[svn-commits] lmadsen: tag 1.6.0.11-rc2 r209443 - /tags/1.6.0.11-rc2/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 28 11:08:45 CDT 2009
Author: lmadsen
Date: Tue Jul 28 11:08:41 2009
New Revision: 209443
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=209443
Log:
Importing files for 1.6.0.11-rc2 release.
Added:
tags/1.6.0.11-rc2/.lastclean (with props)
tags/1.6.0.11-rc2/.version (with props)
tags/1.6.0.11-rc2/ChangeLog (with props)
Added: tags/1.6.0.11-rc2/.lastclean
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.0.11-rc2/.lastclean?view=auto&rev=209443
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--- tags/1.6.0.11-rc2/ChangeLog (added)
+++ tags/1.6.0.11-rc2/ChangeLog Tue Jul 28 11:08:41 2009
@@ -1,0 +1,54592 @@
+2009-07-28 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.6.0.11-rc2
+
+2009-07-28 12:01 +0000 [r209394] Kevin P. Fleming <kpfleming at digium.com>
+
+ * apps/app_fax.c: Correct error in backport of latest app_fax
+ fixes.
+
+2009-07-28 00:19 +0000 [r209325] Tilghman Lesher <tlesher at 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 [r209259-209280] Kevin P. Fleming <kpfleming at 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 20:23 +0000 [r209221] David Brooks <dbrooks at digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /,
+ include/asterisk/module.h, main/features.c, res/res_agi.c,
+ res/res_jabber.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 20:16 +0000 [r209133-209198] Mark Michelson <mmichelson at digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
+ 2009) | 9 lines Honor channel's music class when using realtime
+ music on hold. (closes issue #15051) Reported by: alexh Patches:
+ 15051.patch uploaded by mmichelson (license 60) Tested by: alexh
+ ........
+
+ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
+ 209132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
+ 2009) | 24 lines Merged revisions 209131 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
+ 2009) | 18 lines Allow for UDPTL to use only even-numbered ports
+ if desired. There are some VoIP providers out there that will not
+ accept SDP offers with odd numbered UDPTL ports. While it is my
+ personal opinion that these VoIP providers are misinterpreting
+ RFC 2327, it really is not a big deal to play along with their
+ silly little games. Of course, since restricting UDPTL ports to
+ only even numbers reduces the range of available ports by half,
+ so the option to use only even port numbers is off by default. A
+ user can enable the behavior by setting use_even_ports=yes in
+ udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
+ 15182.patch uploaded by mmichelson (license 60) Tested by:
+ CGMChris ........ ................
+
+2009-07-27 16:06 +0000 [r209061] David Brooks <dbrooks at digium.com>
+
+ * res/res_smdi.c, pbx/pbx_dundi.c: Just replacing typos "recieved"
+ with "received". From issue #15360, forgot to apply to trunk and
+ other branches.
+
+2009-07-27 15:39 +0000 [r209057] Kevin P. Fleming <kpfleming at 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 01:21 +0000 [r208925] Jeff Peeler <jpeeler at digium.com>
+
+ * /, main/translate.c, channels/chan_iax2.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 ........ ................
+
+2009-07-25 06:24 +0000 [r208752] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_skinny.c, main/translate.c,
+ channels/chan_iax2.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 18:49 +0000 [r208594] Russell Bryant <russell at 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 18:31 +0000 [r208589] Mark Michelson <mmichelson at 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 15:04 +0000 [r208468-208549] Kevin P. Fleming <kpfleming at 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. ........
+
+ * UPGRADE.txt: Use correct formatting for T.38 change note in
+ UPGRADE.txt
+
+ * 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, include/asterisk/frame.h: Merged
+ revisions 208464 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 |
+ kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46
+ lines Rework of T.38 negotiation and UDPTL API to address
+ interoperability problems Over the past couple of months, a
+ number of issues with Asterisk negotiating (and successfully
+ completing) T.38 sessions with various endpoints have been found.
+ This patch attempts to address many of them, primarily focused
+ around ensuring that the endpoints' MaxDatagram size is honored,
+ and in addition by ensuring that T.38 session parameter
+ negotiation is performed correctly according to the ITU T.38
+ Recommendation. The major changes here are: 1) T.38 applications
+ in Asterisk (app_fax) only generate/receive IFP packets, they do
+ not ever work with UDPTL packets. As a result of this, they
+ cannot be allowed to generate packets that would overflow the
+ other endpoints' MaxDatagram size after the UDPTL stack adds any
+ error correction information. With this patch, the application is
+ told the maximum *IFP* size it can generate, based on a
+ calculation using the far end MaxDatagram size and the active
+ error correction mode on the T.38 session. The same is true for
+ sending *our* MaxDatagram size to the remote endpoint; it is
+ computed from the value that the application says it can accept
+ (for a single IFP packet) combined with the active error
+ correction mode. 2) All treatment of T.38 session parameters as
+ 'capabilities' in chan_sip has been removed; these parameters are
+ not at all like audio/video stream capabilities. There are strict
+ rules to follow for computing an answer to a T.38 offer, and
+ chan_sip now follows those rules, using the desired parameters
+ from the application (or channel) that wants to accept the T.38
+ negotiation. 3) chan_sip now stores and forwards
+ ast_control_t38_parameters structures for tracking 'our' and
+ 'their' T.38 session parameters; this greatly simplifies
+ negotiation, especially for pass-through calls. 4) Since T.38
+ negotiation without specifying parameters or receiving the final
+ negotiated parameters is not very worthwhile, the AST_CONTROL_T38
+ control frame has been removed. A note has been added to
+ UPGRADE.txt about this removal, since any out-of-tree
+ applications that use it will no longer function properly until
+ they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
+ https://reviewboard.asterisk.org/r/310/ ........
+
+2009-07-23 19:35 +0000 [r208389] Mark Michelson <mmichelson at 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 19:23 +0000 [r208384] Jeff Peeler <jpeeler at 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 [r208264-208316] Mark Michelson <mmichelson at 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 ........ ................
+
+2009-07-21 22:47 +0000 [r207947] Tilghman Lesher <tlesher at 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 20:27 +0000 [r207783-207860] Jeff Peeler <jpeeler at 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 r207636, this approach could
+ potentially block for an unacceptable amount of time.
+
+2009-07-21 14:30 +0000 [r207725] Mark Michelson <mmichelson at 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 13:39 +0000 [r207683] Kevin P. Fleming <kpfleming at digium.com>
+
+ * funcs/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile,
+ Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, /,
+ main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
+ Makefile.rules, pbx/Makefile, res/Makefile, channels/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 04:38 +0000 [r207636] Jeff Peeler <jpeeler at 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 19:55 +0000 [r207425] Mark Michelson <mmichelson at 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 16:37 +0000 [r207362] Russell Bryant <russell at 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 01:35 +0000 [r207286] Richard Mudgett <rmudgett at 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 19:38 +0000 [r207097-207157] Jeff Peeler <jpeeler at 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 [r206871-207032] David Vossel <dvossel at 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 ........ ................
+
+ * 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 16:52 +0000 [r206809] Tilghman Lesher <tlesher at 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 [r206775] David Vossel <dvossel at 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 21:34 +0000 [r206762] Richard Mudgett <rmudgett at 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 [r206705] David Vossel <dvossel at 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 16:02 +0000 [r206637] Sean Bright <sean at 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 20:22 +0000 [r206585] Tilghman Lesher <tlesher at 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 18:17 +0000 [r206555] Richard Mudgett <rmudgett at 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 14:54 +0000 [r206387] Russell Bryant <russell at 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 01:25 +0000 [r206369] Richard Mudgett <rmudgett at 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
+ ........ ................
+
+2009-07-10 22:50 +0000 [r206017] David Vossel <dvossel at 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 18:44 +0000 [r205940] Kevin P. Fleming <kpfleming at 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 17:44 +0000 [r205879-205880] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix build.
+
+ * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
[... 53917 lines stripped ...]
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