to dereference the
+ NULL pointer (closes issue #13858) Reported by: eliel Patches:
+ astmm.c.patch uploaded by eliel (license 64) ........
+
+2008-11-19 00:38 +0000 [r157602] Sean Bright <sean.bright@gmail.com>
+
+ * build_tools/make_buildopts_h, makeopts.in, Makefile, /,
+ build_tools/make_version, configure, configure.ac: Merged
+ revisions 157600 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r157600 |
+ seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10
+ lines Fix a few build problems on Solaris (and check for an md5
+ utility in configure instead of the icky loop I was doing
+ before). (closes issue #13842) Reported by: snuffy Patches:
+ bug13842_20081106.diff uploaded by snuffy (license 35) 13842.diff
+ uploaded by seanbright (license 71) Tested by: snuffy ........
+
+2008-11-18 23:59 +0000 [r157429-157596] Mark Michelson <mmichelson@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 157592 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r157592 | mmichelson | 2008-11-18 17:59:02 -0600 (Tue, 18 Nov
+ 2008) | 10 lines This change prevents a crash from occurring if
+ res_musiconhold.so is unloaded and then Asterisk is stopped. The
+ problem was that we are not unregistering the ast_moh_destroy
+ function at exit. (closes issue #13761) Reported by: eliel
+ Patches: res_musiconhold.c.patch uploaded by eliel (license 64)
+ ........
+
+ * apps/app_voicemail.c, /: Merged revisions 157562 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r157562 | mmichelson | 2008-11-18 17:28:23 -0600 (Tue, 18 Nov
+ 2008) | 11 lines Fix the logic for when delete=yes when IMAP
+ storage is in use so that the message is deleted from both local
+ and IMAP storage. (closes issue #13642) Reported by: jaroth
+ Patches: deleteyes.patch uploaded by jaroth (license 50) ........
+
+ * /, channels/chan_sip.c: Merged revisions 157512 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov
+ 2008) | 21 lines Merged revisions 157503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov
+ 2008) | 13 lines Add some missing invite state changes necessary
+ in the sip_write function. Not setting the invite state correctly
+ on the call was resulting in the Record application leaving empty
+ files. I also have updated the doxygen comment next to the
+ declaration of the INV_EARLY_MEDIA constant to reflect that we
+ also use this state when we *send* a 18X response to an INVITE.
+ (closes issue #13878) Reported by: nahuelgreco Patches:
+ sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
+ (license 162) Tested by: putnopvut ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 157496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 |
+ mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6
+ lines Based on Russell's advice on the asterisk-dev list, I have
+ changed from using a global lock in update_call_counter to using
+ the locks within the sip_pvt and sip_peer structures instead.
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 157427 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 |
+ mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13
+ lines * Add a lock to be used in the update_call_counter
+ function. * Revert logic to mirror 1.4's in the sense that it
+ will not allow the call counter to dip below 0. These two
+ measures prevent potential races that could cause a SIP peer to
+ appear to be busy forever. (closes issue #13668) Reported by: mjc
+ Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic
+ (license 586) ........
+
+2008-11-18 19:18 +0000 [r157367] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 157366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r157366 | jpeeler | 2008-11-18 13:16:00 -0600 (Tue, 18 Nov 2008)
+ | 14 lines Merged revisions 157365 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008)
+ | 6 lines (closes issue #13899) Reported by: akkornel This fix is
+ the result of a bug fix in ast_app_separate_args r124395. If an
+ argument does not exist it should always be set to a null string
+ rather than a null pointer. ........ ................
+
+2008-11-18 18:32 +0000 [r157308] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_followme.c, apps/app_dial.c, channels/chan_local.c, /,
+ main/features.c, include/asterisk/channel.h: Merged revisions
+ 157306 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov
+ 2008) | 20 lines Merged revisions 157305 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov
+ 2008) | 12 lines Fix a crash in the end_bridge_callback of
+ app_dial and app_followme which would occur at the end of an
+ attended transfer. The error occurred because we initially stored
+ a pointer to an ast_channel which then was hung up due to a
+ masquerade. This commit adds a "fixup" callback to the
+ bridge_config structure to allow for end_bridge_callback_data to
+ be changed in the case that a new channel pointer is needed for
+ the end_bridge_callback. ........ ................
+
+2008-11-18 18:20 +0000 [r157304] Steve Murphy <murf@digium.com>
+
+ * main/config.c, /: Merged revisions 157302 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r157302 |
+ murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines
+ (closes issue #13420) Reported by: alex70 Patches:
+ 13420.13539.patch uploaded by murf (license 17) Tested by: murf,
+ awk This fixes two problems: a spurious linefeed insertion
+ probably left over from pre-precomment times. Only generated when
+ category had no previous comments. The other problem: Insertions
+ could get the line-numbering out of whack and generate negative
+ line numbers, causing chunks of line numbers to be emitted, on
+ the scale of the number of lines up to that point in the file. In
+ such cases, abort the looping, and all is well. ........
+
+2008-11-17 22:39 +0000 [r157255] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 157253 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r157253 |
+ tilghman | 2008-11-17 16:25:06 -0600 (Mon, 17 Nov 2008) | 8 lines
+ Can't use items duplicated off the stack frame in an element
+ returned from a function: in these cases, we have to use the
+ heap, or garbage will result. (closes issue #13898) Reported by:
+ alecdavis Patches: 20081114__bug13898__2.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: alecdavis ........
+
+2008-11-15 19:49 +0000 [r157108-157166] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged
+ revisions 157164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov
+ 2008) | 13 lines Merged revisions 157162-157163 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov
+ 2008) | 1 line dist-clean should remove dependency information
+ files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03
+ +0100 (Sat, 15 Nov 2008) | 1 line when an individual directory
+ dist-clean is run, run clean in that directory first, and when
+ running top-level dist-clean, do not run subdirectory clean
+ operations twice ........ ................
+
+ * /, contrib/asterisk-ng-doxygen: Merged revisions 157105 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15
+ Nov 2008) | 13 lines major update to doxygen configuration file:
+ 1) update to doxygen 1.5.x style file, as used in trunk 2) tell
+ doxygen where are header files are, so include-file processing
+ can be done 3) make all macros that are used to define
+ variables/functions be expanded, so that doxygen will properly
+ document the resulting variable/function 4) make all macros that
+ are used to provide the contents of a variable (structure) be
+ expanded, so that doxygen will be able to document the resulting
+ fields 5) suppress compiler attributes (__attribute__(xxx)) from
+ being seen by doxygen, so it will properly match up function
+ definition and usage (for an example of th effect of this, look
+ at the doxygen docs for ast_log() from before and afte this
+ commit) ........
+
+2008-11-15 04:30 +0000 [r157040-157042] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c, main/features.c, main/taskprocessor.c:
+ Merged revisions 157041 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r157041 |
+ russell | 2008-11-14 22:25:57 -0600 (Fri, 14 Nov 2008) | 3 lines
+ Fix a few more places where the case insensitive hash should be
+ used since the comparison is case insensitive. ........
+
+ * /, channels/chan_console.c: Merged revisions 157039 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r157039 | russell | 2008-11-14 22:08:42 -0600 (Fri, 14 Nov 2008)
+ | 3 lines Use the new case insensitive hash function for console
+ interfaces. The comparison function is case insensitive. ........
+
+2008-11-14 21:21 +0000 [r156963] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 156962 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r156962 |
+ mmichelson | 2008-11-14 15:19:58 -0600 (Fri, 14 Nov 2008) | 7
+ lines Revision 155513 of chan_sip.c in trunk inadvertently
+ removed a very important line to set the "len" field for incoming
+ SIP requests. The result was that all incoming SIP messages
+ appeared to be 0-length, meaning Asterisk could do no meaningful
+ processing of anything SIP-related ........
+
+2008-11-14 17:04 +0000 [r156913] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /: Merged revisions 156911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r156911 |
+ tilghman | 2008-11-14 11:02:00 -0600 (Fri, 14 Nov 2008) | 4 lines
+ Ping is missing the standard double-newline after the event.
+ (closes issue #13903) Reported by: kebl0155 ........
+
+2008-11-14 16:57 +0000 [r156819-156894] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, include/asterisk/strings.h: This is the 1.6.1
+ version of trunk commit 156883. It is functionally equivalent to
+ the 1.6.0 commit
+
+ * apps/app_voicemail.c, /: Merged revisions 156817 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r156817 | mmichelson | 2008-11-14 09:20:03 -0600
+ (Fri, 14 Nov 2008) | 18 lines Merged revisions 156816 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov
+ 2008) | 10 lines If the prompt to reenter a voicemail password
+ timed out, it resulted in the password not being saved, even if
+ the input matched what you gave when first prompted to enter a
+ new password. This is because the return value of ast_readstring
+ was checked, but not checked properly. This bug was discovered by
+ Jared Smith during an Asterisk training course. Thanks for
+ reporting it! ........ ................
+
+2008-11-14 00:44 +0000 [r156691-156757] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_while.c, /: Merged revisions 156756 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008)
+ | 13 lines Merged revisions 156755 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008)
+ | 6 lines ast_waitfordigit() requires that the channel be up, for
+ no good logical reason. This prevents While/EndWhile from working
+ within the "h" extension. Reported by: jgalarneau (for ABE C.2)
+ Fixed by: me ........ ................
+
+ * main/manager.c, /: Merged revisions 156690 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r156690 | tilghman | 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008)
+ | 14 lines Merged revisions 156688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008)
+ | 7 lines Provide more space for all the data which can appear in
+ an originating channel name. (closes issue #13398) Reported by:
+ bamby Patches: manager.c.diff uploaded by bamby (license 430)
+ ........ ................
+
+2008-11-13 19:29 +0000 [r156654] Brandon Kruse <bkruse@digium.com>
+
+ * main/manager.c: Merged revisions 156017 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r156017 |
+ pari | 2008-11-11 17:02:43 -0600 (Tue, 11 Nov 2008) | 5 lines
+ Patch by Ryan Brindley -- Make sure that manager refuses any
+ duplicate 'new category' requests in updateconfig (closes issue
+ #13539) ........
+
+2008-11-13 19:18 +0000 [r156650] Jeff Peeler <jpeeler@digium.com>
+
+ * main/pbx.c, /: Merged revisions 156649 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r156649 |
+ jpeeler | 2008-11-13 13:17:50 -0600 (Thu, 13 Nov 2008) | 6 lines
+ (closes issue #13891) Reported by: smurfix This reverts a change
+ I made in 116297. At the time it seemed the change was required
+ to solve an issue with attempting a transfer but then letting it
+ timeout without dialing any digits. However, I didn't realize
+ that having an empty extension was possible. I'm removing the
+ immediate return that was added in pbx_find_extension if the
+ extension is null. ........
+
+2008-11-13 17:12 +0000 [r156614] Mark Michelson <mmichelson@digium.com>
+
+ * autoconf/ast_c_compile_check.m4, /, configure: Merged revisions
+ 156612 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r156612 |
+ mmichelson | 2008-11-13 11:07:56 -0600 (Thu, 13 Nov 2008) | 4
+ lines Kevin sent a note indicating that this change is not
+ necessary, so I am reverting it ........
+
+2008-11-12 21:36 +0000 [r156389] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 156388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008)
+ | 12 lines Merged revisions 156386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008)
+ | 5 lines When using call limits under 1 second, infinite call
+ lengths are allowed, instead. (closes issue #13851) Reported by:
+ ruddy ........ ................
+
+2008-11-12 20:11 +0000 [r156354] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /: Merged revisions 156299 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) |
+ 26 lines Merged revisions 156297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) |
+ 18 lines It turns out that the 0x0XX00 codes being returned for
+ N, X, and Z are off by one, as per conversation with jsmith on
+ #asterisk-dev; he was teaching a class and disconcerted that this
+ published rule was not being followed, with patterns _NXX,
+ _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
+ have been. This change, tested on these 3 patterns now picks the
+ proper one. However, this change may surprise users who set up
+ dialplans based on previous behavior, which has been there for
+ what, 2 and half years or so now. ........ ................
+
+2008-11-12 19:29 +0000 [r156296] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 156295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r156295 | tilghman | 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008)
+ | 13 lines Merged revisions 156294 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008)
+ | 6 lines If the SLA thread is not started, then reload causes a
+ memory leak. (closes issue #13889) Reported by: eliel Patches:
+ app_meetme.c.patch uploaded by eliel (license 64) ........
+ ................
+
+2008-11-12 19:11 +0000 [r156291] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 156290 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r156290 | jpeeler | 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008)
+ | 11 lines Merged revisions 156289 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008)
+ | 3 lines For whatever reason, gcc only warned me about the
+ possible use of an uninitialized variable when compiling 1.6.1.
+ ........ ................
+
+2008-11-12 19:05 +0000 [r156284-156288] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 156243 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r156243 | tilghman | 2008-11-12 12:55:18 -0600
+ (Wed, 12 Nov 2008) | 18 lines Merged revisions 156229 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008)
+ | 11 lines Revert revision 132506, since it occasionally caused
+ IAX2 HANGUP packets not to be sent, and instead, schedule a task
+ to destroy the iax2 pvt structure 10 seconds later. This allows
+ the IAX2 HANGUP packet to be queued, transmitted, and ACKed
+ before the pvt is destroyed. (closes issue #13645) Reported by:
+ dzajro Patches: 20081111__bug13645__3.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: vazir Reviewed:
+ http://reviewboard.digium.com/r/51/ ........ ................
+
+ * apps/app_meetme.c: Fix build (res possibly unused in this
+ function, says gcc)
+
+2008-11-12 18:55 +0000 [r156247] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 156228 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008)
+ | 16 lines Merged revisions 156178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008)
+ | 8 lines (closes issue #13173) Reported by: pep This change adds
+ an announce_thread responsible for playing announcements to an
+ existing conference. This allows all announcing to be immediately
+ stopped if necessary but more importantly allows other threads
+ that need to play something to not block. There are multiple
+ benefits to this, but the actual bug is for solving the scenario
+ for a channel to be unusable after hang up for the entire
+ duration of the parting announcement. The parting announcement
+ can be extremely long depending on what the user recorded upon
+ joining the conference. Reviewed by Russell on Review Board:
+ http://reviewboard.digium.com/r/25/ ........ ................
+
+2008-11-12 17:48 +0000 [r156171] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 156169 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov
+ 2008) | 15 lines Merged revisions 156167 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov
+ 2008) | 7 lines When doing some tests, I was having a crash at
+ the end of every call if an attended transfer occurred during the
+ call. I traced the cause to the CDR on one of the channels being
+ NULL. murf suggested a check in the end bridge callback to be
+ sure the CDR is non-NULL before proceeding, so that's what I'm
+ adding. ........ ................
+
+2008-11-12 17:38 +0000 [r156168] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, /: Merged revisions 156166 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r156166 | russell | 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008)
+ | 15 lines Merged revisions 156164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008)
+ | 7 lines Move the sanity check that makes sure "always fork" is
+ not set along with the console option to be after the code that
+ reads options from asterisk.conf. This resolves a situation where
+ Asterisk can start taking up 100% when misconfigured. (Thanks to
+ Bryce Porter (x86 on IRC) for letting me log in to his system to
+ figure out what was causing the 100% CPU problem.) ........
+ ................
+
+2008-11-12 15:34 +0000 [r156128] Mark Michelson <mmichelson@digium.com>
+
+ * autoconf/ast_c_compile_check.m4, /, configure: Merged revisions
+ 156127 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r156127 |
+ mmichelson | 2008-11-12 09:33:11 -0600 (Wed, 12 Nov 2008) | 5
+ lines Add a couple of AC_SUBST calls to the AST_C_COMPILE_CHECK
+ macro. These missing calls were discovered when working on
+ timerfd support in a separate branch. ........
+
+2008-11-11 19:52 +0000 [r156005] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_realtime.c: Merged revisions 155862 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r155862 |
+ tilghman | 2008-11-10 15:12:28 -0600 (Mon, 10 Nov 2008) | 5 lines
+ Make documentation of update method match documentation and
+ update update2 method to match. Reported by: atis, via -dev
+ mailing list. Fixed by: me ........
+
+2008-11-10 21:15 +0000 [r155864] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 155863 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r155863 | mmichelson | 2008-11-10 15:14:44 -0600
+ (Mon, 10 Nov 2008) | 22 lines Merged revisions 155861 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov
+ 2008) | 14 lines Channel drivers assume that when their indicate
+ callback is invoked, that the channel on which the callback was
+ called is locked. This patch corrects an instance in chan_agent
+ where a channel's indicate callback is called directly without
+ first locking the channel. This was leading to some observed
+ locking issues in chan_local, but considering that all channel
+ drivers operate under the same expectations, the generic fix in
+ chan_agent is the right way to go. AST-126 ........
+ ................
+
+2008-11-10 20:56 +0000 [r155764-155826] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/valgrind.txt, /: Merged revisions 155804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r155803 |
+ tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line
+ I got tired of saying this in every single bugnote referring to
+ this file. ........
+
+ * /, main/editline/readline.c: Merged revisions 155763 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r155763 | tilghman | 2008-11-10 12:04:30 -0600 (Mon, 10 Nov 2008)
+ | 6 lines Fix memory leak when MALLOC_DEBUG is enabled. (closes
+ issue #13864) Reported by: eliel Patches: readline.c.patch
+ uploaded by eliel (license 64) ........
+
+2008-11-09 16:32 +0000 [r155556-155672] Sean Bright <sean.bright@gmail.com>
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 155671 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r155671 | seanbright | 2008-11-09 11:30:29 -0500 (Sun,
+ 09 Nov 2008) | 1 line Fix this as well. Pointed out by tzafrir.
+ ........
+
+ * apps/app_followme.c, apps/app_queue.c, apps/app_dial.c, /,
+ main/features.c, include/asterisk/channel.h: Merged revisions
+ 155554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov
+ 2008) | 14 lines Merged revisions 155553 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov
+ 2008) | 6 lines Use static functions here instead of nested ones.
+ This requires a small change to the ast_bridge_config struct as
+ well. To understand the reason for this change, see the following
+ post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
+ ........ ................
+
+2008-11-08 21:48 +0000 [r155515-155517] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c, include/asterisk/strings.h: Merged
+ revisions 155516 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r155516 |
+ russell | 2008-11-08 15:46:43 -0600 (Sat, 08 Nov 2008) | 3 lines
+ - Check for failure when putting the packet in the ast_str - fix
+ a spelling error in a header file ........
+
+ * /, channels/chan_sip.c: Merged revisions 155513 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r155513 |
+ russell | 2008-11-08 15:34:36 -0600 (Sat, 08 Nov 2008) | 3 lines
+ Remove some code that is basically a no-op. Code above this
+ already ensures that the buffer is terminated. ........
+
+2008-11-07 23:42 +0000 [r155469] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 155467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 |
+ mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12
+ lines Set the invite state to INV_CANCELLED in a place that makes
+ more sense. Where it was set before, it was impossible to
+ actually delay sending a CANCEL if we had not yet received a
+ provisional response to an INVITE. (closes issue #13626) Reported
+ by: atis Patches: 13626.patch uploaded by putnopvut (license 60)
+ Tested by: atis ........
+
+2008-11-07 22:29 +0000 [r155396-155400] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 155399 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008)
+ | 14 lines Merged revisions 155398 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008)
+ | 7 lines Clarify error message. (closes issue #13809) Reported
+ by: denke Patches: 20081104__bug13809.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: denke ........ ................
+
+ * /, funcs/func_odbc.c: Merged revisions 155395 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r155395 |
+ tilghman | 2008-11-07 16:03:50 -0600 (Fri, 07 Nov 2008) | 2 lines
+ Two bugs relating to colnames found by Marquis42 on #asterisk-dev
+ ........
+
+2008-11-07 21:16 +0000 [r155362] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 155360 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri,
+ 07 Nov 2008) | 8 lines Remove one more instance of the sample
+ configuration lying about what's possible. The tz cannot be set
+ in a context like this. It can only be set in the general section
+ or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing
+ this out ........
+
+2008-11-07 20:19 +0000 [r155325] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 155324 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r155324 | tilghman | 2008-11-07 14:13:32 -0600 (Fri, 07 Nov 2008)
+ | 7 lines Send call release with unallocated cause instead of
+ normal call clearing, when invalid extension is called. (closes
+ issue #13408) Reported by: adomjan Patches:
+ chan_dahdi.c-ss7-unallocated-2 uploaded by adomjan (license 487)
+ ........
+
+2008-11-07 15:43 +0000 [r155242-155272] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 155264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r155264 |
+ russell | 2008-11-07 09:42:04 -0600 (Fri, 07 Nov 2008) | 3 lines
+ Remove a bogus ast_free() that Kevin noticed. This was probably
+ just left over from pre-astobj2ified chan_sip. ........
+
+ * /, include/asterisk/astobj2.h: Merged revisions 155244 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r155244 | russell | 2008-11-07 09:01:02 -0600 (Fri, 07
+ Nov 2008) | 4 lines Clarify which part of OBJ_MULTIPLE is not
+ implemented, and under what case it is perfectly fine to use.
+ (Inspired by a question I received about my last commit.)
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 155241 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r155241 |
+ russell | 2008-11-07 08:50:30 -0600 (Fri, 07 Nov 2008) | 4 lines
+ Fix some code in chan_sip that was intended to unlink multiple
+ objects from a container. The OBJ_MULTIPLE flag must be provided
+ here. Otherwise, this would only remove a single object. ........
+
+2008-11-06 22:49 +0000 [r155117-155122] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/ael/ael.flex, /, res/ael/ael_lex.c, utils/extconf.c: Merged
+ revisions 155121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r155121 |
+ kpfleming | 2008-11-06 16:49:19 -0600 (Thu, 06 Nov 2008) | 3
+ lines don't blindly assume that Darwin and Cygwin need
+ GLOB_ABORTED defined; only define it if it is not already defined
+ ........
+
+ * configure, configure.ac: ensure that an adequately new version of
+ libpri is in place so that chan_dahdi will compile with PRI
+ support
+
+2008-11-06 19:48 +0000 [r155014] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 155012 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r155012 | mmichelson | 2008-11-06 13:46:53 -0600
+ (Thu, 06 Nov 2008) | 16 lines Merged revisions 155011 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov
+ 2008) | 8 lines The documentation listed the ability to set
+ 'maxmsg' per context. The truth is that you can only set this in
+ the general section or per mailbox. Thus I am updating the sample
+ config file to be more accurate. Thanks to sasargen on IRC for
+ bringing up this issue. ........ ................
+
+2008-11-05 22:02 +0000 [r154920] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk.h, /: Merged revisions 154919 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r154919 |
+ seanbright | 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) | 2
+ lines Fix a problem found while building res_snmp. ........
+
+2008-11-05 22:00 +0000 [r154917] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 154428 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r154428 | tilghman | 2008-11-04 17:03:00 -0600 (Tue, 04 Nov 2008)
+ | 7 lines Switch to using a thread condition to signal that a
+ child thread is ready for work, rather than a busy wait. (closes
+ issue #13011) Reported by: jpgrayson Patches:
+ chan_iax2_find_idle.patch uploaded by jpgrayson (license 492)
+ ........
+
+2008-11-05 16:14 +0000 [r154690] Steve Murphy <murf@digium.com>
+
+ * main/channel.c, /: Merged revisions 154687 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r154687 | murf | 2008-11-05 09:11:11 -0700 (Wed, 05 Nov 2008) | 9
+ lines Merged revisions 154685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1
+ line This fix was prompted by communication from user, who was
+ seeing thousands of error logs... looks like EAGAIN. Made such
+ uninteresting. ........ ................
+
+2008-11-04 20:52 +0000 [r154367] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 154366 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r154366 | tilghman | 2008-11-04 14:51:18 -0600
+ (Tue, 04 Nov 2008) | 16 lines Merged revisions 154365 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008)
+ | 9 lines On busy systems, it's possible for the values checked
+ within a single line of code to change, unless the structure is
+ locked to ensure a consistent state. (closes issue #13717)
+ Reported by: kowalma Patches: 20081102__bug13717.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: kowalma ........
+ ................
+
+2008-11-04 19:09 +0000 [r154269] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 154268 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r154268 | rmudgett | 2008-11-04 13:07:26 -0600
+ (Tue, 04 Nov 2008) | 11 lines Merged revisions 154266 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008)
+ | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state
+ when it receives the indication AST_CONTROL_RINGING. ........
+ ................
+
+2008-11-04 19:02 +0000 [r154024-154267] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_h323.c: Merged revisions 154264 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r154264 | tilghman | 2008-11-04 12:59:48 -0600
+ (Tue, 04 Nov 2008) | 10 lines Recorded merge of revisions 154263
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008)
+ | 3 lines Make the monitor thread non-detached, so it can be
+ joined (suggested by Russell on -dev list). ........
+ ................
+
+ * apps/app_voicemail.c, /: Recorded merge of revisions 154072 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r154072 | tilghman | 2008-11-03 16:28:12 -0600
+ (Mon, 03 Nov 2008) | 12 lines Merged revisions 154066 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008)
+ | 5 lines Attempting to expunge a mailbox when the mailstream is
+ NULL will crash Asterisk. (Closes issue #13829) Reported by:
+ jaroth Patch by: me (modified jaroth's patch) ........
+ ................
+
+ * main/rtp.c, /: Merged revisions 154060 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008)
+ | 3 lines Remove the potential for a division by zero error.
+ (Closes issue #13810) ........
+
+ * /, funcs/func_odbc.c: Recorded merge of revisions 154023 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r154023 | tilghman | 2008-11-03 15:01:30 -0600 (Mon, 03
+ Nov 2008) | 4 lines Should have passed the string pointer, not
+ the ast_str structure. (closes issue #13830) Reported by: Marquis
+ ........
+
+2008-11-03 00:21 +0000 [r153710-153711] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/compiler.h, apps/app_stack.c,
+ include/asterisk/agi.h, configure,
+ include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4,
+ configure.ac: Merged revision 153709 from trunk
+ ------------------------------------------------------------------------
+ r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov
+ 2008) | 3 lines instead of trying to forcibly load res_agi when
+ app_stack is loaded (even if the administrator didn't want it
+ loaded), use GCC weak symbols to determine whether it was loaded
+ already or not; if it was loaded, then use it.
+ ------------------------------------------------------------------------
+
+ * channels/chan_iax2.c, res/res_jabber.c, channels/chan_oss.c,
+ utils/stereorize.c, main/channel.c, main/manager.c,
+ res/ael/ael_lex.c, main/file.c, pbx/pbx_dundi.c,
+ formats/format_gsm.c, main/asterisk.c, utils/muted.c, /,
+ formats/format_wav.c, apps/app_authenticate.c,
+ res/res_phoneprov.c, res/res_crypto.c, utils/astman.c,
+ res/res_musiconhold.c, res/res_http_post.c, apps/app_queue.c,
+ res/res_config_sqlite.c, agi/eagi-sphinx-test.c, utils/frame.c,
+ channels/chan_dahdi.c, res/ael/ael.tab.c, funcs/func_odbc.c,
+ main/ast_expr2f.c, res/res_agi.c, main/http.c, main/logger.c,
+ channels/chan_h323.c, apps/app_sms.c, res/ael/ael.flex,
+ pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c,
+ utils/streamplayer.c, apps/app_adsiprog.c, apps/app_voicemail.c,
+ apps/app_dial.c, channels/chan_sip.c, apps/app_festival.c,
+ main/db1-ast/hash/hash_page.c, res/ael/ael.y, agi/eagi-test.c,
+ pbx/pbx_lua.c, formats/format_ogg_vorbis.c, main/utils.c,
+ utils/astcanary.c, formats/format_wav_gsm.c: import gcc 4.3.2
+ warning fixes from trunk, with a few changes specific to this
+ branch
+
+2008-11-02 20:07 +0000 [r153363-153653] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/features.h, /: Merged revisions 153652 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r153652 | russell | 2008-11-02 14:06:03 -0600
+ (Sun, 02 Nov 2008) | 10 lines Merged revisions 153651 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r153651 | russell | 2008-11-02 13:51:17 -0600 (Sun, 02 Nov 2008)
+ | 2 lines features.h depends on linkedlists.h, so include it
+ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 153362 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r153362 |
+ russell | 2008-11-01 15:41:38 -0500 (Sat, 01 Nov 2008) | 3 lines
+ Ensure that the sip_pvt properly has its refcount incremented
+ when the scheduler holds a reference to it for session timer
+ processing. ........
+
+2008-10-31 22:11 +0000 [r153266] Terry Wilson <twilson@digium.com>
+
+ * apps/app_followme.c, apps/app_queue.c, apps/app_dial.c, /,
+ main/features.c, include/asterisk/channel.h: Merged revisions
+ 153181 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r153181 |
+ twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines
+ Recent CDR fixes moved execution of the 'h' exten into the
+ bridging code, so variables that were set after ast_bridge_call
+ was called would not show up in the 'h' exten. Added a callback
+ function to handle setting variables, etc. from w/in the bridging
+ code. Calls back into a nested function within the function
+ calling ast_bridge_call (closes issue #13793) Reported by:
+ greenfieldtech ........
+
+2008-10-31 20:10 +0000 [r153225] Mark Michelson <mmichelson@digium.com>
+
+ * main/dial.c, include/asterisk/dial.h: This commit contains the
+ bug fixes and documentation updates which were committed to trunk
+ in revision 153223. I blocked that commit from 1.6.1 since it
+ also contained a new feature. Note to self: Separate commits so
+ that you don't end up with a situation where part of a commit
+ should be merged but part should be blocked from stable branches.
+
+2008-10-31 16:36 +0000 [r153123] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Turn off qualify on uncached realtime
+ peers. (Closes issue #13383)
+
+2008-10-30 21:01 +0000 [r152995] Sean Bright <sean.bright@gmail.com>
+
+ * bootstrap.sh: The -I argument to aclocal needs a space before
+ the include directory name.
+
+2008-10-30 20:36 +0000 [r152924-152974] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_h323.c: Cannot join detached threads. See
+ http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
+ (Closes issue #13400)
+
+ * channels/chan_local.c: Unlock before returning, when extension
+ doesn't exist. (closes issue #13807) Reported by: eliel Patches:
+ chan_local.c.patch uploaded by eliel (license 64)
+
+2008-10-30 19:41 +0000 [r152878-152921] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix the sip_peer reference count with
+ respect to scheduler entries for scheduling peer pokes, and
+ scheduling peer poke expirations.
+
+ * channels/chan_sip.c: Fix the sip_peer reference count with
+ respect to scheduler entries for registration expirations.
+
+ * include/asterisk/sched.h: Fix a bug in AST_SCHED_REPLACE_UNREF().
+ The reference count of the object _must_ be increased before
+ creating the scheduler entry. Otherwise, you create a race
+ condition where the reference count may hit zero and the object
+ can disappear out from under you. This could also would have
+ incorrectly decreased the reference count in the case that the
+ scheduler add failed.
+
+ * channels/chan_sip.c: Modify the documentation of the sip_registry
+ struct - Remove a comment that says that the monitor thread is the
+ only one that ever touches these objects. This is no longer the
+ case with TCP. Also, I would eventually like to get the scheduler
+ in its own thread, so this is just a poor assumption to make. -
+ Note that reference counting of these objects with respect to
+ scheduler entries is not complete. There are some leaked
+ references when deleting scheduler entries.
+
+2008-10-30 16:55 +0000 [r152814] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/cdr.c: instead of comparing the string pointer to 0,
+ let's compare the value that was actually parsed out of the
+ string (found by sparse)
+
+2008-10-30 04:29 +0000 [r152690-152777] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample: Set up an example stdexten that
+ preserves the original context and extension in the CDR. (Related
+ to issue #13799) Reported by: davidw
+
+ * main/pbx.c: Track down and fix annoying lock errors. These would
+ occur when merging hints that resulted from a pattern matched hint
+ during a 'dialplan reload'.
+
+2008-10-29 20:55 +0000 [r152648] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_directory.c: If there was no named defined in a
+ voicemail.conf mailbox entry, then app_directory would crash when
+ attempting to read that entry from the file. We now check for the
+ NULL or empty string properly so that there will be no crash.
+ (closes issue #13804) Reported by: bluecrow76
+
+2008-10-29 20:16 +0000 [r152645] Terry Wilson <twilson@digium.com>
+
+ * apps/app_queue.c: Small modification to putnopvut's patch to fix
+ this issue. Thanks for all the help, putnopvut! (closes issue
+ #12884) Reported by: bcnit Patches: 12884v4-1.6.0-branch.patch
+ uploaded by otherwiseguy (license 396) Tested by: otherwiseguy
+
+2008-10-29 05:52 +0000 [r152606] Steve Murphy <murf@digium.com>
+
+ * apps/app_queue.c, configs/features.conf.sample, apps/app_dial.c:
+ A little documentation cross-ref between features and dial and
+ queue... I wasted some time (stupidly) trying to get the
+ one-touch parking stuff working, because it didn't occur to me
+ that I had to also have the corresponding options in the dial
+ command! Duh! (In all this time, I never set this up before!) So,
+ to keep some poor fool from suffering the same fate, I made the
+ features.conf.sample file mention the corresponding opts in
+ dial/queue; and the docs for dial/app specifically mention the
+ corresponding decls in the feature.conf file. I hope this doesn't
+ spoil some vast, eternal plan...
+
+2008-10-29 05:35 +0000 [r152573] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes
+ issue #13795) Reported by: andrew53 Patches:
+ chan_sip_sizeof.patch uploaded by andrew53 (license 519)
+
+2008-10-29 05:09 +0000 [r152537] Steve Murphy <murf@digium.com>
+
+ * apps/app_queue.c, include/asterisk/features.h, apps/app_dial.c,
+ main/features.c, include/asterisk/pbx.h: The magic trick to avoid
+ this crash is not to try to find the channel by name in the list,
+ which is slow and resource consuming, but rather to pay attention
+ to the result codes from the ast_bridge_call, to which I added the
+ AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when
+ a channel is parked. Why? because CDR's aren't generated via
+ parking, so nothing is needed, but if a transfer occurred, there
+ are critical things I need. If you get AST_PBX_KEEPALIVE, then
+ don't touch the channel pointer. If you get
+ AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then
+ don't touch the peer pointer. Updated the several places where
+ the results from a bridge were not being properly obeyed, and
+ fixed some code I had introduced so that the results of the
+ bridge were not overridden (in trunk). All the places that
+ previously tested for AST_PBX_NO_HANGUP_PEER now have to check
+ for both AST_PBX_NO_HANGUP_PEER and
+ AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common
+ parking scenarios: 1. A calls B; B answers; A parks B; B hangs up
+ while A is getting the parking slot announcement, immediately
+ after being put on hold. 2. A calls B; B answers; A parks B; B
+ hangs up after A has been hung up, but before the park times out.
+ 3. A calls B; B answers; B parks A; A hangs up while B is getting
+ the parking slot announcement, immediately after being put on
+ hold. 4. A calls B; B answers; B parks A; A hangs up after B has
+ been hung up, but before the park times out. No crash. I also ran
+ the scenarios above against valgrind, and accesses looked good.
+
+2008-10-28 22:35 +0000 [r152370-152471] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Quoting in the wrong direction (Fixes
+ AST-107)
+
+ * channels/chan_mgcp.c: Only re-add the io port if it was closed,
+ otherwise reload causes a memory leak. (closes issue #13785)
+ Reported by: eliel Patches: chan_mgcp.c.patch uploaded by eliel
+ (license 64)
+
+ * apps/app_dial.c: Reset all DIAL variables back to blank, in case
+ Dial is called multiple times per call (which could otherwise
+ lead to inconsistent status reports). (closes issue #13216)
+ Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded
+ by Corydon76 (license 14) Tested by: ruddy
+
+2008-10-27 23:32 +0000 [r152288] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Buffer policy setting for half is not
+ needed.
+
+2008-10-27 21:53 +0000 [r152173-152217] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: Inherit ALL elements of CallerID across a
+ local channel. (closes issue #13368) Reported by: Peter Schlaile
+ Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
+ (license 14)
+
+ * apps/app_stack.c: Oops, only delete the ARG variables once upon
+ release. The following section would have removed them again
+ (removing variables from 2 stack frames, instead of just one).
+
+2008-10-27 16:06 +0000 [r152133] Jason Parker <jparker@digium.com>
+
+ * apps/app_transfer.c: Remove options argument parsing/syntax (it
+ isn't used any longer) (closes issue #13789) Reported by: IgorG
+ Patches: app_transfer.c.diff uploaded by IgorG (license 20)
+
+2008-10-26 20:27 +0000 [r152068] Sean Bright <sean.bright@gmail.com>
+
+ * funcs/func_strings.c: Since passing \0 as the second argument to
+ strchr is valid (and will match the trailing \0 of a string) we
+ need to check that first, otherwise we end up with incorrect
+ results. Fix suggested by reporter. (closes issue #13787)
+ Reported by: meitinger
+
+2008-10-25 11:11 +0000 [r151907] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Move AMI initialization to occur after loading
+ modules. This prevents a deadlock when someone tries to
+ initiate a module reload from the AMI just as Asterisk is
+ starting. (closes issue #13778) Reported by: hotsblanc Fix
+ suggested by hotsblanc
+
+2008-10-22 20:08 +0000 [r151603] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/live_ast: Add a contributed script for running
+ Asterisk without installing it, first. (closes issue #11680)
+ Reported by: tzafrir Patches: live_ast_6 uploaded by tzafrir
+ (license 46)
+
+2008-10-22 20:05 +0000 [r151421-151602] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c: Change some logical ands to bitwise ands
+ and add messages alerting that a channel is being ignored if the
+ PROC_DAHDI_NOCHAN option is set in process_dahdi. (closes issue
+ #13759) Reported by: smurfix Patches: dahdi.patch uploaded by
+ smurfix (license 547)
+
+ * channels/chan_sip.c: The logic of a strncasecmp call was reversed.
+ (closes issue #13706) Reported by: andrew53 Patches:
+ sip_notify_from_rfc3265.patch uploaded by andrew53 (license 519)
+
+ * channels/chan_sip.c: Make the sip_standard_port function more
+ granular by allowing separate type and port arguments. This is
+ necessary because when building our From and Contact headers, we
+ need to be absolutely sure that we are placing our source port
+ there and not the peer's source port. (closes issue #12761)
+ Reported by: asbestoshead Patches: patch-chan-sip-contact-port.txt
+ uploaded by asbestoshead (license 455)
+
+ * channels/chan_sip.c: Get this compiling in dev-mode
+
+ * channels/chan_sip.c: If a peer uses any transport other than UDP,
+ then MWI will fail for that peer since sip_alloc will allocate a
+ sip_pvt with a default transport of UDP. This change resets the
+ socket type immediately after allocating the sip_pvt in
+ sip_send_mwi_from_peer, so that the proceeding call to
+ create_addr_from_peer does not fail right away. The socket data
+ from the peer is properly copied to the sip_pvt in
+ create_addr_from_peer. (closes issue #13710) Reported by:
+ andrew53 Patches: sip_notify_use_tcp.patch uploaded by andrew53
+ (license 519)
+
+ * channels/chan_sip.c: When attempting to resolve hostnames, we
+ need to be sure to remove any parameters from the string so that
+ name resolution succeeds. (closes issue #13727) Reported by:
+ fnordian Patches: resolvewithouturiparameter.patch uploaded by
+ fnordian (license 110)
+
+2008-10-21 15:21 +0000 [r151372] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_mixmonitor.c: Default file modes should always be full
+ read and write, to allow the system administrator to make the
+ decision of what permissions will actually be given, through the
+ use of the process umask. (Closes issue# 13751)
+
+2008-10-21 11:03 +0000 [r151328] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_sip.c: Fix configuration parsing so type=friend
+ still identifies "friend" as a peer even though it is now a
+ legacy configuration verb. (closes issue #13705) reported by:
+ blitzrage patched by: bweschke
+
+2008-10-20 05:06 +0000 [r151135-151245] Kevin P. Fleming <kpfleming@digium.com>
+
+ * autoconf (added), autoconf/ast_check_pwlib.m4,
+ autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure,
+ autoconf/ast_gcc_attribute.m4, bootstrap.sh,
+ autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4,
+ autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4,
+ autoconf/ast_c_define_check.m4, autoconf/ast_prog_egrep.m4,
+ autoconf/ast_ext_tool_check.m4, autoconf/ast_check_mandatory.m4,
+ autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4,
+ configure.ac, acinclude.m4 (removed), autoconf/ast_prog_sed.m4:
+ break up acinclude.m4 into individual files, which will make it
+ easier to maintain, easier to add new macros (less patching) and
+ will ease maintenance of these macros across Asterisk branches.
+ Rename this macro to properly reflect what it does
+
+ * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
+ apps/app_externalivr.c, include/asterisk/tcptls.h: cleaup of the
+ TCP/TLS socket API: 1) rename 'struct server_args' to 'struct
+ ast_tcptls_session_args', to follow coding guidelines 2) make
+ ast_make_file_from_fd() static and rename it to something that
+ indicates what it really is for (again coding guidelines) 3)
+ rename address variables inside 'struct ast_tcptls_session_args'
+ to be more descriptive (dare i say it... coding guidelines) 4)
+ change ast_tcptls_client_start() to use the new 'remote_address'
+ field of the session args for the destination of the connection,
+ and use the 'local_address' field to bind() the socket to the
+ proper source address, if one is supplied 5) in chan_sip, ensure
+ that we pass in the PP address we are bound to when creating
+ outbound (client) connections, so that our connections will
+ appear from the correct address
+
+2008-10-18 02:29 +0000 [r150829] BJ Weschke <bweschke@btwtech.com>
+
+ * main/manager.c: Using the GetVar handler in AMI is potentially
+ dangerous (insta-crash [tm]) when you use a dialplan function
+ that requires a channel and then you don't provide one or provide
+ an invalid one in the Channel: parameter. We'll handle this
+ situation exactly the same way it was handled in pbx.c back on
+ r61766. We'll create a bogus channel for the function call and
+ destroy it when we're done. If we have trouble allocating the
+ bogus channel then we're not going to try executing the function
+ call at all and run the risk of crashing. (closes issue #13715)
+ reported by: makoto patch by: bweschke
+
+2008-10-17 17:10 +0000 [r150606-150636] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Make helper call a little safer (suggested
+ by Russell on IRC)
+
+ * channels/chan_iax2.c, include/asterisk/sched.h: Fix the FRACK!
+ warnings in chan_iax2 when POKE/LAGRQ packets are not answered.
+
+2008-10-16 23:41 +0000 [r150208-150306] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c: Reverting changes from commits 150298 and 150301
+ since I was mistakenly under the assumption that dialplan
+ functions *always* required that a channel be present. I need to
+ go home earlier, I think :)
+
+ * main/manager.c: Don't try to call a dialplan function's read
+ callback from the manager's GetVar handler if an invalid channel
+ has been specified. Several dialplan functions, including CHANNEL
+ and SIP_HEADER, do not check for NULL-ness of the channel being
+ passed in. (closes issue #13715) Reported by: makoto
+ And don't forget to return on the error condition
+
+ * apps/app_sms.c: Answer the channel prior to checking for the 'a'
+ option in app_sms. (closes issue #13675) Reported by: alecdavis
+ Patches: app_sms.bug13675.148985.diff.txt uploaded by alecdavis
+ (license 585)
+
+ * configure, configure.ac: Change configure script to search for
+ openais in both /usr/lib and /usr/lib64 since some distros place
+ 64-bit libraries only in the /usr/lib64 directory. (closes issue
+ #13721) Reported by: jcollie Patches:
+ 0007-Look-in-64bit-dirs-for-openais.patch uploaded by jcollie
+ (license 412)
+
+ * channels/chan_sip.c: INVITES with proxy auth were sent with a
+ different branch than what was in the invite_branch of a sip_pvt,
+ meaning that if a CANCEL were sent later, the branch in the
+ CANCEL would not match the branch in the latest INVITE sent out,
+ leading to some endpoints responding to the CANCEL with a 481.
+ (closes issue #13714) Reported by: fnordian Patches:
+ invite_branch.patch uploaded by fnordian (license 110)
+
+2008-10-16 16:17 +0000 [r150127] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Fix memory leak found by customer
+
+2008-10-16 13:32 +0000 [r149919-149995] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: return this logic to where it used to be,
+ *after* the dialog->needdestroy flag has been determined to be set;
+ otherwise, we generate these debug messages every time we inspect
+ every active dialog
+
+ * apps/app_stack.c: building this module depends on res_agi being
+ built as well
+
+ * res/res_phoneprov.c: inter-module dependencies should be included
+ in the source code, not just in sample config files
+
+ * res/res_phoneprov.c: correct file name in message
+
+2008-10-15 21:00 +0000 [r149803] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Make the sip_proxy struct reference counted.
+ This is necessary to allow for a sip_pvt to maintain a reference
+ to a sip_peer's outboundproxy even after the peer has been freed.
+ (closes issue #13700) Reported by: fnordian Patches: 13700.patch
+ uploaded by putnopvut (license 60) Tested by: fnordian
+
+2008-10-15 20:22 +0000 [r149758] BJ Weschke <bweschke@btwtech.com>
+
+ * configs/agents.conf.sample: An update to the documentation/example
+ of agents.conf.sample with the correct parameter for this feature
+ as defined in chan_agent.c (closes issue #13709)
+
+2008-10-15 19:09 +0000 [r149589-149688] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: Permit data fields to contain more than 255
+ characters. (closes issue #13631) Reported by: seanbright Patches:
+ 20081015__bug13631.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: blitzrage
+
+ * funcs/func_odbc.c: Only set buf to blank before the goto.
+
+ * codecs/lpc10/lpcini.c: When using MALLOC_DEBUG, codec_lpc10 leaks
+ memory, because it matches a library malloc() with an ast_free
+ (which, of course, doesn't match up with known allocated memory,
+ so the free fails). (closes issue #13702) Reported by: eliel
+ Patches: codec_lpc10_lpcini.c uploaded by eliel (license 64)
+
+ * apps/app_echo.c: Minor spacing change (closes issue #13697)
+ Reported by: alecdavis Patches: app_echo.bug13697.103249.diff.txt
+ uploaded by alecdavis (license 585)
+
+2008-10-15 11:32 +0000 [r149512] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: fix some problems when parsing SIP messages
+ that have the maximum number of headers or body lines that we
+ support
+
+2008-10-14 23:58 +0000 [r149203-149280] Mark Michelson <mmichelson@digium.com>
+
+ * CHANGES, apps/app_dial.c: When specifying an invalid timeout to
+ Dial, take it to mean that no timeout is desired. (closes issue
+ #13625) Reported by: atis
+
+ * channels/chan_sip.c: Change this warning to an error message.
+ Suggestion comes from Sean Bright. Thanks Sean!
+
+ * channels/chan_sip.c: Call register_peer_exten even in the case
+ that the peer's IP/port does not change. (closes issue #13309)
+ Reported by: dimas Patches: v2-13309.patch uploaded by dimas
+ (license 88)
+
+ * include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance
+ period for sync-triggered audiohooks so that if packetization of
+ audio is close (but not equal) we don't end up flushing the
+ audiohooks over small inconsistencies in synchronization. Related
+ to issue #13005, and solves the issue for most people who were
+ experiencing the problem. However, a small number of people are
+ still experiencing the problem on long calls, so I am not
+ closing the issue yet
+
+ * apps/app_queue.c: Update the queue with the correct number of
+ calls and whether the call was completed within the service level
+ when a transfer takes place. This way, we do not "break" the
+ leastrecent and fewestcalls strategies by not logging a call
+ until after the transferred call has ended. (closes issue #13395)
+ Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded
+ by Marquis (license 32)
+
+2008-10-14 22:42 +0000 [r149202] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/hashtab.h, main/chanvars.c, main/config.c,
+ main/hashtab.c, pbx/pbx_spool.c, channels/chan_sip.c,
+ include/asterisk/chanvars.h, include/asterisk/config.h,
+ include/asterisk/strings.h, res/res_indications.c: Add additional
+ memory debugging to several core APIs, and fix several memory
+ leaks found with these changes. (Closes issue #13505, closes
+ issue #13543) Reported by: mav3rick, triccyx Patches:
+ 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: mav3rick, triccyx
+
+2008-10-14 21:09 +0000 [r149132] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Don't allow reserved characters to be used in
+ register lines in sip.conf. (closes issue #13570) Reported by:
+ putnopvut
+
+2008-10-14 20:17 +0000 [r149063] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_waitforsilence.c: Check correct values in the return of
+ ast_waitfor(); also, get rid of a possible memory leak. (closes
+ issue #13658) Reported by: explidous Patch by: me
+
+2008-10-14 19:42 +0000 [r149060] Leif Madsen <lmadsen@digium.com>
+
+ * doc/manager_1_1.txt: Add missing documentation for SipShowRegistry
+ action and RegistryEntry event. (closes issue #13342) Reported and
+ patch by: Laureano
+
+2008-10-14 18:59 +0000 [r148918-148986] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_sms.c: App is ignoring 'p' parameter -- initial pause.
+ (closes issue #13617) Reported by: alecdavis Patches:
+ app_sms.13oct.diff.txt uploaded by alecdavis (license 585)
+
+ * apps/app_voicemail.c: Ensure that mail headers are 7-bit clean,
+ even when UTF-8 characters are used in headers like 'Subject' and
+ 'To'. Closes AST-107.
+
+2008-10-14 17:39 +0000 [r148915] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Deadlock prevention in chan_local. (closes
+ issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded by
+ putnopvut (license 60) Tested by: tacvbo
+
+2008-10-14 15:18 +0000 [r148869] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_fax.c: API differences in spandsp 0.0.6pre1 and higher
+ (closes issue #13688) Reported by: irroot Patches:
+ app_fax-span6.patch uploaded by irroot (license 52) with minor
+ modifications by me
+
+2008-10-14 11:35 +0000 [r148614-148763] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: fix some references to the owner of a
+ private structure that may not be present
+
+ * Makefile: on Ubuntu (at least), recent versions of ld in binutils
+ delete all debugging symbols when -x is supplied; since the
+ reasons why -x is being passed are lost in the mists of time,
+ remove it so debugging will work properly
+
+ * channels/chan_sip.c: ensure that *all* fields in the req
+ structure are cleared out before reusing it; has_to_tag was not
+ cleared, which caused the second incoming call over a TCP socket
+ to fail if pedantic checking was enabled
+
+ * main/translate.c: it would be nice if this message printing code
+ had actually been tested before it was committed...
+
+2008-10-13 17:56 +0000 [r148562] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: Hmmm. Nobody (but me) is interested in seeing the
+ trie info when they do 'dialplan show ...' (even with debug set
+ to non-zero); so I set up a 'dialplan debug [context]' cli
+ command instead, to explicitly show just the trie info. I even
+ added an extension_exists() call to make sure the trie info is
+ built. I moved the explanatory header to above the extension loop
+ to ensure it only prints once. And it will do this now, whether
+ debug is set or not. I removed the trie printing from the
+ 'dialplan show' command entirely.
+
+2008-10-13 15:36 +0000 [r148472] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Sending a 403 after a 200 is considered very
+ bad. (found at SIPit)
+
+2008-10-10 21:22 +0000 [r148375-148377] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: The logic used when checking a peer got
+ changed subtly in the "kill the user" commit and caused calls
+ relying on the insecure setting to not work properly. I changed
+ for finding a peer back to how it was prior to that commit.
+ (closes issue #13644) Reported by: pj Patches: 13644_trunkv2.patch
+ uploaded by putnopvut (license 60) Tested by: pj
+
+ * channels/chan_sip.c: Make sure that the inUse and inRinging
+ fields for a sip peer cannot go below zero. This is a
+ regression from 1.4 and so it will be applied to 1.6.0 as
+ well. (closes issue #13668) Reported by: mjc
+
+2008-10-10 16:37 +0000 [r148269] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: User not notified of temporary greeting, if
+ ODBC storage is in use. (closes issue #13659) Reported by:
+ moliveras Patches: 20081009__bug13659.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: moliveras
+
+2008-10-10 01:33 +0000 [r148240] Sean Bright <sean.bright@gmail.com>
+
+ * res/res_config_sqlite.c, apps/app_voicemail.c,
+ include/asterisk.h, main/tdd.c, main/cryptostub.c: Don't include
+ logger.h in asterisk.h by default as it is causing problems
+ building app_voicemail. Instead, include it where it is needed.
+ This turned out to be a relatively minor issue because other
+ headers include logger.h as well. Need to test -addons before
+ merging this back to 1.6.0. (closes issue #13605) Reported by:
+ tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright
+ (license 71) Tested by: mmichelson
+
+2008-10-09 23:55 +0000 [r148151-148161] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c: The priority was unnecessary for the manager
+ atxfer, so it has been removed. Furthermore, now we actually use
+ the Context argument passed to set the transfer context and don't
+ error out if no context is specified. This addresses the actual
+ problems outlined in issue 12158. Regarding the other points
+ brought up, regarding the inability to not transfer to extensions
+ which cannot be represented by DTMF, it is not enough of a
+ constraint that it is worth attempting to rework the feature.
+ (closes issue #12158) Reported by: davidw
+
+ * apps/app_voicemail.c: Read the callerid in the correct order and
+ make sure to read the Urgent flag value from the IMAP headers.
+ (closes issue #13652) Reported by: jaroth Patches:
+ imapheaders.patch uploaded by jaroth (license 50)
+
+2008-10-09 23:27 +0000 [r148128] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/res_ldap.conf.sample: Fix example schema (closes issue
+ #12860) Reported by: flyn Patches: res_ldap.conf.patch uploaded
+ by flyn (license 503)
+
+2008-10-09 23:20 +0000 [r148115] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: (closes issue #13579) Reported by: dwagner
+ (closes issue #13584) Reported by: dwagner Tested by: murf,
+ putnopvut The thought occurred to me that the res= from the
+ extension spawn was ending up being returned from the bridge.
+ "Thou shalt not poison the return value". Made the change and it
+ appears to allow blind xfers to work as normal. If I'm wrong,
+ reopen the bugs. But it looks good to me! Many thanks to
+ putnopvut for helping me reproduce this!
+
+2008-10-09 20:01 +0000 [r148006-148011] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile, sounds/sounds.xml: Publish MOH files in sln16
+ format
+
+ * apps/app_voicemail.c: When blank, callerid name and number
+ should display "unknown caller" in voicemail emails. (Closes
+ issue #13643)
+
+2008-10-09 19:28 +0000 [r147957] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: (closes issue #13139) Reported by: krisk84
+ Tested by: krisk84 This change prevents a call that is placed in
+ the parkinglot to be picked up before the PBX is finished. If
+ another extension dials the parking extension before the PBX
+ thread has completed at minimum warnings will occur about the PBX
+ not properly being terminated. At worst, a crash could occur.
+
+2008-10-09 17:54 +0000 [r147901] Michiel van Baak <michiel@vanbaak.info>
+
+ * include/asterisk/endian.h: only include this for OpenBSD. At least
+ FreeBSD is borked when including it (closes issue #13649)
+ Reported by: ys
+
+2008-10-09 17:47 +0000 [r147898] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/extensions.conf.sample: Remove "second form" of
+ extensions, as it no longer applies. Also, cleanup the grammar,
+ formatting, and introduce several clarifications to the text.
+ (Closes issue #13654)
+
+2008-10-09 15:06 +0000 [r147811] Steve Murphy <murf@digium.com>
+
+ * channels/chan_iax2.c, main/astobj2.c, channels/chan_oss.c,
+ main/config.c, main/rtp.c, main/cli.c, channels/chan_usbradio.c,
+ configure, channels/console_gui.c, utils/extconf.c, main/pbx.c,
+ include/asterisk.h, doc/CODING-GUIDELINES,
+ include/asterisk/autoconfig.h.in, main/translate.c,
+ channels/vcodecs.c, configure.ac, channels/console_video.c:
+ (closes issue #13557) Reported by: nickpeirson Patches:
+ pbx.c.patch uploaded by nickpeirson (license 579)
+ replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
+ Tested by: nickpeirson, murf 1. replaced all refs to bzero and
+ bcopy to memset and memmove instead. 2. added a note to the
+ CODING-GUIDELINES 3. add two macros to asterisk.h to prevent
+ bzero, bcopy from creeping back into the source 4. removed bzero
+ from configure, configure.ac, autoconfig.h.in
+
+2008-10-08 22:33 +0000 [r147719] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_meetme.c: Some small tweaks regarding realtime conference
+ announcements. (closes issue #13522) Reported by: DEA Patches:
+ meetme-rt-fixes.txt uploaded by DEA (license 3)
+
+2008-10-08 22:27 +0000 [r147692] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: when parsing a text configuration option,
+ ensure that the buffer on the stack is actually large enough to
+ hold the legal values of that option, and also ensure that
+ sscanf() knows to stop parsing if it would overrun the buffer
+ (without these changes, specifying "buffers=...,immediate" would
+ overflow the buffer on the stack, and could not have worked as
+ expected)
+
+2008-10-08 19:09 +0000 [r147593] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_sms.c: Correct a typo in the help; also, ensure that the
+ date and time are correctly set, if not specified in the message.
+ (Closes issue #13594, closes issue #13595) Reported by: alecdavis
+ Patches: 20081001__bug13595.diff.txt uploaded by Corydon76
+ (license 14) Tested by: alecdavis
+
+2008-10-08 15:10 +0000 [r147519] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_speech_utils.c: If we receive DTMF make sure that the
+ state of the speech structure goes back to being not ready.
+ (issue #LUMENVOX-8)
+
+2008-10-07 16:54 +0000 [r147196] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_voicemail.c: Make 'imapsecret' an alias to
+ 'imappassword' in voicemail.conf.
+
+2008-10-07 16:05 +0000 [r147147] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Explicitly setting these fields to NULL was done
+ because I wasn't sure if they would be NULL otherwise. Since they
+ will be set automatically, removing.
+
+2008-10-07 15:06 +0000 [r147100] Richard Mudgett <rmudgett@digium.com>
+
+ * funcs/func_callerid.c: Independent change from branch issue8824
+ that is not part of COLP. (-r142574 rmudgett)
+
+2008-10-07 12:03 +0000 [r147052] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_dial.c: Make sure to compare the correct number of
+ characters when special-casing our DAHDI operator mode stuff.
+ Technically, it would work fine, as 'DAH' is currently unique
+ amongst our channel technologies, but as Jared points out:
+ <@jsmith> Sure... as long as the technology starts whith DAH....
+ but it could be DAHDOO!
+
+2008-10-07 00:13 +0000 [r146972] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: A blind transfer to the parking thread would
+ cause a segfault because copy_request accesses dst->data w/o
+ being able to tell whether it is proerly initialized
+
+2008-10-06 23:22 +0000 [r146930] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/threadstorage.h: Update documentation;
+ AST_THREADSTORAGE() in trunk only takes a single argument.
+
+2008-10-06 23:08 +0000 [r146876-146924] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/features.h, main/features.c, res/res_agi.c:
+ Similar to r143204, masquerade the channel in the case of Park
+ being called from AGI. ........
+
+ * include/asterisk/endian.h: Mvanbaak said this was needed to
+ compile on OpenBSD, so put it in the OpenBSD section.
+
+ * main/features.c: This commit squashes together three commits
+ because the wrong approach was originally used. (One of the
+ commits was only one line.) 1) r143204: The main change here was
+ to masquerade the channel if the channel that was to be parked
+ was running a PBX on it. The PBX thread can then maintain full
+ control of the channel (the zombie) as it expects to while
+ allowing the parking thread full control of the real (parked)
+ channel. 2) r143270: Changed park_call_full to hold the
+ parkinglot lock a little longer, which protects the parkeduser
+ struct from being freed out from underneath. Made sure that the
+ parking extension is added to the parking context while holding
+ the lock thereby ensuring that there are no spurious warnings
+ from removal attempts when a hangup occurs while the parking lot
+ is being announced. 3) r143475: (the one liner) compare peer and
+ chan instead of looking at the parked user (pu), which could have
+ possibly already have been freed by the parking thread
+
+ * main/features.c: fix some comment placement
+
+ * main/features.c: Explicitly set args in park_call_exec NULL so in
+ the case of no options being passed in, there is no garbage
+ attempted to be used. Also, do not set args to unknown value
+ again if there are no options passed in.
+
+2008-10-06 21:53 +0000 [r146874] Michiel van Baak <michiel@vanbaak.info>
+
+ * include/asterisk/endian.h: make aescrypt.c compile on OpenBSD again
+
+2008-10-06 21:32 +0000 [r146715-146838] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c, funcs/func_callerid.c,
+ apps/app_speech_utils.c, funcs/func_curl.c,
+ funcs/func_groupcount.c, res/res_smdi.c, channels/chan_sip.c,
+ funcs/func_timeout.c, funcs/func_odbc.c, funcs/func_cdr.c,
+ funcs/func_math.c: Dialplan functions should not actually return
+ 0, unless they have modified the workspace. To signal an error
+ (and no change to the workspace), -1 should be returned instead.
+ (closes issue #13340) Reported by: kryptolus Patches:
+ 20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
+
+ * channels/chan_local.c: Check whether an extension exists in the
+ _call method, rather than the _alloc method, because we need to
+ evaluate the callerid (since that data affects whether an
+ extension exists). (closes issue #13343) Reported by: efutch
+ Patches: 20080915__bug13343.diff.txt uploaded by Corydon76
+ (license 14) Tested by: efutch
+
+2008-10-06 16:39 +0000 [r146698] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: ensure that the private structure for
+ pseudo channels is created without 'leaking' configuration data
+ from other configured channels (closes issue #13555) Reported by:
+ jeffg Patches: issue_13555.patch uploaded by kpfleming (license
+ 421) Tested by: jeffg
+
+2008-10-06 00:23 +0000 [r146557] Sean Bright <sean.bright@gmail.com>
+
+ * utils/Makefile: Quote arguments to cp so we can handle spaces in
+ our paths.
+
+2008-10-05 21:24 +0000 [r146451] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Fix silly formatting.
+
+2008-10-04 01:57 +0000 [r146314] Sean Bright <sean.bright@gmail.com>
+
+ * configs/sip_notify.conf.sample: Add ability to remotely reboot
+ snom phones. Also cleaned up and reorganized
+ sip_notify.conf.sample a bit as well. Tested snom reboot on snom
+ 360 and verified snom-check-cfg worked as well. (closes issue
+ #13601) Reported by: mjc Tested by: seanbright
+
+2008-10-03 22:42 +0000 [r146243] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: remove superfluous reference counting operations
+ in manage_parkinglot since ao2_interator_next increments the ref
+ count automatically
+
+2008-10-03 22:13 +0000 [r146200] Sean Bright <sean.bright@gmail.com>
+
+ * main/cli.c: Resolve a subtle bug where we would never
+ successfully be able to get the first item in the CLI entry list.
+ This was preventing '!' from showing up in either 'help' or in tab
+ completion. (closes issue #13578) Reported by: mvanbaak
+
+2008-10-02 19:31 +0000 [r145960-145964] Russell Bryant <russell@digium.com>
+
+ * CHANGES: The 'P' command for ExternalIVR was also added in 1.6.0
+
+ * CHANGES: TCP support for ExternalIVR went in to 1.6.1, not 1.6.0
+
+2008-10-02 15:30 +0000 [r145781] Sean Bright <sean.bright@gmail.com>
+
+ * configure, configure.ac: This is much cleaner, methinks.
+
+2008-10-02 15:19 +0000 [r145754] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Some sanity checks that may have led to prior
+ crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup
+ of incorrectly-used constants.
+
+2008-10-01 23:54 +0000 [r145694] Sean Bright <sean.bright@gmail.com>
+
+ * configure, configure.ac: Try a test compile using the GMime
+ library. Some distros install gmime-config in the base package
+ instead of the -devel package. Now we print a notice and disable
+ GMime support instead of bombing during the main compilation.
+ (closes issue #13583) Reported by: arkadia
+
+2008-10-01 22:24 +0000 [r145557-145609] Mark Michelson <mmichelson@digium.com>
+
+ * main/features.c: Okay, this should really do it now. While I did
+ manage to fix blind transfers with my last commit here, I also
+ caused an unwanted side-effect. That is, only the first priority
+ of the 'h' extension would be executed when a blind transfer
+ occurred instead of all priorities. Essentially, my last commit
+ corrected the return value of ast_bridge_call. However, the
+ implementation still was not 100% correct. Now it is.
+
+ * main/features.c: if (!(x) == 0) is the same as if (x).
+
+ * main/features.c: The logic surrounding the return value of
+ ast_spawn_extension within ast_bridge_call was reversed. This
+ problem was observed when a blind transfer placed from the callee
+ channel of a test call failed. While the problem I am solving
+ here is exactly the same as what was reported in issue #13584,
+ the difference is that this fix I am applying is trunk-only.
+ Issue #13584 was reported against the 1.4 branch, and my tests of
+ 1.4's blind transfers appear to work fine.
+
+2008-10-01 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.6.0 released.
+
+2008-09-09 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.6.0-rc6 released.
+
+2008-09-09 15:44 +0000 [r142065] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 142064 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008)
+ | 13 lines Merged revisions 142063 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008)
+ | 5 lines Ensure that the stored CDR reference is still valid
+ after the bridge before poking at it. Also, keep the channel
+ locked while messing with this CDR. (fixes crashes reported in
+ issue #13409) ........ ................
+
+2008-09-09 12:34 +0000 [r141996-141999] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_oss.c, /: Merged revisions 141995 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r141995 |
+ mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8
+ lines Fix a memory leak in chan_oss (closes issue #13311)
+ Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel
+ (license 64) ........
+
+2008-09-09 01:49 +0000 [r141950] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 141949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 |
+ russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines
+ Modify ast_answer() to not hold the channel lock while calling
+ ast_safe_sleep() or when calling ast_waitfor(). These are
+ inappropriate times to hold the channel lock. This is what has
+ caused "could not get the channel lock" messages from chan_sip
+ and has likely caused a negative impact on performance results of
+ SIP in Asterisk 1.6. Thanks to file for pointing out this section
+ of code. (closes issue #13287) (closes issue #13115) ........
+
+2008-09-08 21:07 +0000 [r141808] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, /: Merged revisions 141807 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008)
+ | 15 lines Merged revisions 141806 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008)
+ | 7 lines When doing an async goto, detect if the channel is
+ already in the middle of a masquerade. This can happen when
+ chan_local is trying to optimize itself out. If this happens,
+ fail the async goto instead of bursting into flames. (closes
+ issue #13435) Reported by: geoff2010 ........ ................
+
+2008-09-08 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.6.0-rc5 released.
+
+2008-09-08 20:19 +0000 [r141746] Jason Parker <jparker@digium.com>
+
+ * Makefile, /, redhat (removed): Merged revisions 141745 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r141745 | qwell | 2008-09-08 15:18:17 -0500
+ (Mon, 08 Sep 2008) | 16 lines Merged revisions 141741 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) |
+ 8 lines Remove RPM package targets from Makefile (and all
+ associated parts). This has never worked in 1.4, and we decided
+ that it makes no sense to be done here. There are many distros
+ out there that already have "proper" spec files that can be
+ (re)used. Closes issue #13113 Closes issue #10950 Closes issue
+ #10952 ........ ................
+
+2008-09-08 17:14 +0000 [r141683] Sean Bright <sean.bright@gmail.com>
+
+ * /, build_tools/make_buildopts_h: Merged revisions 141682 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon,
+ 08 Sep 2008) | 9 lines Quote the arguments to grep so that sh on
+ various platforms doesn't choke on the special characters (like
+ ^). (closes issue #13417) Reported by: dougm Patches:
+ 13417.make_buildopts_h.patch uploaded by seanbright (license 71)
+ Tested by: dougm ........
+
+2008-09-06 20:21 +0000 [r141567] Steve Murphy <murf@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 141566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9
+ lines Merged revisions 141565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1
+ line This fix comes from Joshua Colp The Brilliant, who, given
+ the trace, came up with a solution. This will most likely will
+ close 13235 and 13409. I'll wait till Monday to verify, and then
+ close these bugs. ........ ................
+
+2008-09-06 15:40 +0000 [r141505-141508] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 141504 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008)
+ | 12 lines Merged revisions 141503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008)
+ | 4 lines Reverting behavior change (AGI should not exit non-zero
+ on SUCCESS) (closes issue #13434) Reported by: francesco_r
+ ........ ................
+
+2008-09-05 22:06 +0000 [r141368-141426] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 141367 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r141367 | mmichelson | 2008-09-05 16:12:09 -0500
+ (Fri, 05 Sep 2008) | 15 lines Merged revisions 141366 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep
+ 2008) | 7 lines Agent's should not try to call a channel's
+ indicate callback if the channel has been hung up. It will likely
+ crash otherwise ABE-1159 ........ ................
+
+2008-09-05 14:24 +0000 [r141116-141158] Steve Murphy <murf@digium.com>
+
+ * main/channel.c, /: Merged revisions 141157 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9
+ lines Merged revisions 141156 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1
+ line A small change to prevent double-posting of CDR's; thanks to
+ Daniel Ferrer for bringing it to our attention ........
+ ................
+
+ * pbx/ael/ael-test/ref.ael-vtest25 (added), /,
+ pbx/ael/ael-test/ael-vtest25/extensions.ael,
+ pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c,
+ pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged
+ revisions 141115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, 04 Sep 2008) |
+ 78 lines Merged revisions 141094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) |
+ 70 lines (closes issue #13357) Reported by: pj Tested by: murf
+ (closes issue #13416) Reported by: yarns Tested by: murf If you
+ find this message overly verbose, relax, it's probably not meant
+ for you. This message is meant for probably only two people in
+ the whole world: me, or the poor schnook that has to maintain
+ this code because I'm either dead or unavailable at the moment.
+ This fix solves two reports, both having to do with embedding a
+ function call in a ${} construct. It was tricky because the
+ funccall syntax has parenthesis () in it. And up till now, the
+ 'word' token in the flex stuff didn't allow that, because it
+ would tend to steal the LP and RP tokens. To be truthful, the
+ "word" token was the trickiest, most unstable thing in the whole
+ lexer. I was lucky it made this long without complaints. I had to
+ choose every character in the pattern with extreme care, and I
+ knew that someday I'd have to revisit it. Well, the day has come.
+ So, my brilliant idea (and I'm being modest), was to use the
+ surrounding ${} construct to make a state machine and capture
+ everything in it, no matter what it contains. But, I have to now
+ treat the word token like I did with comments, in that I turn the
+ whole thing into a state-machine sort of spec, with new contexts
+ "curlystate", "wordstate", and "brackstate". Wait a minute,
+ "brackstate"? Yes, well, it didn't take very many regression
+ tests to point out if I do this for ${} constructs, I also have
+ to do it with the $[] constructs, too. I had to create a separate
+ pcbstack2 and pcbstack3 because these constructs can occur inside
+ macro argument lists, and when we have two state machines
+ operating on the same structures we'd get problems otherwise. I
+ guess I could have stopped at pcbstack2 and had the brackstate
+ stuff share it, but it doesn't hurt to be safe. So, the pcbpush
+ and pcbpop routines also now have versions for "2" and "3". I had
+ to add the {KEYWORD} construct to the initial pattern for "word",
+ because previously word would match stuff like "default7",
+ because it was a longer match than the keyword "default". But,
+ not any more, because the word pattern only matches only one or
+ two characters now, and it will always lose. So, I made it the
+ winner again by making an optional match on any of the keywords
+ before it's normal pattern. I added another regression test to
+ make sure we don't lose this in future edits, and had to fix just
+ one regression, where it no longer reports a 'cascaded' error,
+ which I guess is a plus. I've given some thought as to whether to
+ apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I
+ decided to put it in 1.4 because one of the bug reports was
+ against 1.4; and it is unexpected that AEL cannot handle this
+ situation. It actually reduced the amount of useless "cascade"
+ error messages that appeared in the regressions (by one line,
+ ehhem). There is a possible side-effect in that it does now do
+ more careful checking of what's in those ${} constructs, as far
+ as matching parens, and brackets are concerned. Some users may
+ find a an insidious problem and correct it this way. This should
+ be exceedingly rare, I hope. ........ ................
+
+2008-09-04 18:35 +0000 [r141086] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c, res/res_agi.c: Merged revisions 141039 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r141039 | jpeeler | 2008-09-04 12:27:56 -0500
+ (Thu, 04 Sep 2008) | 15 lines Merged revisions 141028 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008)
+ | 7 lines (closes issue #11979) Fixes multiple parking problems:
+ Crash when executing a park on an extension dialed by AGI due to
+ not returning the proper return code. Crash when using a builtin
+ feature that was a subset of a enabled dynamic feature. Crash due
+ to always hanging up the peer despite the fact that the peer was
+ supposed to be parked. ........ ................
+
+2008-09-03 20:18 +0000 [r140976] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 140975 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r140975 |
+ mmichelson | 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) | 4
+ lines Fix some locking order issues in app_queue. This was
+ brought up by atis on IRC a while ago. ........
+
+2008-09-03 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.6.0-rc4 released.
+
+2008-09-03 14:17 +0000 [r140825-140827] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, /: Merged revisions 140749 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) |
+ 11 lines Merged revisions 140747 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1
+ line I am turning the warnings generated in ast_cdr_free and
+ post_cdr into verbose level 2 messages. Really, they matter
+ little to end users. You either get the CDR's you wanted, or you
+ don't, and it is a bug. For trunk, I am going one step further.
+ These messages were pretty worthless even for debug, so I'm
+ completely removing them. ........ ................
+
+ * main/channel.c, /: Merged revisions 140692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) |
+ 13 lines Merged revisions 140690 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1
+ line After reconsidering, with respect to 13409, ast_cdr_detach
+ should be OK, better in fact, than ast_cdr_free, which generates
+ lots of useless warnings that will undoubtably generate
+ complaints. Hmmm. It doesn't hush the useless warnings, but it
+ does allow control of posting via the detach and post routines,
+ for those possible situations, where you'd want to post
+ single-channel cdrs. ........ ................
+
+ * main/channel.c, main/pbx.c, /: Merged revisions 140691 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue,
+ 02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) |
+ 14 lines (closes issue #13409) Reported by: tomaso Patches:
+ asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license
+ 564) I basically spent the day, verifying that this patch solves
+ the problem, and doesn't hurt in non-problem cases. Why valgrind
+ did not plainly reveal this leak absolutely mystifies and stuns
+ me. Many, many thanks to tomaso for finding and providing the
+ fix. ........ ................
+
+2008-09-03 13:27 +0000 [r140818] Russell Bryant <russell@digium.com>
+
+ * main/poll.c, /: Merged revisions 140817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008)
+ | 12 lines Merged revisions 140816 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008)
+ | 4 lines Don't freak out if the poll emulation receives NULL for
+ the pollfds array (closes issue #13307) Reported by: jcovert
+ ........ ................
+
+2008-09-02 18:17 +0000 [r140607] Sean Bright <sean.bright@gmail.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 140606 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r140606 | seanbright | 2008-09-02 14:15:54 -0400
+ (Tue, 02 Sep 2008) | 16 lines Merged revisions 140605 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep
+ 2008) | 8 lines Make sure to use the correct length of the
+ mohinterpret and mohsuggest buffers when copying configuration
+ values. (closes issue #13336) Reported by:
+ decryptus_proformatique Patches:
+ chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
+ by decryptus (license 555) ........ ................
+
+2008-09-02 15:12 +0000 [r140564-140567] Russell Bryant <russell@digium.com>
+
+ * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions
+ 140566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 |
+ russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines
+ Update instructions for getting libresample ........
+
+2008-08-27 20:15 +0000 [r140302-140304] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Revert commit 140302. Should not be merging
+ changes like that into a release-candidate branch
+
+ * channels/chan_sip.c: Merged revisions 140301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug
+ 2008) | 19 lines Merged revisions 140299 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug
+ 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when
+ in pedantic mode. The problem was that the wrong tags would be
+ compared depending on the direction of the call. (closes issue
+ #13353) Reported by: flefoll Patches:
+ chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
+ (license 244) ........ ................
+
+2008-08-26 18:12 +0000 [r140170] Russell Bryant <russell@digium.com>
+
+ * Makefile, /: Merged revisions 140169 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r140169 |
+ russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines
+ Fix building menuselect-tree with PRINT_DIR set. We _must_ use
+ the --quiet flag here, or else some arbitrary text will end up in
+ the resulting menuselect-tree file and things will explode.
+ ........
+
+2008-08-25 21:33 +0000 [r139918] Sean Bright <sean.bright@gmail.com>
+
+ * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged
+ revisions 139915 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug
+ 2008) | 17 lines Merged revisions 139909 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug
+ 2008) | 9 lines Some versions of awk (nawk, for example) don't
+ like empty regular expressions so be slightly more verbose.
+ (closes issue #13374) Reported by: dougm Patches: 13374.diff
+ uploaded by seanbright (license 71) Tested by: dougm ........
+ ................
+
+2008-08-25 21:05 +0000 [r139872] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 139870 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008)
+ | 10 lines Merged revisions 139869 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008)
+ | 2 lines Make SIPADDHEADER() propagate indefinitely ........
+ ................
+
+2008-08-25 16:00 +0000 [r139774] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /, main/features.c: Merged revisions 139770 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon,
+ 25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9
+ lines This patch reverts the changes made via 139347, and 139635,
+ as users are seeing adverse difference. I will un-close 13251.
+ Back to the drawing board/ concept/ beginning/ whatever! ........
+ ................
+
+2008-08-24 16:30 +0000 [r139705-139708] Tilghman Lesher <tlesher@digium.com>
+
+ * /, cdr/cdr_pgsql.c: Merged revisions 139707 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r139707 |
+ tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines
+ Memory leak ........
+
+2008-08-22 22:35 +0000 [r139628-139671] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 139662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) |
+ 14 lines Merged revisions 139635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6
+ lines I found some problems with the code I committed earlier,
+ when I merged them into trunk, so I'm coming back to clean up.
+ And, in the process, I found an error in the code I added to
+ trunk and 1.6.x, that I'll fix using this patch also. ........
+ ................
+
+ * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions
+ 139627 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) |
+ 59 lines Merged revisions 139347 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) |
+ 47 lines (closes issue #13251) Reported by: sergee Tested by:
+ murf THis is a bold move for a static release fix, but I wouldn't
+ have made it if I didn't feel confident (at least a *bit*
+ confident) that it wouldn't mess everyone up. The reasoning goes
+ something like this: 1. We simply cannot do anything with CDR's
+ at the current point (in pbx.c, after the __ast_pbx_run loop).
+ It's way too late to have any affect on the CDRs. The CDR is
+ already posted and gone, and the remnants have been cleared. 2. I
+ was very much afraid that moving the running of the 'h' extension
+ down into the bridge code (where it would be now practical to do
+ it), would result in a lot more calls to the 'h' exten, so I
+ implemented it as another exten under another name, but found, to
+ my pleasant surprise, that there was a 1:1 correspondence to the
+ running of the 'h' exten in the pbx_run loop, and the new spot at
+ the end of the bridge. So, I ifdef'd out the current 'h' loop,
+ and moved it into the bridge code. The only difference I can see
+ is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this
+ is still an important decision point, I can replicate it if there
+ are complaints. To be perfectly honest, the KEEPALIVE situation
+ is not totally clear to me, and how it relates to a post-bridge
+ situation is less clear. I suspect the users will point out
+ everything in total clarity if this steps on anyone's toes! 3. I
+ temporarily swap the bridge_cdr into the channel before running
+ the 'h' exten, which makes it possible for users to edit the cdr
+ before it goes out the door. And, of course, with the
+ endbeforehexten config var set, the users can also get at the
+ billsec/duration vals. After the h exten finishes, the cdr is
+ swapped back and processing continues as normal. Please, all who
+ deal with CDR's, please test this version of Asterisk, and file
+ bug reports as appropriate! ........ I also made a little fix to
+ the app_dial's 'e' option, that is related to my updates.
+ ................
+
+2008-08-22 20:21 +0000 [r139458-139564] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/threadstorage.h, /: Merged revisions 139554 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500
+ (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug
+ 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is
+ selected (closes issue #13298) Reported by: snuffy Patches:
+ bug13298_20080822.diff uploaded by snuffy (license 35) ........
+ ................
+
+ * /, channels/chan_iax2.c: Merged revisions 139469 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r139469 | mmichelson | 2008-08-22 12:25:12 -0500
+ (Fri, 22 Aug 2008) | 11 lines Merged revisions 139466 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug
+ 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........
+ ................
+
+ * /, channels/chan_iax2.c: Merged revisions 139457 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r139457 | mmichelson | 2008-08-22 11:58:21 -0500
+ (Fri, 22 Aug 2008) | 15 lines Merged revisions 139456 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug
+ 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from
+ incorrect locking order between iax2_pvt and ast_channel
+ structures. AST-13 ........ ................
+
+2008-08-21 23:46 +0000 [r139400] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 139391 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r139391 | jpeeler | 2008-08-21 18:41:50 -0500
+ (Thu, 21 Aug 2008) | 11 lines Merged revisions 139387 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008)
+ | 3 lines Fixes loop that could possibly never exit in the event
+ of a channel never being able to be opened or specify after a
+ restart. (closes issue #11017) ........ ................
+
+2008-08-21 10:02 +0000 [r139282] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 139281 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008)
+ | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel!
+ (closes issue #13310) Reported by: eliel Patches:
+ chan_gtalk.c.patch uploaded by eliel (license 64) ........
+
+2008-08-20 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.6.0-rc3 released.
+
+2008-08-20 22:17 +0000 [r139216] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 139215 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008)
+ | 19 lines Merged revisions 139213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008)
+ | 11 lines Fix a crash in the ChanSpy application. The issue here
+ is that if you call ChanSpy and specify a spy group, and sit in
+ the application long enough looping through the channel list, you
+ will eventually run out of stack space and the application with
+ exit with a seg fault. The backtrace was always inside of a
+ harmless snprintf() call, so it was tricky to track down.
+ However, it turned out that the call to snprintf() was just the
+ biggest stack consumer in this code path, so it would always be
+ the first one to hit the boundary. (closes issue #13338) Reported
+ by: ruddy ........ ................
+
+2008-08-20 20:12 +0000 [r139155] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: Fix bug where the samples were not accurate
+ when in G723 mode, which would cause the timestamp field of the
+ RTP header to be invalid.
+
+2008-08-20 17:30 +0000 [r139104] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, /: Merged revisions 139083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) |
+ 20 lines Merged revisions 139074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) |
+ 12 lines (closes issue #13263) Reported by: brainy Tested by:
+ murf The specialized reset routine is tromping on the flags field
+ of the CDR. I made a change to not reset the DISABLED bit. This
+ should get rid of this problem. ........ ................
+
+2008-08-20 15:39 +0000 [r138889-139017] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 139016 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug
+ 2008) | 14 lines Merged revisions 139015 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug
+ 2008) | 6 lines sip_read should properly handle a NULL return
+ from sip_rtp_read. (closes issue #13257) Reported by: travishein
+ ........ ................
+
+ * apps/app_chanspy.c: Manually add revision 138887 from trunk to
+ the 1.6.0 branch. I had misunderstood the policy for when to
+ merge to 1.6.0 since it moved to rc status.
+
+2008-08-19 16:38 +0000 [r138846-138847] Steve Murphy <murf@digium.com>
+
+ * utils/conf2ael.c, /, res/ael/ael.tab.c, res/ael/ael.y,
+ res/ael/ael.tab.h, utils/ael_main.c: Merged revisions 138845 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug
+ 2008) | 1 line Oops. put a decl in a generated file. My bad, but
+ fixed now. ........
+
+ * main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y,
+ res/ael/ael.tab.h: Merged revisions 138815 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 |
+ murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines
+ These changes are in regards to bug 13249, where users are being
+ surprised by the changes made to the Set app in trunk/1.6.x, as
+ they come from the 1.4 world. They are only bitten if they write
+ their AEL dialplan in the 1.4 world, and then carry it over to a
+ trunk/1.6.x installation where a "make samples" was executed, or
+ where they hand-edited the asterisk.conf file and added the
+ [compat] category with app_set = 1.6 (or higher). (this commit
+ does not totally solve 13249, at least not yet) The change
+ involves issueing a single warning while the AEL file is loading,
+ if: 1. app_set is present in the config file, and set to 1.6 or
+ higher. 2. there are double quotes in an assignment statement (eg
+ x = "hi there";) 3. the warning was not already issued. The
+ standalone app, aelparse, does not (yet) issue this warning. I'd
+ have to have it read in the asterisk.conf file, and that's a bit
+ of hassle. I'll add it if users request it, tho. ........
+
+2008-08-19 00:15 +0000 [r138776-138781] Sean Bright <sean.bright@gmail.com>
+
+ * /, channels/chan_sip.c: Merged revisions 138778-138780 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon,
+ 18 Aug 2008) | 1 line While we're at it, make this machine
+ parseable too. ........ r138779 | seanbright | 2008-08-18
+ 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we
+ don't need anymore. ........ r138780 | seanbright | 2008-08-18
+ 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now,
+ too (woops) ........
+
+ * /, channels/chan_sip.c: Merged revisions 138775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 |
+ seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3
+ lines Change event header to RegistrationTime to be more
+ consistent (and avoid breaking existing frameworks). Pointed out
+ by Laureano on #asterisk-dev. ........
+
+2008-08-18 20:23 +0000 [r138688-138695] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 138687 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug
+ 2008) | 18 lines Merged revisions 138685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug
+ 2008) | 10 lines Change the inequalities used in app_queue with
+ regards to timeouts from being strict to non-strict for more
+ accuracy. (closes issue #13239) Reported by: atis Patches:
+ app_queue_timeouts_v2.patch uploaded by atis (license 242)
+ ........ ................
+
+2008-08-18 15:54 +0000 [r138632] Jason Parker <jparker@digium.com>
+
+ * Makefile, /: Merged revisions 138631 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r138631 |
+ qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line
+ Remove option that isn't valid here. ........
+
+2008-08-18 02:14 +0000 [r138519] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 138518 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008)
+ | 1 line add missing define for SS7 in dahdi_restart ........
+
+2008-08-17 14:14 +0000 [r138443-138483] Sean Bright <sean.bright@gmail.com>
+
+ * /, main/features.c: Merged revisions 138482 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 |
+ seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6
+ lines Move Uniqueid to the end of the event for those that rely
+ on the position of the name/value pairs, pointed out by
+ snuffy-home on #asterisk-commits. For those of you who rely on
+ the position of name/value pairs in manager events... stop...
+ that is why associative arrays were invented. ........
+
+ * /, main/features.c: Merged revisions 138479 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 |
+ seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7
+ lines Add Uniqueid header to ParkedCall manager event. (closes
+ issue #13323) Reported by: srt Patches:
+ 13323_unique_id_for_parkedcalls_event.diff uploaded by srt
+ (license 378) ........
+
+ * main/rtp.c, /: Merged revisions 138476 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 |
+ seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7
+ lines Add missing colons to RTCPReceived and RTCPSent manager
+ events. (closes issue #13319) Reported by: srt Patches:
+ 13319_rtcp_manager_event_headers.diff uploaded by srt (license
+ 378) ........
+
+ * /, channels/chan_iax2.c: Merged revisions 138473 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug
+ 2008) | 7 lines Fix the output of the JitterBufStats manager
+ event. (closes issue #13324) Reported by: srt Patches:
+ 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt
+ (license 378) ........
+
+ * configs/cdr_tds.conf.sample, /: Merged revisions 138442 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat,
+ 16 Aug 2008) | 4 lines Since it's introduction in revision 3497,
+ cdr_tds has *never* read the port configuration option from
+ cdr_tds.conf. So go ahead and remove it from the sample config.
+ ........
+
+2008-08-16 13:07 +0000 [r138410-138413] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 138412 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008)
+ | 2 lines Fix compilation warnings (found with dev-mode) ........
+
+2008-08-16 01:14 +0000 [r138333-138362] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 138361 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r138361 | jpeeler | 2008-08-15 20:13:26 -0500
+ (Fri, 15 Aug 2008) | 9 lines Merged revisions 138360 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15
+ Aug 2008) | 1 line fixes use count to properly decrement if an
+ active dahdi channel is destroyed allowing module to be unloaded
+ ........ ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 138311 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r138311 | jpeeler | 2008-08-15 18:46:09 -0500
+ (Fri, 15 Aug 2008) | 20 lines Merged revisions
+ 138119,138151,138238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008)
+ | 4 lines Fixes the dahdi restart functionality. Dahdi restart
+ allows one to restart all DAHDI channels, even if they are
+ currently in use. This is different from unloading and then
+ loading the module since unloading requires the use count to be
+ zero. Reloading the module is different in that the signalling is
+ not changed from what it was originally configured. Also, this
+ fixes not closing all the file descriptors for D-channels upon
+ module unload (which would prevent loading the module
+ afterwards). (closes issue #11017) ........ r138151 | jpeeler |
+ 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared
+ static mutexes using AST_MUTEX_DEFINE_STATIC macro ........
+ r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008)
+ | 1 line initialize condition variable ss_thread_complete using
+ ast_cond_init ........ ................
+
+2008-08-15 23:03 +0000 [r138207-138262] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 138260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008)
+ | 16 lines Merged revisions 138258 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008)
+ | 8 lines More fixes for realtime peers. (closes issue #12921)
+ Reported by: Nuitari Patches: 20080804__bug12921.diff.txt
+ uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: Corydon76 ........
+ ................
+
+ * configs/extensions.conf.sample, main/pbx.c, /: Merged revisions
+ 138206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 |
+ tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines
+ Remove deprecated syntax from sample config file (closes issue
+ #13314) Reported by: kue ........
+
+2008-08-15 20:20 +0000 [r138156-138157] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to
+ dfd to match 1.4 (left over from DAHDI transition)
+
+2008-08-15 15:12 +0000 [r138029] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 138028 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008)
+ | 17 lines Merged revisions 138027 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008)
+ | 9 lines Ensure that when a hangup occurs in autoservice, that a
+ hangup frame gets properly deferred to be read from the channel
+ owner when it gets taken out of autoservice. (closes issue
+ #12874) Reported by: dimas Patches: v1-12874.patch uploaded by
+ dimas (license 88) ........ ................
+
+2008-08-15 15:04 +0000 [r138025] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 138024 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r138024 | tilghman | 2008-08-15 10:03:32 -0500
+ (Fri, 15 Aug 2008) | 16 lines Merged revisions 138023 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008)
+ | 8 lines Additional check for more string specifiers than
+ arguments. (closes issue #13299) Reported by: adomjan Patches:
+ 20080813__bug13299.diff.txt uploaded by Corydon76 (license 14)
+ func_strings.c-sprintf.patch uploaded by adomjan (license 487)
+ Tested by: adomjan ........ ................
+
+2008-08-14 22:43 +0000 [r137988] Russell Bryant <russell@digium.com>
+
+ * /, doc/tex/Makefile: Merged revisions 137987 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r137987 |
+ russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines
+ Fix a bashism that causes an error when trying to build the pdf
+ on ubuntu ........
+
+2008-08-14 18:48 +0000 [r137934] Sean Bright <sean.bright@gmail.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 137933 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug
+ 2008) | 8 lines Fix memory leak in cdr_sqlite3_custom. (closes
+ issue #13304) Reported by: eliel Patches: sqlite.patch uploaded
+ by eliel (license 64) (Slightly modified by me) ........
+
+2008-08-14 17:01 +0000 [r137849-137852] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 137848 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r137848 | tilghman | 2008-08-14 11:52:43 -0500
+ (Thu, 14 Aug 2008) | 17 lines Merged revisions 137847 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008)
+ | 9 lines When creating the secondary subchannel name, it is
+ necessary to compare to the existing channel name without the
+ "Zap/" or "DAHDI/" prefix, since our test string is also without
+ that prefix. (closes issue #13027) Reported by: dferrer Patches:
+ chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525)
+ (Slightly modified by me, to compensate for both names) ........
+ ................
+
+2008-08-14 Jason Parker <jparker@digium.com>
+
+ * Asterisk 1.6.0-rc2 released.
+
+2008-08-14 15:37 +0000 [r137814] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 137812 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 |
+ qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines
+ Make sure we set the socket port, so we don't try to use <ip
+ address>:0. (closes issue #13255) Reported by: falves11 Patches:
+ 13255-socketport.diff uploaded by qwell (license 4) Tested by:
+ falves11 ........
+
+2008-08-14 15:20 +0000 [r137783] Russell Bryant <russell@digium.com>
+
+ * /, configs/sip.conf.sample: Merged revisions 137732 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r137732 | russell | 2008-08-14 09:15:50 -0500
+ (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008)
+ | 4 lines Comments in this config file were aligned only if your
+ tab size was set to 8. So, convert tabs to spaces so that things
+ should be aligned regardless of what tab size you use in your
+ editor. ........ ................
+
+2008-08-14 15:05 +0000 [r137781] Sean Bright <sean.bright@gmail.com>
+
+ * cdr/cdr_tds.c, /: Merged revisions 137780 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r137780 |
+ seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8
+ lines If we detect that we are no longer connected, try to
+ reconnect a few times before giving up. This relies on the
+ timeout settings in the freetds.conf file and, unfortunately, on
+ a recent version of FreeTDS (0.82 or newer). I either need to
+ change the current execs to be non-blocking (which I do not want
+ to do) or we have to force people to run with the latest and
+ greatest of FreeTDS. I'm on the fence... ........
+
+2008-08-14 02:04 +0000 [r137681] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, Zaptel-to-DAHDI.txt: Merged revisions 137680 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug
+ 2008) | 9 lines Merged revisions 137679 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug
+ 2008) | 1 line forgot one module name that changed ........
+ ................
+
+2008-08-13 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.6.0-rc1 released.
+
+2008-08-13 23:00 +0000 [r137631-137641] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, build_tools/prep_tarball: Merged revisions 137640 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug
+ 2008) | 1 line make this script actually work ........
+
+ * /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions
+ 137627 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug
+ 2008) | 9 lines Merged revisions 137530 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug
+ 2008) | 1 line add document describing what users will need to be
+ aware of when upgrading to this version and using DAHDI ........
+ ................
+
+2008-08-13 21:09 +0000 [r137497-137533] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 137532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 |
+ qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines
+ Correctly end locally ended calls. (closes issue #12170) Reported
+ by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff
+ uploaded by bbryant (license 36) Tested by: bbryant, pabelanger
+ ........
+
+ * /, apps/app_fax.c: Merged revisions 137496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 |
+ qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines
+ Add FAXMODE variable with what fax transport was used. (closes
+ issue #13252) Patches: v1-13252.patch uploaded by dimas (license
+ 88) ........
+
+2008-08-13 14:47 +0000 [r137350-137407] Sean Bright <sean.bright@gmail.com>
+
+ * /, doc/tex/cdrdriver.tex: Merged revisions 137406 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r137406 | seanbright | 2008-08-13 10:41:49 -0400
+ (Wed, 13 Aug 2008) | 9 lines Merged revisions 137405 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed,
+ 13 Aug 2008) | 1 line Update docs to reflect the change to
+ cdr_tds ........ ................
+
+ * cdr/cdr_tds.c, /: Merged revisions 137403 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r137403 |
+ seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1
+ line Use the ast_vasprintf macro instead of vasprintf directly.
+ ........
+
+2008-08-12 19:48 +0000 [r137300-137302] Russell Bryant <russell@digium.com>
+
+ * doc/tex/asterisk.tex, /: Merged revisions 137301 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008)
+ | 2 lines Grammar hax from Qwell ........
+
+ * doc/tex/asterisk.tex, /: Merged revisions 137299 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008)
+ | 3 lines Note that developer documentation belongs in doxygen,
+ and not integrated with the user manual stuff in doc/tex/.
+ ........
+
+2008-08-11 16:15 +0000 [r137240] Russell Bryant <russell@digium.com>
+
+ * Makefile, /: Merged revisions 137239 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r137239 |
+ russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines
+ Make PRINT_DIR work as advertised. ........
+
+2008-08-11 14:31 +0000 [r137217] Sean Bright <sean.bright@gmail.com>
+
+ * cdr/cdr_tds.c, /, UPGRADE.txt: Merged revisions 137203 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon,
+ 11 Aug 2008) | 7 lines Log the userfield CDR variable like the
+ other CDR backends, assuming the column is actually there. If
+ it's not, we still log everything else as before. (closes issue
+ #13281) Reported by: falves11 ........
+
+2008-08-11 00:27 +0000 [r137160] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c, /: Merged revisions 137150 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008)
+ | 13 lines Merged revisions 137138 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008)
+ | 5 lines Deallocate database connection handle on disconnect, as
+ we allocate another one on connect. (closes issue #13271)
+ Reported by: dveiga ........ ................
+
+2008-08-09 15:27 +0000 [r136948] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged
+ revisions 136947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008)
+ | 18 lines Merged revisions 136946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500
+ (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008)
+ | 2 lines Regression fixes for Solaris ........ ................
+ ................
+
+2008-08-09 01:16 +0000 [r136860] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 136859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r136859 |
+ tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines
+ Update documentation as to the behavior of AGI in 1.6.0 and
+ higher. Also, add an OOB message that answers the question of, if
+ AGI no longer shuts down the connection on hangup, how will
+ FastAGI know when to stop processing the call? ........
+
+2008-08-08 15:33 +0000 [r136785] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 136784 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug
+ 2008) | 3 lines Fix compilation for ODBC voicemail ........
+
+2008-08-08 06:45 +0000 [r136778] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
+ pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, /,
+ pbx/ael/ael-test/ref.ael-ntest10, include/asterisk/ael_structs.h,
+ utils/ael_main.c: Merged revisions 136746 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) |
+ 40 lines Merged revisions 136726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) |
+ 32 lines (closes issue #13236) Reported by: korihor Wow, this one
+ was a challenge! I regrouped and ran a new strategy for setting
+ the ~~MACRO~~ value; I set it once per extension, up near the
+ top. It is only set if there is a switch in the extension. So, I
+ had to put in a chunk of code to detect a switch in the pval
+ tree. I moved the code to insert the set of ~~exten~~ up to the
+ beginning of the gen_prios routine, instead of down in the switch
+ code. I learned that I have to push the detection of the switches
+ down into the code, so everywhere I create a new exten in
+ gen_prios, I make sure to pass onto it the values of the
+ mother_exten first, and the exten next. I had to add a couple
+ fields to the exten struct to accomplish this, in the
+ ael_structs.h file. The checked field makes it so we don't repeat
+ the switch search if it's been done. I also updated the
+ regressions. ........ ................
+
+2008-08-08 02:36 +0000 [r136753] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Merged revisions 136751 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r136751 |
+ tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines
+ Removing bad properties ........
+
+2008-08-07 23:42 +0000 [r136719-136724] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: This is weird. Either SVN or vim tabbed a
+ bunch of functions over one level during a merge.
+
+ * apps/app_voicemail.c, /: Merged revisions 136722 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug
+ 2008) | 3 lines Remove one last batch of debug messages ........
+
+ * apps/app_voicemail.c, /: Merged revisions 136715 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug
+ 2008) | 18 lines Merging the imap_consistency_trunk branch to
+ trunk. For an explanation of what "imap_consistency" is, please
+ see svn revision 134223 to the 1.4 branch. Coincidentally, this
+ also fixes a recent bug report regarding the inability to save
+ messages to the new folder when using IMAP storage since they
+ will would be flagged as "seen" and not be recognized as new
+ messages. (closes issue #13234) Reported by: jaroth ........
+
+2008-08-07 20:41 +0000 [r136672-136674] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: Removing code that was commented out.
+
+ * codecs/codec_dahdi.c: Updated codec_dahdi to use the transcoder
+ interface in the DAHDI. (Issue: DAHDI-42)
+
+2008-08-07 20:26 +0000 [r136632-136663] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/features.c: Merged revisions 136660 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 |
+ mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4
+ lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears
+ once for every bridged call ........
+
+ * main/pbx.c, /: Merged revisions 136635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 |
+ mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5
+ lines Don't allow Answer() to accept a negative argument.
+ Negative argument means an infinite delay and we don't want that.
+ ........
+
+ * main/channel.c, /: Merged revisions 136633 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 |
+ mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7
+ lines Fix a calculation error I had made in the poll. The poll
+ would reset to 500 ms every time a non-voice frame was received.
+ The total time we poll should be 500 ms, so now we save the
+ amount of time left after the poll returned and use that as our
+ argument for the next call to poll ........
+
+ * main/channel.c, /: Merged revisions 136631 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 |
+ mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13
+ lines Scrap the 500 ms delay when Asterisk auto-answers a
+ channel. Instead, poll the channel until receiving a voice frame.
+ The cap on this poll is 500 ms. The optional delay is still
+ allowable in the Answer() application, but the delay has been
+ moved back to its original position, after the call to the
+ channel's answer callback. The poll for the voice frame will not
+ happen if a delay is specified when calling Answer(). (closes
+ issue #12708) Reported by: kactus ........
+
+2008-08-07 19:19 +0000 [r136598] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c, /,
+ configs/misdn.conf.sample: Merged revisions 136594 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r136594 | rmudgett | 2008-08-07 14:01:03 -0500
+ (Thu, 07 Aug 2008) | 13 lines Merged revisions 136241 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008)
+ | 5 lines * The allowed_bearers setting in misdn.conf misspelled
+ one of its options: digital_restricted. * Fixed some other
+ spelling errors and typos. ........ ................
+
+2008-08-07 17:44 +0000 [r136506-136543] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/doxyref.h, /: Merged revisions 136542 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500
+ (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ ........ ................
+
+2008-08-07 16:57 +0000 [r136490] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 136489 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008)
+ | 15 lines Merged revisions 136488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008)
+ | 7 lines Update persistent state on all exit conditions. (closes
+ issue #12916) Reported by: sgenyuk Patches: app_queue.patch.txt
+ uploaded by neutrino88 (license 297) Tested by: sgenyuk, aragon
+ ........ ................
+
+2008-08-06 20:16 +0000 [r136113-136192] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136191 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r136191 | tilghman | 2008-08-06 15:15:34 -0500
+ (Wed, 06 Aug 2008) | 12 lines Merged revisions 136190 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008)
+ | 4 lines -C option takes a filename, not a directory path.
+ (closes issue #13007) Reported by: klaus3000 ........
+ ................
+
+ * /, funcs/func_dialgroup.c: Merged revisions 136112 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008)
+ | 7 lines Persist DIALGROUP() values in astdb (closes issue
+ #13138) Reported by: Corydon76 Patches:
+ 20080725__bug13138.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: pj ........
+
+2008-08-06 16:00 +0000 [r136064] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /, channels/chan_skinny.c: Merged revisions 136063
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500
+ (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug
+ 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame
+ type, there are places where ast_rtp_new_source may be called
+ where the tech_pvt of a channel may not yet have an rtp structure
+ allocated. This caused a crash in chan_skinny, which was fixed
+ earlier, but now the same crash has been reported against
+ chan_h323 as well. It seems that the best solution is to modify
+ ast_rtp_new_source to not attempt to set the marker bit if the
+ rtp structure passed in is NULL. This change to
+ ast_rtp_new_source also allows the removal of what is now a
+ redundant pointer check from chan_skinny. (closes issue #13247)
+ Reported by: pj ........ ................
+
+2008-08-06 13:59 +0000 [r136006] Olle Johansson <oej@edvina.net>
+
+ * /, res/res_jabber.c: Merged revisions 136005 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r136005 |
+ oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines -
+ Formatting - Changing debug messages from VERBOSE to DEBUG
+ channel - Adding a few todo's - Adding a few more "XMPP"'s to
+ compliment Jabber... ........
+
+2008-08-06 03:56 +0000 [r135901-135951] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 135950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008)
+ | 12 lines Merged revisions 135949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008)
+ | 4 lines Fix a longstanding bug in channel walking logic, and
+ fix the explanation to make sense. (Closes issue #13124) ........
+ ................
+
+ * /, main/translate.c: Merged revisions 135938 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008)
+ | 12 lines Merged revisions 135915 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008)
+ | 4 lines Since powerof() can return an error condition, it's
+ foolhardy not to detect and deal with that condition. (Related to
+ issue #13240) ........ ................
+
+ * include/asterisk/threadstorage.h, include/asterisk/utils.h, /:
+ Merged revisions 135900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008)
+ | 12 lines Merged revisions 135899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008)
+ | 4 lines 1) Bugfix for debugging code 2) Reduce compiler
+ warnings for another section of debugging code (Closes issue
+ #13237) ........ ................
+
+2008-08-06 00:31 +0000 [r135852] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/abstract_jb.h, main/channel.c, /,
+ main/abstract_jb.c, main/fixedjitterbuf.h: Merged revisions
+ 135851 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug
+ 2008) | 48 lines Merged revisions 135841,135847,135850 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug
+ 2008) | 27 lines Merging the issue11259 branch. The purpose of
+ this branch was to take into account "burps" which could cause
+ jitterbuffers to misbehave. One such example is if the L option
+ to Dial() were used to inject audio into a bridged conversation
+ at regular intervals. Since the audio here was not passed through
+ the jitterbuffer, it would cause a gap in the jitterbuffer's
+ timestamps which would cause a frames to be dropped for a brief
+ period. Now ast_generic_bridge will empty and reset the
+ jitterbuffer each time it is called. This causes injected audio
+ to be handled properly. ast_generic_bridge also will empty and
+ reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE
+ frame since the change in audio source could negatively affect
+ the jitterbuffer. All of this was made possible by adding a new
+ public API call to the abstract_jb called ast_jb_empty_and_reset.
+ (closes issue #11259) Reported by: plack Tested by: putnopvut
+ ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue,
+ 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel
+ that occurred when I was testing for a memory leak ........
+ r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug
+ 2008) | 3 lines Remove properties that should not be here
+ ........ ................
+
+2008-08-05 23:52 +0000 [r135822] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c,
+ include/asterisk/cdr.h: Merged revisions 135821 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) |
+ 42 lines Merged revisions 135799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) |
+ 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf
+ I discovered that also, in the previous bug fixes and changes,
+ the cdr.conf 'unanswered' option is not being obeyed, so I fixed
+ this. And, yes, there are two 'answer' times involved in this
+ scenario, and I would agree with you, that the first answer time
+ is the time that should appear in the CDR. (the second 'answer'
+ time is the time that the bridge was begun). I made the necessary
+ adjustments, recording the first answer time into the peer cdr,
+ and then using that to override the bridge cdr's value. To get
+ the 'unanswered' CDRs to appear, I purposely output them, using
+ the dial cmd to mark them as DIALED (with a new flag), and
+ outputting them if they bear that flag, and you are in the right
+ mode. I also corrected one small mention of the Zap device to
+ equally consider the dahdi device. I heavily tested 10-sec-wait
+ macros in dial, and without the macro call; I tested hangups
+ while the macro was running vs. letting the macro complete and
+ the bridge form. Looks OK. Removed all the instrumentation and
+ debug. ........ ................
+
+2008-08-05 21:38 +0000 [r135749] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 135748 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r135748 | tilghman | 2008-08-05 16:37:35 -0500
+ (Tue, 05 Aug 2008) | 17 lines Merged revisions 135747 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008)
+ | 9 lines In a conversion to use ast_strlen_zero, the meaning of
+ the flag IAX_HASCALLERID was perverted. This change reverts IAX2
+ to the original meaning, which was, that the callerid set on the
+ client should be overridden on the server, even if that means the
+ resulting callerid is blank. In other words, if you set
+ "callerid=" in the IAX config, then the callerid should be
+ overridden to blank, even if set on the client. Note that there's
+ a distinction, even on realtime, between the field not existing
+ (NULL in databases) and the field existing, but set to blank
+ (override callerid to blank). ........ ................
+
+2008-08-05 13:27 +0000 [r135599] Sean Bright <sean.bright@gmail.com>
+
+ * main/cli.c, /: Merged revisions 135598 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug
+ 2008) | 9 lines Merged revisions 135597 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug
+ 2008) | 1 line Use PATH_MAX for filenames ........
+ ................
+
+2008-08-04 20:15 +0000 [r135538] Russell Bryant <russell@digium.com>
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 135537 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r135537 | russell | 2008-08-04 15:15:27 -0500
+ (Mon, 04 Aug 2008) | 10 lines Merged revisions 135536 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008)
+ | 2 lines fix a config sample typo ........ ................
+
+2008-08-04 17:12 +0000 [r135478-135486] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.mandriva.asterisk (added), Makefile,
+ contrib/init.d/rc.mandrake.asterisk (removed), /,
+ contrib/init.d/rc.mandriva.zaptel (added),
+ contrib/init.d/rc.mandrake.zaptel (removed): Merged revisions
+ 135485 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r135485 |
+ tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines
+ Rename Mandrake scripts to Mandriva (Closes issue #13221)
+ ........
+
+ * contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135483
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r135483 | tilghman | 2008-08-04 12:08:42 -0500
+ (Mon, 04 Aug 2008) | 11 lines Merged revisions 135482 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008)
+ | 2 lines Define ASTSBINDIR for script (Closes issue #13221)
+ ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 135480 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r135480 | tilghman | 2008-08-04 11:58:29 -0500
+ (Mon, 04 Aug 2008) | 14 lines Merged revisions 135479 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008)
+ | 6 lines Memory leak on unload (closes issue #13231) Reported
+ by: eliel Patches: app_voicemail.leak.patch uploaded by eliel
+ (license 64) ........ ................
+
+2008-08-04 16:28 +0000 [r135440-135475] Russell Bryant <russell@digium.com>
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 135474 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r135474 | russell | 2008-08-04 11:28:07 -0500
+ (Mon, 04 Aug 2008) | 10 lines Merged revisions 135473 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008)
+ | 2 lines Add a minor clarification to the documentation of
+ mohinterpret and mohsuggest ........ ................
+
+ * /, channels/chan_console.c: Merged revisions 135439 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008)
+ | 4 lines Be explicit that we don't want a result from this
+ callback. The callback would never indicate a match, so nothing
+ would have been returned anyway, but it was still a poor example
+ of proper usage. ........
+
+2008-08-02 05:15 +0000 [r135266] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, /: Merged revisions 135265 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 |
+ murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines
+ (closes issue #13202) Reported by: falves11 Tested by: murf
+ falves11 == The changes I introduce here seem to clear up the
+ problem for me. However, if they do not for you, please reopen
+ this bug, and we'll keep digging. The root of this problem seems
+ to be a subtle memory corruption introduced when creating an
+ extension with an empty extension name. While valgrind cannot
+ detect it outside of DEBUG_MALLOC mode, when compiled with
+ DEBUG_MALLOC, this is certain death. The code in main/features.c
+ is a puzzle to me. On the initial module load, the code is
+ attempting to add the parking extension before the features.conf
+ file has even been opened! I just wrapped the offending call with
+ an if() that will not try to add the extension if the extension
+ name is empty. THis seems to solve the corruption, and let the
+ "memory show allocations" work as one would expect. But, really,
+ adding an extension with an empty name is a seriously bad thing
+ to allow, as it will mess up all the pattern matching algorithms,
+ etc. So, I added a statement to the add_extension2 code to return
+ a -1 if this is attempted. in 1.6.0, the changes to only
+ main/pbx.c were applicable, as apparently the code added to
+ main/features by jpeeler were not included in 1.6.0. ........
+
+2008-08-01 19:30 +0000 [r135198] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_mgcp.c, /: Merged revisions 135197 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug
+ 2008) | 6 lines Remove some code that used to do something but
+ does not anymore, mainly to get rid of a shadow warning (but this
+ seemed legitimate enough to fix here instead of in my branch).
+ Thanks to putnopvut for taking a look as well. ........
+
+2008-08-01 17:10 +0000 [r135127-135129] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 135128 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 |
+ tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines
+ Picky, picky, buildbot ........
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 135126 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 |
+ tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines
+ SIP should use the transport type set in the Moved Temporarily
+ for the next invite. (closes issue #11843) Reported by:
+ pestermann Patches:
+ 20080723__issue11843_302_ignores_transport_16branch.diff uploaded
+ by bbryant (license 36)
+ 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by
+ bbryant (license 36) Tested by: pabelanger ........
+
+2008-08-01 14:43 +0000 [r135070] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
+ revisions 135067-135068 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r135067 |
+ mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13
+ lines IMAP storage functioned under the assumption that folders
+ such as "Work" and "Family" would be subfolders of the INBOX.
+ This is an invalid assumption to make, but it could be desirable
+ to set up folders in this manner, so a new option for
+ voicemail.conf, "imapparentfolder" has been added to allow for
+ this. (closes issue #13142) Reported by: jaroth Patches:
+ parentfolder.patch uploaded by jaroth (license 50) ........
+ r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug
+ 2008) | 3 lines IMAP-specific items must go in IMAP_STORAGE
+ defines... ........
+
+2008-08-01 12:18 +0000 [r135057-135062] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, apps/app_ices.c: Merged revisions 135059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008)
+ | 10 lines Merged revisions 135058 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008)
+ | 2 lines make app_ices compile on OpenBSD. ........
+ ................
+
+ * /, channels/chan_skinny.c: Merged revisions 135056 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r135056 | mvanbaak | 2008-08-01 13:00:13 +0200
+ (Fri, 01 Aug 2008) | 16 lines Merged revisions 135055 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 Aug 2008)
+ | 8 lines fix some potential deadlocks in chan_skinny (closes
+ issue #13215) Reported by: qwell Patches:
+ 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
+ Tested by: mvanbaak ........ ................
+
+2008-07-31 22:34 +0000 [r135034] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/http.c: Merged revisions 135016 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul
+ 2008) | 11 lines Merged revisions 134983 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul
+ 2008) | 3 lines accomodate users who seem to lack a sense of
+ humor :-) ........ ................
+
+2008-07-31 21:58 +0000 [r134926-134981] Tilghman Lesher <tlesher@digium.com>
+
+ * sample.call, main/manager.c, pbx/pbx_spool.c, /: Merged revisions
+ 134980 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008)
+ | 16 lines Blocked revisions 134976 via svnmerge ........ r134976
+ | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9
+ lines Specify codecs in callfiles and manager, to allow video
+ calls to be set up from callfiles and AMI. (closes issue #9531)
+ Reported by: Geisj Patches: 20080715__bug9531__1.4.diff.txt
+ uploaded by Corydon76 (license 14)
+ 20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license
+ 14) Tested by: Corydon76 ........ ................
+
+ * res/res_config_sqlite.c, /: Merged revisions 134977 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008)
+ | 2 lines Switch command order, to meet with current specs
+ ........
+
+2008-07-31 19:54 +0000 [r134923] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c: Merged revisions 134922 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) |
+ 63 lines Merged revisions 134883 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) |
+ 51 lines (closes issue #11849) Reported by: greyvoip Tested by:
+ murf OK, a few days of debugging, a bunch of instrumentation in
+ chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook
+ pages of notes later, I have made the small tweek necc. to get
+ the start time right on the second CDR when: A Calls B B answ. A
+ hits Xfer button on sip phone, A dials C and hits the OK button,
+ A hangs up C answers ringing phone B and C converse B and/or C
+ hangs up But does not harm the scenario where: A Calls B B answ.
+ B hits xfer button on sip phone, B dials C and hits the OK
+ button, B hangs up C answers ringing phone A and C converse A
+ and/or C hangs up The difference in start times on the second CDR
+ is because of a Masquerade on the B channel when the xfer number
+ is sent. It ends up replacing the CDR on the B channel with a
+ duplicate, which ends up getting tossed out. We keep a pointer to
+ the first CDR, and update *that* after the bridge closes. But,
+ only if the CDR has changed. I hope this change is specific
+ enough not to muck up any current CDR-based apps. In my defence,
+ I assert that the previous information was wrong, and this change
+ fixes it, and possibly other similar scenarios. I wonder if I
+ should be doing the same thing for the channel, as I did for the
+ peer, but I can't think of a scenario this might affect. I leave
+ it, then, as an exersize for the users, to find the scenario
+ where the chan's CDR changes and loses the proper start time.
+ ........ ................
+
+2008-07-31 19:41 +0000 [r134918] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_ices.c: Merged revisions 134917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008)
+ | 17 lines Merged revisions 134915 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008)
+ | 9 lines Get app_ices working again (closes issue #12981)
+ Reported by: dlogan Patches:
+ 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant
+ (license 36) 20080709__app_ices_v2_update_14.diff uploaded by
+ bbryant (license 36) Tested by: bbryant ........ ................
+
+2008-07-31 16:53 +0000 [r134816] Russell Bryant <russell@digium.com>
+
+ * channels/iax2-parser.c: Merged revisions 134815 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008)
+ | 15 lines Merged revisions 134814 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008)
+ | 7 lines In case we have some processing threads that free more
+ frames than they allocate, do not let the frame cache grow
+ forever. (closes issue #13160) Reported by: tavius Tested by:
+ tavius, russell ........ ................
+
+2008-07-31 16:07 +0000 [r134760] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 134759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul
+ 2008) | 24 lines Merged revisions 134758 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul
+ 2008) | 16 lines Add more timeout checks into app_queue,
+ specifically targeting areas where an unknown and potentially
+ long time has just elapsed. Also added a check to try_calling()
+ to return early if the timeout has elapsed instead of potentially
+ setting a negative timeout for the call (thus making it have *no*
+ timeout at all). (closes issue #13186) Reported by:
+ miquel_cabrespina Patches: 13186.diff uploaded by putnopvut
+ (license 60) Tested by: miquel_cabrespina ........
+ ................
+
+2008-07-30 22:41 +0000 [r134651-134707] Tilghman Lesher <tlesher@digium.com>
+
+ * main/sched.c, /, include/asterisk/sched.h: Merged revisions
+ 134703 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 |
+ tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines
+ Oops, wrong define ........
+
+ * /, configure, configure.ac: Merged revisions 134650 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r134650 | tilghman | 2008-07-30 16:40:08 -0500
+ (Wed, 30 Jul 2008) | 12 lines Merged revisions 134649 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008)
+ | 4 lines Qwell pointed out, via IRC, that the previous fix only
+ worked when explicitly set. When nothing is set, and the option
+ is implied, it breaks, because configure sets the prefix to
+ 'NONE'. Fixing. ........ ................
+
+2008-07-30 21:06 +0000 [r134599] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 134598 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul
+ 2008) | 15 lines Merged revisions 134556 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 |
+ mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7
+ lines Fix the parsing of the "reason" parameter in the Diversion:
+ header. (closes issue #13195) Reported by: woodsfsg ........
+ ................
+
+2008-07-30 20:39 +0000 [r134597] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 134596 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008)
+ | 14 lines Merged revisions 134595 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008)
+ | 6 lines Reduce stack consumption by 12.5% of the max stack size
+ to fix a crash when compiled with LOW_MEMORY. (closes issue
+ #13154) Reported by: edantie ........ ................
+
+2008-07-30 20:25 +0000 [r134561] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 134556 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 |
+ mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7
+ lines Fix the parsing of the "reason" parameter in the Diversion:
+ header. (closes issue #13195) Reported by: woodsfsg ........
+
+2008-07-30 19:56 +0000 [r134542] Russell Bryant <russell@digium.com>
+
+ * funcs/func_curl.c, /: Merged revisions 134541 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008)
+ | 12 lines Merged revisions 134540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008)
+ | 4 lines Fix a memory leak in func_curl. Every thread that used
+ this function leaked an allocation the size of a pointer.
+ (reported by jmls in #asterisk-dev) ........ ................
+
+2008-07-30 19:49 +0000 [r134482-134539] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, configure.ac: Merged revisions 134538 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r134538 | tilghman | 2008-07-30 14:48:37 -0500
+ (Wed, 30 Jul 2008) | 12 lines Merged revisions 134536 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 Jul 2008)
+ | 4 lines Only override sysconfdir and mandir when prefix=/usr
+ (closes issue #13093) Reported by: pabelanger ........
+ ................
+
+ * /, apps/app_queue.c: Merged revisions 134483 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r134483 |
+ tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines
+ Let "roundrobin" also reference rrmemory, for the 1.6 release (as
+ described in UPGRADE-1.4.txt) (Closes issue #13181) ........
+
+ * /, res/res_agi.c: Merged revisions 134481 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008)
+ | 13 lines Merged revisions 134480 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008)
+ | 5 lines launch_netscript sometimes returns -1, which fails to
+ set AGISTATUS. Map failure to -1, so that AGISTATUS is always
+ set. (closes issue #13199) Reported by: smw1218 ........
+ ................
+
+2008-07-30 18:33 +0000 [r134477] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/app.c: Merged revisions 134476 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul
+ 2008) | 12 lines Merged revisions 134475 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul
+ 2008) | 4 lines Fix a spot where a function could return without
+ bringing a channel out of autoservice. ........ ................
+
+2008-07-30 15:34 +0000 [r134356] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 134355 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul
+ 2008) | 10 lines Merged revisions 134352 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul
+ 2008) | 2 lines use the proper method for building version.h
+ ........ ................
+
+2008-07-29 22:29 +0000 [r134283] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_rpt.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, /,
+ apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c:
+ Merged revisions 134260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r134260 |
+ kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2
+ lines build against the now-typedef-free dahdi/user.h, and remove
+ some #ifdefs for features that will always be present in DAHDI
+ ........
+
+2008-07-28 22:16 +0000 [r134164] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 134163 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r134163 | tilghman | 2008-07-28 17:07:12 -0500
+ (Mon, 28 Jul 2008) | 15 lines Merged revisions 134161 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008)
+ | 7 lines Detect when sox fails to raise the volume, because sox
+ can't read the file. (closes issue #12939) Reported by:
+ rickbradley Patches: 20080728__bug12939.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: rickbradley ........
+ ................
+
+2008-07-28 19:55 +0000 [r134126] Mark Michelson <mmichelson@digium.com>
+
+ * /, configure, main/Makefile, configure.ac, CHANGES: Merged
+ revisions 134125 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 |
+ mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27
+ lines This commit compensates for buggy poll(2) implementations.
+ Asterisk has, for a long time, had its own implementation of
+ poll(2) which just used the input arguments to call select(2). In
+ 1.4, this internal implementation was used for Darwin systems.
+ This was removed in Asterisk trunk at some point, but it seems as
+ though this was not the right move to make. On Mac OS X, it
+ appears as though the poll used to gather CLI input does not
+ respond properly when connecting via a remote Asterisk console.
+ Reverting to the use of Asterisk's poll fixed the issue. Also,
+ there is now an option for the configure script,
+ --enable-internal-poll, which will allow for anyone to use
+ Asterisk's internal poll implementation in case they suspect that
+ their system's poll implementation is buggy. closes issue #11928)
+ Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded
+ by putnopvut (license 60) ........
+
+2008-07-28 16:49 +0000 [r134087] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_parkandannounce.c, main/loader.c, sample.call,
+ contrib/scripts/autosupport, build_tools/cflags.xml,
+ main/channel.c, apps/app_dahdibarge.c, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, doc/ss7.txt, /, main/features.c,
+ doc/osp.txt, main/file.c, pbx/pbx_config.c: Merged revisions
+ 134086 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 |
+ kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3
+ lines remove remaining Zaptel references in various places
+ ........
+
+2008-07-28 16:13 +0000 [r134052] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
+ /, apps/app_meetme.c, apps/app_dahdiscan.c: Merged revisions
+ 134050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 |
+ mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3
+ lines merging the zap_and_dahdi_trunk branch up to trunk ........
+
+2008-07-26 15:34 +0000 [r133942-133982] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, include/asterisk/doxyref.h, /: Include the
+ licensing page in 1.6.0 as well. Now, this page exists in 1.4,
+ trunk, and 1.6.0.
+
+ * /: unblock 133575
+
+ * /, main/devicestate.c: Merged revisions 133945-133946 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26
+ Jul 2008) | 6 lines ast_device_state() gets called in two
+ different ways. The first way is when called from elsewhere in
+ Asterisk to find the current state of a device. In that case, we
+ want to use the cached value if it exists. The other way is when
+ processing a device state change. In that case, we do not want to
+ check the cache because returning the last known state is counter
+ productive. ........ r133946 | russell | 2008-07-26 10:16:20
+ -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache
+ argument ........
+
+2008-07-25 22:09 +0000 [r133863-133905] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/asterisk.ldif, /: Merged revisions 133902 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25
+ Jul 2008) | 6 lines Update version (closes issue #13163) Reported
+ by: suretec Patches: asterisk.ldif uploaded by suretec (license
+ 70) ........
+
+2008-07-25 19:37 +0000 [r133804-133806] Brandon Kruse <bkruse@digium.com>
+
+ * /: Blocking revert of code changes in r133770
+
+ * main/http.c: Include the http_decode function from trunk to
+ replace the + with a space.
+
+2008-07-25 17:33 +0000 [r133694] Brandon Kruse <bkruse@digium.com>
+
+ * /: Blocking a fix from trunk for the function http_decode. 1.6.0
+ does not have this function.
+
+2008-07-25 17:26 +0000 [r133671] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /, channels/chan_agent.c, main/devicestate.c:
+ Merged revisions 133665 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008)
+ | 16 lines Merged revisions 133649 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008)
+ | 8 lines Fix some errant device states by making the devicestate
+ API more strict in terms of the device argument (only without the
+ unique identifier appended). (closes issue #12771) Reported by:
+ davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76
+ (license 14) Tested by: davidw, jvandal, murf ........
+ ................
+
+2008-07-25 15:01 +0000 [r133576-133580] Russell Bryant <russell@digium.com>
+
+ * /, LICENSE: Merged revisions 133579 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008)
+ | 18 lines Merged revisions 133578 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r133578 | russell | 2008-07-25 10:00:31 -0500
+ (Fri, 25 Jul 2008) | 10 lines Merged revisions 133577 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008)
+ | 2 lines Fix the IAX2 URI for calling Digium ........
+ ................ ................
+
+2008-07-25 14:41 +0000 [r133571-133574] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 133573 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul
+ 2008) | 15 lines Merged revisions 133572 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul
+ 2008) | 7 lines We need to make sure to null-terminate the "name"
+ portion of SIP URI parameters so that there are no bogus
+ comparisons. Thanks to bbryant for pointing this out. ........
+ ................
+
+2008-07-25 13:25 +0000 [r133567-133569] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 133568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 |
+ russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines
+ Minor coding guidelines tweaks ... - Use ast_strlen_zero in one
+ place - check for successful string comparison the way most of
+ Asterisk code does it ........
+
+2008-07-24 21:31 +0000 [r133524] Til