[asterisk-commits] lmadsen: tag 1.8.4-rc1 r308634 - /tags/1.8.4-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Feb 23 18:05:27 CST 2011
Author: lmadsen
Date: Wed Feb 23 18:05:22 2011
New Revision: 308634
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=308634
Log:
Importing files for 1.8.4-rc1 release.
Added:
tags/1.8.4-rc1/.lastclean (with props)
tags/1.8.4-rc1/.version (with props)
tags/1.8.4-rc1/ChangeLog (with props)
Added: tags/1.8.4-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.4-rc1/.lastclean?view=auto&rev=308634
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--- tags/1.8.4-rc1/ChangeLog (added)
+++ tags/1.8.4-rc1/ChangeLog Wed Feb 23 18:05:22 2011
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+2011-02-23 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.4-rc1 Released.
+
+2011-02-23 23:38 +0000 [r308622] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: sig_pri_new_ast_channel() should return NULL
+ when new_ast_channel() fails. (closes issue #18874) Reported by:
+ cmaj Patches:
+ patch-sig_pri-crash-possible-null-channel-pointer.diff.txt
+ uploaded by cmaj (license 830) JIRA SWP-3172
+
+2011-02-22 15:31 +0000 [r308526] Andrew Latham <lathama at gmail.com>
+
+ * main/http.c: Use ast_debug for console logging Guessed the log
+ levels based on info that level 3 is the soft roof. Can we create
+ a page / document to define the levels?
+
+2011-02-21 15:02 +0000 [r308416] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/udptl.c, /: Merged revisions 308414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600
+ (Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb
+ 2011) | 5 lines Properly check the bounds of arrays when decoding
+ UDPTL packets. Also, remove broken support for receiving UDPTL
+ packets larger than 16k. That shouldn't ever happen anyway.
+ AST-2011-002 FAX-281 ........ ................
+
+2011-02-21 14:24 +0000 [r308393] Andrew Latham <lathama at gmail.com>
+
+ * main/http.c: Add HTTP URI Debug logging and update notice enable
+ reporting of the request URI / URL in debugging change funny
+ debug note to a serious note.
+
+2011-02-19 14:06 +0000 [r308330] Andrew Latham <lathama at gmail.com>
+
+ * main/http.c: Add CSS MIME Type Modern browsers are checking for
+ the MIME Type of pages and in some cases will not load a file if
+ the type is wrong.
+
+2011-02-19 11:02 +0000 [r308288] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * utils: A few more (copies of) files to ignore in this directory.
+
+2011-02-18 00:07 +0000 [r308242] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323cDriver.c, addons/ooh323cDriver.h,
+ addons/chan_ooh323.c: added g729onlyA option for announce only
+ AnnexA g.729 codec in h.323 capabilities. Option can be global or
+ per user/peer.
+
+2011-02-16 20:21 +0000 [r308150] Paul Belanger <pabelanger at digium.com>
+
+ * addons/ooh323c/src/ooSocket.c: Fix FreeBSD builds.
+
+2011-02-16 07:57 +0000 [r308098] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooSocket.c: ifdef __linux__ keepalive
+ variables also
+
+2011-02-15 23:34 +0000 [r308010] Jason Parker <jparker at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 308007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r308007 | qwell | 2011-02-15 17:33:24 -0600
+ (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) |
+ 10 lines Fix regression that changed behavior of queues when
+ ringing a queue member. This reverts r298596, which was to fix a
+ highly bizarre and contrived issue with a queue member that
+ called into his own queue being transferred back into his own
+ queue. I couldn't reproduce that issue in any way. I think one of
+ the other recent transfer fixes actually fixed this. (closes
+ issue #18747) Reported by: vrban ........ ................
+
+2011-02-15 23:08 +0000 [r307970] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooSocket.c: include tcp keepalive socket calls
+ only on linux, freebsd and others don't have these options on
+ sockets.
+
+2011-02-15 19:52 +0000 [r307879-307962] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_dial.c: Don't crash when forcing caller id.
+
+ * channels/sig_pri.c, include/asterisk/ccss.h, channels/sig_pri.h,
+ channels/chan_dahdi.c, channels/chan_sip.c, main/ccss.c: No
+ response sent for SIP CC subscribe/resubscribe request. Asterisk
+ does not send a response if we try to subscribe for call
+ completion after we have received a 180 Ringing. You can only
+ subscribe for call completion when the call has been cleared.
+ When we receive the 180 Ringing, for this call, its
+ call-completion state is 'CC_AVAILABLE'. If we then send a
+ subscribe message to Asterisk, it trys to change the
+ call-completion state to 'CC_CALLER_REQUESTED'. Because this is
+ an invalid state change, it just ignores the message. The only
+ state Asterisk will accept our subscribe message is in the
+ 'CC_CALLER_OFFERED' state. Asterisk will go into the
+ 'CC_CALLER_OFFERED' when the SIP client clears the call by
+ sending a CANCEL. Asterisk should always send a response. Even if
+ its a negative one. The fix is to allow for the CCSS core to
+ notify a CC agent that a failure has occurred when CC is
+ requested. The "ack" callback is replaced with a "respond"
+ callback. The "respond" callback has a parameter indicating
+ either a successful response or a specific type of failure that
+ may need to be communicated to the requester. (closes issue
+ #18336) Reported by: GeorgeKonopacki Tested by: mmichelson,
+ rmudgett JIRA SWP-2633 (closes issue #18337) Reported by:
+ GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634
+
+2011-02-15 07:02 +0000 [r307750-307837] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * /, funcs/func_odbc.c: Merged revisions 307836 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011)
+ | 8 lines Need to retrieve the rows affected before using the
+ associated variable. (closes issue #18795) Reported by: irroot
+ Patches: 20110211__issue18795.diff.txt uploaded by tilghman
+ (license 14) Tested by: tilghman ........
+
+ * res/res_odbc.c, /: Merged revisions 307792 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011)
+ | 8 lines Increment usage count at first reference, to avoid a
+ race condition with many threads creating connections all at
+ once. (issue #18156) Reported by: asgaroth Patches:
+ 20110214__issue18156.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+ * apps/app_queue.c, apps/app_dial.c: Calling a gosub routine
+ defined in AEL from Dial/Queue ceased to work. A bug in AEL did
+ not distinguish between the "s" extension generated by AEL and an
+ "s" extension that was required to exist by the chan_dahdi (or
+ another channel) that was not supplied with a starting extension.
+ Therefore, AEL made incorrect assumptions about what commands
+ were permissable in the context. This was fixed by making AEL
+ generate a different extension name. However, Dial and Queue make
+ additional assumptions about the name of the default gosub
+ extension. Therefore, they needed to be brought into line with a
+ "macro" rendered by AEL (as a gosub), without breaking
+ traditional dialplans written without the aid of AEL. Related to
+ (issue #18480) Reported by: nivek (closes issue #18729) Reported
+ by: kkm Patches: 20110209__issue18729.diff.txt uploaded by
+ tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded
+ by kkm (license 888) Tested by: kkm
+
+2011-02-10 22:39 +0000 [r307536] Jason Parker <jparker at digium.com>
+
+ * main/asterisk.c, contrib/init.d/rc.debian.asterisk, /: Merged
+ revisions 307535 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r307535 | qwell | 2011-02-10 16:35:49 -0600
+ (Thu, 10 Feb 2011) | 15 lines Merged revisions 307534 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) |
+ 8 lines Remove color when executing commands via a remote
+ console. Essentially this makes '-x' imply '-n' on rasterisk.
+ This was done in a different and incomplete way previously, which
+ I'm reverting here. (issue #18776) Reported by: alecdavis
+ ........ ................
+
+2011-02-10 18:50 +0000 [r307509] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h:
+ Corrections for properly work with H.323v2 (older) endpoints and
+ other small fixes. Interpret remote side H.225 version.
+ Corrections for H.323v2 endpoints: don't start TCS and MSD before
+ connect, don't start TCS and MSD by accepting H.245 connection,
+ start TCS and MSD by StartH245 facility message. Other fixes: fix
+ non zeroended remoteDisplayName issue, small fixes in call
+ clearing by closing H.245 connection, tcp keepalive introduced on
+ TCP connections (now is hardcoded, will be configurable in the
+ future), don't force H.245tunneling if FastStart is active, don't
+ send Alerting singal more than once per call. (issue 0018542)
+ Reported by: vmikhelson Patches: issue18542-final-3.patch
+ uploaded by may213 (license 454) Tested by: vmikhelson
+
+2011-02-10 17:44 +0000 [r307467] Mark Michelson <mmichelson at digium.com>
+
+ * configs/ccss.conf.sample: Fix a gaffe in the CCSS sample
+ configuration. Discovered by Philippe Lindheimer and pointed out
+ on #asterisk-dev
+
+2011-02-09 21:44 +0000 [r307314] Andrew Latham <lathama at gmail.com>
+
+ * contrib/init.d/rc.debian.asterisk: Disable color during running
+ test (closes issue #18776) Reported by: alecdavis Patches:
+ ast_deb_init.diff uploaded by lathama (license 1028) Tested by:
+ andrel, lathama
+
+2011-02-09 21:06 +0000 [r307228-307273] Jeff Peeler <jpeeler at digium.com>
+
+ * main/astobj2.c: Add missing debug info for ao2_link for use with
+ REF_DEBUG in ao2 callback. (closes issue #18758) Reported by:
+ rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by
+ rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by
+ rgagnon (license 1202)
+
+ * /, main/features.c: Merged revisions 307227 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011)
+ | 11 lines Make sure to set parking dial context for non-default
+ parking lots. Since parking_con_dial isn't settable, set all
+ parking lots to "park-dial". (closes issue #17946) Reported by:
+ bluecrow76 Patches:
+ asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by
+ bluecrow76 (license 270) modified by me ........
+
+2011-02-09 05:39 +0000 [r307142] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * main/lock.c: Initialize tracking variable in structure properly.
+ Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by
+ me.)
+
+2011-02-08 21:24 +0000 [r307092] Jason Parker <jparker at digium.com>
+
+ * main/logger.c: Fix issue with verbose messages not showing on
+ remote console. This code was reworked recently, and since the
+ logchannel list hadn't been created yet at this point, and it was
+ a verbose message, it was being dropped on the floor. Now it'll
+ continue on to where it should be handled. (closes issue #18580)
+ Reported by: pabelanger
+
+2011-02-08 21:13 +0000 [r307065] Mark Michelson <mmichelson at digium.com>
+
+ * main/ccss.c: Add a couple of useful channel variables for the CC
+ recall macro. CC_EXTEN and CC_CONTEXT will allow you to determine
+ the channel and context that will be called when the recall
+ occurs.
+
+2011-02-08 20:22 +0000 [r306999] Andrew Latham <lathama at gmail.com>
+
+ * doc/asterisk.sgml, doc/asterisk.8, configs/asterisk.conf.sample,
+ configs/voicemail.conf.sample: Documentation Updates Note default
+ polling setting in voicemail.conf Add missing config to
+ asterisk.conf Update manpage (issue #16505) Reported by: tzafrir
+ Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir
+ (license 46) Tested by: lathama, tzafrir
+
+2011-02-08 20:18 +0000 [r306979] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 306973 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306973 | twilson | 2011-02-08 12:14:09 -0800
+ (Tue, 08 Feb 2011) | 9 lines Merged revisions 306972 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08
+ Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with
+ pedantic=yes ........ ................
+
+2011-02-08 19:41 +0000 [r306866-306967] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 306966 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600
+ (Tue, 08 Feb 2011) | 9 lines Merged revisions 306965 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08
+ Feb 2011) | 1 line fix this line again ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 306961 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600
+ (Tue, 08 Feb 2011) | 15 lines Merged revisions 306960 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011)
+ | 9 lines Backup file storing message duration is not used with
+ IMAP_STORAGE, remove code. The message duration is stored in the
+ body of the email when using IMAP_STORAGE, so nothing needs to
+ happen with the backup file. (closes issue #18718) Reported by:
+ kerframil ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 306865 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600
+ (Tue, 08 Feb 2011) | 9 lines Merged revisions 306864 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08
+ Feb 2011) | 1 line make this safer and fully correct, pointed out
+ by Steve Davis ........ ................
+
+2011-02-08 01:45 +0000 [r306826] Andrew Latham <lathama at gmail.com>
+
+ * UPGRADE.txt, include/asterisk/manager.h, doc/asterisk.sgml,
+ include/asterisk/doxygen/mantisworkflow.h: Documentation Updates.
+ More updates to the removed doc folder and start updates to the
+ man page. (issue #16505) Reported by: tzafrir Tested by: lathama
+
+2011-02-07 22:43 +0000 [r306619-306674] Terry Wilson <twilson at digium.com>
+
+ * /, main/features.c: Merged revisions 306673 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306673 | twilson | 2011-02-07 14:40:20 -0800
+ (Mon, 07 Feb 2011) | 17 lines Merged revisions 306672 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011)
+ | 10 lines Don't try to pickup a call in the middle of a
+ masquerade If A calls B which doesn't answer and C & D both try
+ to do a call pickup, it is possible for ast_pickup_call to answer
+ the call, then fail to masquerade one of the calls because the
+ other one is already in the process of masquerading. This patch
+ checks to see if the channel is in the process of masquerading
+ before call before selecting it for a pickup. Review:
+ https://reviewboard.asterisk.org/r/1094/ ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 306618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306618 | twilson | 2011-02-07 13:59:54 -0800
+ (Mon, 07 Feb 2011) | 17 lines Merged revisions 306617 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011)
+ | 10 lines Don't allow a REFER w/replaces to replace its own
+ dialog Asterisk currently accepts a REFER with a Refer-To with an
+ embedded Replaces header that matches the dialog of the REFER.
+ This would be a situation like A calls B, A calls C, A transfers
+ B to A, which is just silly. This patch makes the transfer fail
+ instead of making Asterisk freak out and forget to hang other
+ channels up. Review: https://reviewboard.asterisk.org/r/1093/
+ ........ ................
+
+2011-02-07 17:36 +0000 [r306575] Mark Michelson <mmichelson at digium.com>
+
+ * main/ccss.c: Rearrange a bit of code in the generic CC recall
+ operation. By waiting to call the callback macro after the
+ CC_INTERFACES, extension, priority, and context have been set,
+ this information can be accessed more easily within the callback
+ macro. Reported by Philippe Lindheimer.
+
+2011-02-04 19:24 +0000 [r306356] Jason Parker <jparker at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 306346 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) |
+ 9 lines Don't fallthrough to 'unknown' in the 'ringing' case.
+ This could cause improper exits from the queue. (closes issue
+ #18499) Reported by: zaltar Patches: app_queue.patch uploaded by
+ zaltar (license 1148) ........
+
+2011-02-04 18:53 +0000 [r306324] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c: Don't send redirecting updates
+ to the caller if the dialplan forked the call. Each fork in the
+ dial could be redirected and confuse the caller. For ISDN the
+ DivLeg1 and DivLeg3 messages would get confused because ISDN
+ redirects calls in sequence not in parallel. * Also fixed a
+ formatting inconsistency in app_dial.c and make a warning message
+ more useful about what frame type could not be written.
+
+2011-02-03 23:49 +0000 [r306215] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: Fix SIP deadlock involving state changes.
+ Once again a call to pbx_builtin_getvar_helper (and
+ pbx_builtin_setvar_helper) has caused locking problems. Both of
+ these functions lock the channel when the channel argument is
+ passed in! In this case, the suspected problem (the backtrace
+ makes it impossible to tell) was the private being locked in
+ sip_set_rtp_peer and then: transmit_reinvite_with_sdp
+ try_suggested_sip_codec pbx_builtin_getvar_helper (Traced to
+ verify that the fix was only required in 1.8 and later.) (closes
+ issue #18491) Reported by: cmaj Patches:
+ chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license
+ 830) Tested by: cmaj
+
+2011-02-03 21:03 +0000 [r306127] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 306126 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r306126 | twilson | 2011-02-03 12:56:00 -0800
+ (Thu, 03 Feb 2011) | 16 lines Merged revisions 306119 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011)
+ | 9 lines Set hangup cause in local_hangup When a call involves a
+ local channel (like SIP -> Local -> SIP), the hangup cause was
+ not being set. This resulted in SIP channels sometimes getting a
+ 503 error instead of a 486 when the far side sent a busy. In
+ Asterisk 1.8+ this also can cause issues with CCSS that involve a
+ local channel. This patch sets the hangupcause for one side of
+ the local channel to the other in local_hangup for outbound
+ calls. ........ ................
+
+2011-02-03 20:50 +0000 [r306124] Jeff Peeler <jpeeler at digium.com>
+
+ * /, main/features.c: Merged revisions 306123 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011)
+ | 10 lines Set exception on channel in parking thread when
+ POLLPRI event detected. This is done just to make the code be
+ equivalent to the old select code. As noted in 303106 the same
+ issue was already fixed in this branch, but the exception was not
+ set on the channel in the case of POLLPRI. The reason that this
+ did not cause a problem here is because in 122923 the check in
+ __ast_read to check the exception flag was removed. (related to
+ #18637) ........
+
+2011-02-03 15:50 +0000 [r305987] Andrew Latham <lathama at gmail.com>
+
+ * phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample, /:
+ res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
+ (issue #18713) Reported by: lathama Patches: snom_dir.diff
+ uploaded by lathama (license 1028) Tested by: lathama
+
+2011-02-03 00:24 +0000 [r305923] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, main/manager.c, /, channels/chan_sip.c,
+ apps/app_sendtext.c: Merged revisions 305889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600
+ (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011)
+ | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null
+ terminator in the buffer length. When the frame is queued it is
+ copied. If the null terminator is not part of the frame buffer
+ length, the receiver could see garbage appended onto it. * Add
+ channel lock protection with ast_sendtext(). * Fixed AMI SendText
+ action ast_sendtext() return value check. ........
+ ................
+
+2011-02-02 20:05 +0000 [r305844] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * funcs/func_env.c: Eliminate a file descriptor leak when using the
+ FILE() dialplan function. (closes issue #18731) Reported by:
+ marioabajo
+
+2011-02-02 19:27 +0000 [r305753-305838] Andrew Latham <lathama at gmail.com>
+
+ * apps/app_externalivr.c, configs/sip.conf.sample,
+ configs/skinny.conf.sample, configs/h323.conf.sample,
+ configs/sla.conf.sample, apps/app_voicemail.c,
+ configs/iax.conf.sample, funcs/func_enum.c,
+ configs/dundi.conf.sample, funcs/func_callcompletion.c,
+ configs/mgcp.conf.sample, configs/iaxprov.conf.sample,
+ configs/unistim.conf.sample: Replacing doc/* and asterisk.pdf
+ with wiki links Adding links to http(s)://wiki.asterisk.org
+
+ * configs/ccss.conf.sample, configs/sip.conf.sample,
+ configs/skinny.conf.sample, main/config.c,
+ configs/h323.conf.sample, configs/sla.conf.sample,
+ main/ast_expr2.fl, res/res_srtp.c,
+ configs/chan_dahdi.conf.sample, configs/extconfig.conf.sample,
+ configs/res_snmp.conf.sample, main/ast_expr2f.c,
+ res/res_timing_dahdi.c: Replacing doc/* with wiki links Adding
+ links to http(s)://wiki.asterisk.org
+
+ * channels/chan_sip.c: Replace link to old doc with new wiki page.
+ Link to
+ https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
+
+2011-02-01 22:48 +0000 [r305692] Jason Parker <jparker at digium.com>
+
+ * channels/chan_iax2.c: Reverse sense of an error test when reading
+ from astdb. (closes issue #18545) Reported by: jcovert Patches:
+ chan_iax2.c.patch uploaded by jcovert (license 551)
+
+2011-02-01 21:14 +0000 [r305649] Andrew Latham <lathama at gmail.com>
+
+ * configs/sip.conf.sample: SIP Configuration Documentation sip show
+ settings reports qualifyfreq in milliseconds. sip.conf configures
+ qualifyfreg in seconds.
+
+2011-02-01 19:23 +0000 [r305603] Brett Bryant <bbryant at digium.com>
+
+ * cel/cel_pgsql.c: Add a possible solution to a customer problem
+ with reloading cel_pgsql.so quickly.
+
+2011-02-01 18:02 +0000 [r305560] Andrew Latham <lathama at gmail.com>
+
+ * CHANGES, Makefile, README, /: doc/tex dir removed, but
+ corresponding entries still exists Update README, CHANGES, and
+ Makefile. Direct users to http://wiki.asterisk.org for
+ documentation or to the AST.txt and AST.pdf included in the
+ tarball. (closes issue #18443) Reported by: bas Patches:
+ changes.diff uploaded by lathama (license 1028) readme.diff
+ uploaded by lathama (license 1028) Tested by: lathama bas
+
+2011-02-01 17:04 +0000 [r305473] Jason Parker <jparker at digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 305472 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305472 | qwell | 2011-02-01 11:02:09 -0600
+ (Tue, 01 Feb 2011) | 16 lines Merged revisions 305471 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) |
+ 9 lines Close file descriptor for timing source when a MOH class
+ gets destroyed. (closes issue #18457) Reported by: mcallist
+ Patches: 18457-closetimer.diff uploaded by qwell (license 4)
+ 18457-closetimer_trunk.diff uploaded by qwell (license 4) Tested
+ by: qwell, loloski ........ ................
+
+2011-02-01 00:01 +0000 [r305343] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, /: Merged revisions 305342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600
+ (Mon, 31 Jan 2011) | 14 lines Merged revisions 305341 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011)
+ | 7 lines Obtain the pri lock for PRI queue counters. Need to
+ obtain the pri lock when calling pri_dump_info_str() to avoid a
+ reentrancy problem when calculating the Q.921 Q count statistic.
+ JIRA AST-484 ........ ................
+
+2011-01-31 23:07 +0000 [r305131-305254] Jason Parker <jparker at digium.com>
+
+ * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305253
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305253 | qwell | 2011-01-31 16:59:34 -0600
+ (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) |
+ 10 lines Prevent a crash when dialing a technology with no
+ destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers
+ already had code to prevent this. The attempt that app_dial was
+ making to prevent it was not correct, so I fixed that. (closes
+ issue #18371) Reported by: gbour Patches: 18371.patch uploaded by
+ gbour (license 1162) ........ ................
+
+ * configs/sip.conf.sample, main/tcptls.c: Add alternative name for
+ config option. The SIP sample configuration had "tlscadir" as the
+ option name, but chan_sip used the more correct "tlscapath". Now
+ both are accepted. Discovered (sort of) by a user on IRC in
+ #asterisk
+
+ * res/res_musiconhold.c: Fix compile error. pseudofd no longer
+ exists.
+
+ * res/res_musiconhold.c, /: Merged revisions 305130 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r305130 | qwell | 2011-01-31 14:59:37 -0600
+ (Mon, 31 Jan 2011) | 9 lines Merged revisions 305129 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan
+ 2011) | 2 lines Set file descriptors to -1 on creation, so that
+ we don't see weirdness later. ........ ................
+
+2011-01-31 13:56 +0000 [r305083] Andrew Latham <lathama at gmail.com>
+
+ * main/http.c: Asterisk HTTP response Content-type Address content
+ type for BSD and other platforms (closes issue #18456) Reported
+ by: alexo Patches: asterisk18_http.patch uploaded by alexo
+ (license 1175) Tested by: alexo
+
+2011-01-31 07:51 +0000 [r304950-305040] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * include/asterisk/lock.h: Use the non-specific API aliases, to
+ avoid a problem with building the utils directory.
+
+ * apps/app_voicemail.c, /: Merged revisions 304978 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304978 | tilghman | 2011-01-31 01:25:14 -0600
+ (Mon, 31 Jan 2011) | 9 lines Merged revisions 304952 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31
+ Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined.
+ ........ ................
+
+ * main/utils.c, include/asterisk/lock.h, .cleancount, main/lock.c,
+ main/heap.c: Change mutex tracking so that it only consumes
+ memory in the core mutex object when it's actually being used.
+ This reduces the overall size of a mutex which was 3016 bytes
+ before this back down to 216 bytes (this is on 64-bit Linux with
+ a glibc-implemented mutex). The exactness of the numbers here may
+ vary slightly based upon how mutexes are implemented on a
+ platform, but the long and short of it is that prior to this
+ commit, chan_iax2 held down 98MB of memory on a 64-bit system for
+ nothing more than a table of 32767 locks. After this commit, the
+ same table occupies a mere 7MB of memory. (closes issue #18194)
+ Reported by: job Patches: 20110124__issue18194.diff.txt uploaded
+ by tilghman (license 14) Tested by: tilghman Review:
+ https://reviewboard.asterisk.org/r/1066
+
+2011-01-30 00:11 +0000 [r304908] Andrew Latham <lathama at gmail.com>
+
+ * apps/app_externalivr.c, apps/app_queue.c, apps/app_voicemail.c,
+ funcs/func_realtime.c, res/res_calendar.c,
+ funcs/func_callcompletion.c: Add Function and Application
+ Relationships to documentation Add and extend the see-also
+ sections to the documentation for applications and functions in
+ an effort to expand the online documentation of the wiki. Also
+ check for and update any links to moved documentation in the doc
+ folder.
+
+2011-01-29 23:07 +0000 [r304638-304866] Sean Bright <sean at malleable.com>
+
+ * res/res_config_ldap.c, /: Merged revisions 304865 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r304865 | seanbright | 2011-01-29 18:05:25 -0500 (Sat,
+ 29 Jan 2011) | 7 lines Plug some memory leaks in the LDAP
+ realtime driver. (closes issue #18435) Reported by: zaltar
+ Patches: res_config_ldap.patch uploaded by zaltar (license 1148)
+ ........
+
+ * /, apps/app_meetme.c: Merged revisions 304776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan
+ 2011) | 15 lines If we fail to allocate our announcement objects,
+ make sure we don't leak objects. The majority of this patch was
+ committed already in r304726 and r304729. (issue #18225) Reported
+ by: kenji (issue #18444) Reported by: junky (closes issue #18343)
+ Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz
+ (license 834) ........
+
+ * /, apps/app_meetme.c: Merged revisions 304773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan
+ 2011) | 9 lines When we pass the S() or L() options to MeetMe,
+ make sure that we honor C as well. Without this patch, if the
+ user was kicked from the conference via the S() or L() mechanism,
+ we would just hang up on them even if we also passed C (continue
+ in dialplan when kicked). With this patch we honor the C flag in
+ those cases. (closes issue #17317) Reported by: var ........
+
+ * /, apps/app_meetme.c: Merged revisions 304729 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan
+ 2011) | 15 lines Make sure that we unref the correct object when
+ ejecting the most recent caller. Currently, when we kick the last
+ user to enter, we decrement our own reference count which results
+ in a crash when we kick another user or when we exit the
+ conference ourselves. This will fix #18225 in 1.8 and trunk, but
+ that particular bug does not exist in 1.6.2. (closes issue
+ #18225) Reported by: kenji Patches: issue18225.patch uploaded by
+ seanbright (license 71) Tested by: seanbright ........
+
+ * /, apps/app_meetme.c: Merged revisions 304726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan
+ 2011) | 9 lines Fix user reference leak in MeetMe. We were
+ unlinking the user from the conferences user container, but not
+ decrementing the reference count of the user as well, resulting
+ in a leak. (closes issue #18444) Reported by: junky Tested by:
+ seanbright ........
+
+ * /, apps/app_meetme.c: Merged revisions 304659,304682 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri,
+ 28 Jan 2011) | 5 lines Don't leak references if we can't create a
+ pseudo channel for mixing in MeetMe. If there was a problem
+ allocating a pseudo channel when building our meetme, we weren't
+ destroying our user container or destroying the mutexes that we
+ created. ........ r304682 | seanbright | 2011-01-28 17:38:05
+ -0500 (Fri, 28 Jan 2011) | 2 lines Revert part of the previous
+ commit that snuck in. ........
+
+ * main/acl.c: Restore some conditionals that we lost in r277814.
+ There are some cases where ast_append_ha() is called with a NULL
+ instead of a valid int pointer. So if we get a NULL, don't try to
+ dereference it. (closes issue #18162) Reported by: imcdona
+ Patches: issue0018162.patch uploaded by pabelanger (license 224)
+ Tested by: enegaard
+
+2011-01-27 19:08 +0000 [r304554] Richard Mudgett <rmudgett at digium.com>
+
+ * main/ccss.c: Warning message if CALLCOMPLETION(cc_callback_macro
+ or cc_agent_dialstring) are empty. Test if the value pointer is
+ not NULL instead of not ast_strlen_zero().
+
+2011-01-27 17:03 +0000 [r304462-304466] Jason Parker <jparker at digium.com>
+
+ * /, configure, configure.ac: Merged revisions 304465 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304465 | qwell | 2011-01-27 11:01:24 -0600
+ (Thu, 27 Jan 2011) | 16 lines Merged revisions 304464 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) |
+ 9 lines Fix default prefix=/usr regression on non-Linux systems.
+ This partially reverts a change made in branches/1.4/ r267759,
+ which will cause issue #17013 to be reopened. This issue was
+ pointed out by a user on #asterisk, who helpfully discovered that
+ paths were being set incorrectly. To truly understand what was
+ wrong, one should run: svn diff --force -c<this revision>
+ configure ........ ................
+
+ * /, configure: Merged revisions 304461 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304461 | qwell | 2011-01-27 10:48:00 -0600
+ (Thu, 27 Jan 2011) | 9 lines Merged revisions 304460 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan
+ 2011) | 1 line Rerun bootstrap.sh with no changes, so that it is
+ more obvious what my next commit changes. ........
+ ................
+
+2011-01-26 22:27 +0000 [r304339] Jeff Peeler <jpeeler at digium.com>
+
+ * /, main/features.c: Merged revisions 304338 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26 Jan 2011)
+ | 2 lines Change delimiter used internally for GOTO_ON_BLINDXFR
+ to commas to match 76703. ........
+
+2011-01-26 21:02 +0000 [r304251] Mark Michelson <mmichelson at digium.com>
+
+ * main/udptl.c, /: Merged revisions 304250 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304250 | mmichelson | 2011-01-26 15:02:10 -0600
+ (Wed, 26 Jan 2011) | 9 lines Merged revisions 304242 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed,
+ 26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in
+ ast_udptl ........ ................
+
+2011-01-26 20:43 +0000 [r304245] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/sip/include/sip.h,
+ channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c,
+ channels/sip/reqresp_parser.c: Merged revisions 304244 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600
+ (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan
+ 2011) | 6 lines This patch modifies chan_sip to route responses
[... 28423 lines stripped ...]
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