[asterisk-commits] lmadsen: tag 1.8.0-rc1 r288496 - /tags/1.8.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 22 16:17:20 CDT 2010
Author: lmadsen
Date: Wed Sep 22 16:17:16 2010
New Revision: 288496
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=288496
Log:
Importing files for 1.8.0-rc1 release.
Added:
tags/1.8.0-rc1/.lastclean (with props)
tags/1.8.0-rc1/.version (with props)
tags/1.8.0-rc1/ChangeLog (with props)
Added: tags/1.8.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-rc1/.lastclean?view=auto&rev=288496
==============================================================================
--- tags/1.8.0-rc1/.lastclean (added)
+++ tags/1.8.0-rc1/.lastclean Wed Sep 22 16:17:16 2010
@@ -1,0 +1,3 @@
+38
+
+
Propchange: tags/1.8.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.8.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.8.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.8.0-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-rc1/.version?view=auto&rev=288496
==============================================================================
--- tags/1.8.0-rc1/.version (added)
+++ tags/1.8.0-rc1/.version Wed Sep 22 16:17:16 2010
@@ -1,0 +1,1 @@
+1.8.0-rc1
Propchange: tags/1.8.0-rc1/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.8.0-rc1/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.8.0-rc1/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.8.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-rc1/ChangeLog?view=auto&rev=288496
==============================================================================
--- tags/1.8.0-rc1/ChangeLog (added)
+++ tags/1.8.0-rc1/ChangeLog Wed Sep 22 16:17:16 2010
@@ -1,0 +1,24555 @@
+2010-09-22 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.0-rc1 Released.
+
+2010-09-22 17:49 +0000 [r288345-288418] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288417 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500
+ (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
+ | 5 lines RFC3261 section 12.2 explicitly says out of order
+ requests are responded with a 500 Server Internal Error response.
+ ABE-2458 ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 288344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500
+ (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22
+ Sep 2010) | 2 lines During check_pendings, if the dialog is
+ terminated with a CANCEL, change the invitestate to INV_CANCEL
+ like in sip_hangup. ........ ................
+
+2010-09-22 16:45 +0000 [r288341] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 288340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288340 | russell | 2010-09-22 11:44:13 -0500
+ (Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
+ | 11 lines Fix a 100% CPU consumption problem when setting
+ console=yes in asterisk.conf. The handling of -c and console=yes
+ should be the same, but they were not. When you specify -c, it
+ sets both a flag for console module and for asterisk not to
+ fork() off into the background. The handling of console=yes only
+ set console mode, so you would end up with a background process()
+ trying to run the Asterisk console and freaking out since it
+ didn't have anything to read input from. Thanks to beagles for
+ reporting and helping debug the problem! ........
+ ................
+
+2010-09-22 15:14 +0000 [r288268] Tilghman Lesher <tlesher at digium.com>
+
+ * UPGRADE.txt, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /:
+ Merged revisions 288267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288267 | tilghman | 2010-09-22 10:11:09 -0500
+ (Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
+ | 9 lines Allow the encoding to be set, in case local charset
+ does not agree with database. (closes issue #16940) Reported by:
+ jamicque Patches: 20100827__issue16940.diff.txt uploaded by
+ tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+ r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
+ | 5 lines Document addition of encoding parameter. (issue #16940)
+ Reported by: jamicque ........ ................
+
+2010-09-22 00:06 +0000 [r288194] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 288193 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500
+ (Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010)
+ | 26 lines In chan_iax2.c:schedule_delivery() calls
+ ast_bridged_channel() on an unlocked channel. Near the beginning
+ of schedule_delivery(), ast_bridged_channel() is called on
+ iaxs[fr->callno]->owner. However, the channel is not locked,
+ which can result in ast_bridged_channel() crashing should
+ owner->tech change to a technology that doesn't implement
+ bridged_channel. I also fixed the other calls to
+ ast_bridged_channel() in chan_iax2.c since the owner lock was not
+ held there either. Converted the existing channel deadlock
+ avoidance to use iax2_lock_owner(). Using the new function
+ simplified some awkward code. In the process of fixing the
+ locking on ast_bridged_channel(), I also found a memory leak in
+ socket_process() for v1.6.2 and v1.8. The local struct variable
+ ies.vars is not freed on early/abnormal function exits. (closes
+ issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
+ uploaded by rmudgett (license 664) Review:
+ https://reviewboard.asterisk.org/r/926/ ........ ................
+
+2010-09-21 22:57 +0000 [r288159] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500
+ (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
+ | 15 lines Try both the encoded and unencoded subscription URI
+ for a match in hints. When a phone sends an encoded URI for a
+ subscription, the URI is not matched with the actual hint that is
+ in decoded format. For example, if we have an extension with a
+ hint that is named: "#5601" or "*5601", the subscription will
+ work fine if the phone subscribes with an already decoded URI,
+ but when it's decoded like "%255601" or "%2A5601", Asterisk is
+ unable to match it with the correct hint. (closes issue #17785)
+ Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
+ uploaded by tilghman (license 14) Tested by: ramonpeek ........
+ ................
+
+2010-09-21 22:26 +0000 [r288157] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 288147 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue,
+ 21 Sep 2010) | 9 lines Setup timer before set_config(). (closes
+ issue #18019) Reported by: Netview Patches: issue_0018019.patch
+ uploaded by pabelanger (license 224) Tested by: Netview ........
+
+2010-09-21 21:03 +0000 [r288079-288082] Richard Mudgett <rmudgett at digium.com>
+
+ * doc/tex/partymanip.tex: Add note in party manipulation chapter on
+ interception macros.
+
+ * apps/app_queue.c, apps/app_dial.c: Simplify locking code for
+ REDIRECTING interception macro when forwarding a call. Simplified
+ the locking code by using a local copy of the redirecting party
+ information in app_dial.c:do_forward() and
+ app_queue.c:wait_for_answer() for launching the REDIRECTING
+ interception macro when a call is forwarded. Reduced the lock
+ time of the 'o->chan' and 'in' channels.
+
+ * main/channel.c: Protect channel access in CONNECTED_LINE and
+ REDIRECTING interception macro launch code.
+
+2010-09-21 19:48 +0000 [r288007] Brett Bryant <bbryant at digium.com>
+
+ * main/channel.c, /: Merged revisions 288006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400
+ (Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010)
+ | 8 lines Add a check to fix a rare segmentation fault you'd get
+ if ast_frdup couldn't allocate memory on the first frame being
+ queued in ast_queue_frame. (closes issue #17882) Reported by:
+ seanbright Tested by: seanbright ........ ................
+
+2010-09-21 19:08 +0000 [r287935] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 287934 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287934 | tilghman | 2010-09-21 14:07:53 -0500
+ (Tue, 21 Sep 2010) | 9 lines Merged revisions 287933 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21
+ Sep 2010) | 2 lines Less than zero is an error, not any non-zero
+ value. ........ ................
+
+2010-09-21 19:02 +0000 [r287931] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c: Revert change in favor of a more targeted fix
+
+2010-09-21 18:32 +0000 [r287929] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Send a "415 Unsupported Media Type" after
+ failure to process sdp due to unknown Content-Encoding header.
+ ABE-2258
+
+2010-09-21 15:53 +0000 [r287897] Richard Mudgett <rmudgett at digium.com>
+
+ * main/features.c: Cut-n-paste error in builtin_blindtransfer().
+
+2010-09-21 15:43 +0000 [r287895] Russell Bryant <russell at digium.com>
+
+ * res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
+ main/acl.c: Don't use ast_strdupa() from within the arguments to
+ a function. (closes issue #17902) Reported by: afried Patches:
+ issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+ russell Review: https://reviewboard.asterisk.org/r/927/
+
+2010-09-21 15:24 +0000 [r287893] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Anonymous callerid needs a "sip:" uri
+ prefix. (closes issue #17981) Reported by: avalentin Patches:
+ sip-anonymous-aastra.patch uploaded by avalentin (license 1107)
+ (plus an additional fix by me) Tested by: avalentin
+
+2010-09-21 13:41 +0000 [r287863] Russell Bryant <russell at digium.com>
+
+ * main/logger.c: Fix a regression in verbose logger processing.
+
+2010-09-21 04:37 +0000 [r287833] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c: Don't generate connected line buffer twice for
+ comparison
+
+2010-09-21 00:00 +0000 [r287760] Brett Bryant <bbryant at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 287759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400
+ (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010)
+ | 16 lines Fix misvalidation of meetme pins in conjunction with
+ the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a
+ user and admin pin setup for your conference, using the user pin
+ would gain you admin priviledges. Also, when no user pin was set,
+ an admin pin was, the 'a' MeetMe flag wasn't used, and the user
+ tried to enter a conference then they were still prompted for a
+ pin and forced to hit #. (closes issue #17908) Reported by: kuj
+ Patches: pins_2.patch uploaded by kuj (license 1111) Tested by:
+ kuj Review: [full review board URL with trailing slash] ........
+ ................
+
+2010-09-20 23:51 +0000 [r287757] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c: Avoid infinite loop with certain local channel
+ connected line updates Compare connected line data before sending
+ a connected line indication to avoid possible loops. Review:
+ https://reviewboard.asterisk.org/r/932/
+
+2010-09-20 23:20 +0000 [r287701] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/channel.c, /: Merged revisions 287685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep
+ 2010) | 18 lines ast_channel_masquerade: Avoid recursive
+ masquerades. Check all 4 combinations of (original/clonechan) *
+ (masq/masqr). Initially original->masq and clonechan->masqr were
+ only checked. It's possible with multiple masq's planned - and
+ not yet executed, that the 'original' chan could already have
+ another masq'd into it - thus original->masqr would be set, that
+ masqr would lost. Likewise for the clonechan->masq. (closes issue
+ #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
+ based on bug16057.diff4.txt uploaded by alecdavis (license 585)
+ Tested by: ramonpeek, davidw, alecdavis ........
+
+2010-09-20 23:14 +0000 [r287683] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: The inalarm flag was not set in sig_analog
+ struct if the port is initially in alarm. Fixed initial inalarm
+ value for sig_analog ports. Along with -r261007, this gets the
+ inalarm flag in sync with chan_dahdi for sig_analog ports.
+ (closes issue #16983)
+
+2010-09-20 22:21 +0000 [r287661] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/channel.c: ast_do_masquerade. Keep channels ao2_container
+ locked while unlink and linking channels. Previously, Masquerade
+ would unlock 'original' and 'clonechan' and allow another masq
+ thread to run. End result would be corrupted memory, and the
+ frequent report 'Bad Magic Number'. (closes issue #17801,#17710)
+ Reported by: notthematrix Patches: Based on bug17801.diff1.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/928
+
+2010-09-20 22:09 +0000 [r287645-287647] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/channel.h, CHANGES, include/asterisk/framehook.h
+ (added), main/channel.c, main/framehook.c (added),
+ funcs/func_frame_trace.c (added): Addition of the FrameHook API
+ (AKA AwesomeHooks) So far all our tools for viewing and
+ manipulating media streams within Asterisk have been entirely
+ focused on audio. That made sense then, but is not scalable now.
+ The FrameHook API lets us tap into and manipulate _ANY_ type of
+ media or signaling passed on a channel present today or in the
+ future. This tool is a step in the direction of expanding
+ Asterisk's boundaries and will help generate some rather
+ interesting applications in the future. In addition to the
+ FrameHook API, a simple dialplan function exercising the api has
+ been included as well. This function is called FRAME_TRACE().
+ FRAME_TRACE() allows for the internal ast_frames read and written
+ to a channel to be output. Filters can be placed on this function
+ to debug only certain types of frames. This function could be
+ thought of as an internal way of doing ast_frame packet captures.
+ Review: https://reviewboard.asterisk.org/r/925/
+
+ * channels/chan_sip.c: Fixes issue with registrations not working
+ properly with pedantic=yes. (closes issue #18017) Reported by:
+ schmidts Patches: issues_18017_v1.diff uploaded by dvossel
+ (license 671) Tested by: schmidts
+
+2010-09-20 21:29 +0000 [r287643] Jason Parker <jparker at digium.com>
+
+ * /, channels/chan_skinny.c: Merged revisions 287642 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep
+ 2010) | 8 lines Don't crash when parking a non-bridged call.
+ (closes issue #17680) Reported by: jmhunter Patches:
+ chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by:
+ jmhunter, DEA ........
+
+2010-09-20 21:19 +0000 [r287639] Brett Bryant <bbryant at digium.com>
+
+ * main/logger.c: Fixes an error with the logger that caused verbose
+ messages to be spammed to the screen if syslog was configured in
+ logger.conf (closes issue #17974) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/915/
+
+2010-09-20 15:57 +0000 [r287559] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, /: Merged revisions 287558 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287558 | mnicholson | 2010-09-20 10:56:21 -0500
+ (Mon, 20 Sep 2010) | 14 lines Use ast_str when processing hint
+ state changes Merged revisions 287555 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
+ 2010) | 5 lines Use ast_dynamic_str when processing hint state
+ changes (related to issue #17928) Reported by: mdu113 ........
+ ................
+
+2010-09-19 16:09 +0000 [r287471] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c, /: Merged revisions 287470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön,
+ 19 Sep 2010) | 14 lines Merged revisions 287469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
+ lines Make sure we always free variables properly in manager
+ originate. (closes issue #17891) reported, solved and tested by
+ oej Review: https://reviewboard.asterisk.org/r/869/ ........
+ ................
+
+2010-09-17 21:08 +0000 [r287388] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 287387 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287387 | tilghman | 2010-09-17 16:08:00 -0500
+ (Fri, 17 Sep 2010) | 14 lines Merged revisions 287386 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
+ | 7 lines Blank columns should get set on reload, not ignored.
+ (closes issue #16893) Reported by: haakon Patches:
+ 20100818__issue16893.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2010-09-17 13:37 +0000 [r287309] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, /: Merged revisions 287308 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287308 | mnicholson | 2010-09-17 08:36:07 -0500
+ (Fri, 17 Sep 2010) | 12 lines Merged revisions 287307 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
+ 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while
+ processing in ast_hint_state_changed(). (related to issue #17928)
+ Reported by: mdu113 ........ ................
+
+2010-09-17 08:44 +0000 [r287269-287271] Jan Kalab <pitlicek at gmail.com>
+
+ * res/res_calendar_ews.c: Events are visible after they were
+ removed from EWS calendar Because we must merge calendar even
+ when it's empty. (closes issue #17786)
+
+ * res/res_calendar_ews.c: Asterisk crashing because of double free
+ when EWS request fails The free is done later in code. I think
+ ast_free() should have built in checks for double free. (closes
+ issue #17782)
+
+ * res/res_calendar_caldav.c, res/res_calendar_ews.c,
+ res/res_calendar_exchange.c, res/res_calendar_icalendar.c:
+ Support for HTTP redirects in calendar's URL libneon does not
+ support HTTP redirects (3xx responses) by default. You must tell
+ it to follow them. Also, another little unsigned int fix. (closes
+ issue #17776) Review: https://reviewboard.asterisk.org/r/921/
+
+2010-09-16 22:04 +0000 [r287195] Jason Parker <jparker at digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Don't fail when running the
+ Debian init script directly (as one would normally do). readlink
+ apparently returns 1 when the arg isn't a symlink, which caused
+ the script to exit. (closes issue #17910) Reported by: wurstsalat
+
+2010-09-16 21:57 +0000 [r287193] Russell Bryant <russell at digium.com>
+
+ * UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Set
+ the default for "autofill" and "shared_lastcall" to "yes" in
+ queues.conf. Review: https://reviewboard.asterisk.org/r/922/
+
+2010-09-16 20:07 +0000 [r287116-287120] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, /: Merged revisions 287119 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287119 | mnicholson | 2010-09-16 15:06:16 -0500
+ (Thu, 16 Sep 2010) | 15 lines Merged revisions 287118 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
+ 2010) | 8 lines Don't limit hint processing in
+ ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
+ (closes issue #17928) Reported by: mdu113 Patches:
+ 20100831__issue17928.diff.txt uploaded by tilghman (license 14)
+ Tested by: mdu113 ........ ................
+
+ * main/cdr.c, /: Merged revisions 287115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287115 | mnicholson | 2010-09-16 14:53:41 -0500
+ (Thu, 16 Sep 2010) | 15 lines Merged revisions 287114 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
+ 2010) | 8 lines Don't stop printing cdr variables if we encounter
+ one with a blank name or value. (closes issue #17900) Reported
+ by: under Patches: core-show-channel-cdr-fix1.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson ........
+ ................
+
+2010-09-15 22:17 +0000 [r287056] Terry Wilson <twilson at digium.com>
+
+ * res/res_srtp.c: Don't hang up a call on an SRTP unprotect failure
+ Also make it more obvious when there is an issue en/decrypting.
+ (closes issue #17563) Reported by: Alexcr Patches:
+ res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by:
+ twilson
+
+2010-09-15 20:58 +0000 [r287020] Jeff Peeler <jpeeler at digium.com>
+
+ * main/features.c: fix uninintialized variable
+
+2010-09-15 20:53 +0000 [r287017] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_msg_parser.c, channels/chan_misdn.c: Merged
+ revision 287014 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed,
+ 15 Sep 2010) | 58 lines The handling of call transfer signaling
+ for mISDN PTMP is not fully implemented. The handling of call
+ transfer signaling for mISDN PTMP is not fully implemented. The
+ signaling of number updates with ISDN/DSS1 ECT supplementary
+ services (ETS 300 369-1) comes along with a notification
+ indicator IE and redirection number IE for PTMP. The
+ implementation in the current Asterisk mISDN channel
+ unfortunately can handle these information elements only in a
+ NOTIFY message. These information elements are also signaled in a
+ FACILTY message with a RequestSubaddress facility, when the
+ subscriber is already in the active state (see 9.2.4 and 9.2.5 of
+ ETS 300 369-1). ********** abe_2526_ast.patch * Added support to
+ handle the notification indicator IE and redirection number IE
+ with the RequestSubaddress facility. * Made
+ misdn_update_connected_line() send a NOTIFY message if Asterisk
+ originated the call and it is not connected yet. * Made
+ misdn_update_connected_line() send a FACILITY message if the call
+ is already connected. This patch requires the presence of the
+ associated mISDN patches to compile. I had to enhance mISDN to
+ allow the notification indicator IE and the redirection number IE
+ to be used with a FACILITY message. Earlier versions of the
+ Digium enhanced mISDN are no longer going to work. **********
+ abe_2526_misdn.patch * Made an incoming FACILITY message allow
+ the presence of the notification indicator IE and the redirection
+ number IE. ********** abe_2526_misdnuser_v3.patch * Added support
+ to send and receive a FACILITY message with the notification
+ indicator IE and the redirection number IE. * Added the ability
+ to send a NOTIFY message in PTMP/NT mode to all responding
+ subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches:
+ abe_2526_ast.patch uploaded by rmudgett (license 664)
+ abe_2526_misdn.patch uploaded by rmudgett (license 664)
+ abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
+ Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526
+ ..........
+
+2010-09-15 20:32 +0000 [r286931-287015] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 286998 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500
+ (Wed, 15 Sep 2010) | 14 lines Merged revisions 286941 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010)
+ | 7 lines Ensure mailbox is not filled to capacity before doing
+ message forwarding. Specifically, before prompting to record a
+ prepended message the capacity is checked first. If the mailbox
+ is full the extension will be reprompted. ABE-2517 ........
+ ................
+
+ * CHANGES, channels/chan_iax2.c, channels/sip/include/sip.h,
+ configs/features.conf.sample, channels/chan_mgcp.c,
+ include/asterisk/features.h, channels/chan_dahdi.c,
+ channels/sig_analog.c, channels/chan_sip.c, main/features.c: Add
+ parking extension for non-default parking lots. This is a new
+ feature that allows for parking to custom parking lots to be
+ accessed directly, rather than with channel variables or by
+ changing the default parking lot. The extension is set with the
+ parkext option just as the default parking lot is done. Also, the
+ manager action has been updated to optionally allow a specified
+ parking lot. (closes issue #14882) Reported by: vmikhnevych
+ Patches: patch_14882.txt uploaded by mnick (license 874) modified
+ by me Review: https://reviewboard.asterisk.org/r/884/
+
+2010-09-15 18:29 +0000 [r286904-286905] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_analog.c: Simplify some code in sig_analog.
+
+ * channels/sig_analog.c: Unable to originate calls using E&M over
+ T1. When originating a call from Unit Under Test to Reference
+ Unit using E&M RBS signaling mode, I get the following warning
+ message: "Ring/Off-hook in strange state 3 on channel 1". Fixed
+ the sig_analog outgoing flag. It was never set when sig_analog
+ was extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408
+
+2010-09-15 13:05 +0000 [r286868] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Set tohost to the domain specified in the
+ configuration file instead of the IP address of the host we are
+ calling. This fixes a regression introduced in r274783. (closes
+ issue #17960) Reported by: adriavidal Patches:
+ sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested
+ by: mich, mnicholson, adriavidal (closes issue #17676) Reported
+ by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson
+
+2010-09-14 21:57 +0000 [r286834] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Sets subscribed type for outgoing MWI
+ subscriptions so correct Event header is used.
+
+2010-09-14 19:28 +0000 [r286682-286758] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286757 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500
+ (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
+ 2010) | 13 lines Don't clear the username from a realtime
+ database when a registration expires. Non-realtime chan_sip does
+ not clear the username from memory when a registration expiries
+ so realtime probably shouldn't either. (closes issue #17551)
+ Reported by: ricardolandim Patches:
+ reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
+ 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
+ (license 96) reg-expiry-username-1.8-fix1.diff uploaded by
+ mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: ricardolandim,
+ mnicholson ........ ................
+
+ * main/channel.c, /: Merged revisions 286681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500
+ (Tue, 14 Sep 2010) | 14 lines Merged revisions 286679 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
+ 2010) | 7 lines Only drop duplicate answer frames if the channel
+ is bridged. Back in r3710 ast_read() was modified to drop answer
+ frames on channels that were in the UP state. This modification
+ prevented bridges that were up before the answer from being
+ broken and reestablished by an ANSWER control frame. That change
+ also prevents pickup of channels called from the ast_dial
+ framework from working properly. The ast_dial framework expects
+ to see an ANSWER frame after dialing and the pickup code queues
+ one but ast_read() drops it. This new change only drops ANSWER
+ frames when the channel is bridged, allowing the answer queued by
+ the pickup code to properly pass through ast_read() on to the
+ ast_dial framework. ABE-2473 (related to issue #2342) ........
+ ................
+
+2010-09-14 15:30 +0000 [r286647] Richard Mudgett <rmudgett at digium.com>
+
+ * doc/tex/channelvariables.tex, doc/tex/partymanip.tex: Corrected
+ documented CONNECTED_LINE and REDIRECTING party manipulation
+ macro names.
+
+2010-09-14 06:55 +0000 [r286617] Jan Kalab <pitlicek at gmail.com>
+
+ * res/res_calendar_ews.c: Merging events for Exchange web service
+ doesn't work as expected, resulting in only one event in calendar
+ The solution is to use "global" counter of events, since we do
+ new requests for every event and calendar sync after every
+ request. So now we do sync only after last request. (closes issue
+ #17877) Review: https://reviewboard.asterisk.org/r/916/
+
+2010-09-14 05:07 +0000 [r286528-286588] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/realtime/mysql/voicemail_data.sql (added), /,
+ contrib/realtime/mysql/voicemail_messages.sql (added): Merged
+ revisions 286587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010)
+ | 2 lines Add documentation on missing backend tables for
+ Voicemail ........
+
+ * /, main/features.c: Merged revisions 286557 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010)
+ | 2 lines C precedence got me ........
+
+ * /, main/features.c: Merged revisions 286527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010)
+ | 2 lines Refactor conversion to ast_poll() to fix callparking
+ regression. ........
+
+2010-09-13 19:40 +0000 [r286457] Jason Parker <jparker at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 286456 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) |
+ 5 lines Remove "Internal IP" from sip show settings, as it's not
+ at all useful to display. (closes issue #17840) Reported by: oej
+ ........
+
+2010-09-13 15:52 +0000 [r286426] Richard Mudgett <rmudgett at digium.com>
+
+ * configs/chan_dahdi.conf.sample: Update chan_dahdi.conf.sample to
+ reflect new libpri T309 default value.
+
+2010-09-11 17:09 +0000 [r286270] Olle Johansson <oej at edvina.net>
+
+ * /, main/file.c: Merged revisions 286268 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör,
+ 11 Sep 2010) | 11 lines Merged revisions 286267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
+ lines Handle error response when we can't make file compatible
+ Review: https://reviewboard.asterisk.org/r/911/ ........
+ ................
+
+2010-09-10 22:04 +0000 [r286189] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/channel.h, include/asterisk/pbx.h,
+ include/asterisk/frame.h, channels/chan_local.c,
+ funcs/func_channel.c: Merged revisions 286115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286115 | twilson | 2010-09-10 15:35:25 -0500
+ (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010)
+ | 16 lines Inherit CHANNEL() writes to both sides of a Local
+ channel Having Local (/n) channels as queue members and setting
+ the language in the extension with Set(CHANNEL(language)=fr) sets
+ the language on the Local/...,2 channel. Hold time report
+ playbacks happen on the Local/...,1 channel and therefor do not
+ play in the specified language. This patch modifies
+ func_channel_write to call the setoption callback and pass the
+ CHANNEL() write info to the callback. chan_local uses this
+ information to look up the other side of the channel and apply
+ the same changes to it. (closes issue #17673) Reported by:
+ Guggemand Review: https://reviewboard.asterisk.org/r/903/
+ ........ ................
+
+2010-09-10 21:11 +0000 [r286120] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 286117 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286117 | pabelanger | 2010-09-10 16:55:06 -0400
+ (Fri, 10 Sep 2010) | 11 lines Merged revisions 286114 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep
+ 2010) | 4 lines Load iax.conf before registering any
+ functions/applications/actions. Review:
+ https://reviewboard.asterisk.org/r/914/ ........ ................
+
+2010-09-10 20:55 +0000 [r286118] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, /: Merged revisions 286116 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286116 | rmudgett | 2010-09-10 15:42:44 -0500
+ (Fri, 10 Sep 2010) | 18 lines Merged revisions 286113 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010)
+ | 11 lines An outgoing call may not get hung up if a pre-connect
+ incoming ISDN call is disconnected. If the ISDN link a
+ pre-connect incoming call is using fails or is reset, the
+ outgoing leg may not hang up or be delayed in hanging up.
+ (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
+ PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
+ PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
+ incoming call leg hangs up before connecting for any reason. It
+ makes no sense to send a BUSY or CONGESTION control frame to the
+ outgoing call leg under these circumstances. ........
+ ................
+
+2010-09-10 20:31 +0000 [r286112] Russell Bryant <russell at digium.com>
+
+ * main/db.c: Rate limit calls to fsync() to 1 per second after
+ astdb updates. Astdb was determined to be one of the most
+ significant bottlenecks in SIP registration processing. This
+ patch improved the speed of an astdb load test by 50000% (yes,
+ Fifty-Thousand Percent). On this particular load test setup, this
+ doubled the number of SIP registrations the server could handle.
+ Review: https://reviewboard.asterisk.org/r/825/
+
+2010-09-10 18:31 +0000 [r286025] Tilghman Lesher <tlesher at digium.com>
+
+ * /: Merged revisions 286024 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r286024 | tilghman | 2010-09-10 13:30:21 -0500
+ (Fri, 10 Sep 2010) | 9 lines Merged revisions 286023 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10
+ Sep 2010) | 2 lines Missing newline ........ ................
+
+2010-09-10 13:13 +0000 [r285992] David Ruggles <thedavidfactor at gmail.com>
+
+ * doc/externalivr.txt, CHANGES: Added missing documentation for
+ ExternalIVR feature added in January 2010
+
+2010-09-10 05:32 +0000 [r285931-285962] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/select.h, /: Merged revisions 285961 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010)
+ | 6 lines Another fix for Mac OS X. While trying to fix this the
+ "right" way, I wandered into dependency hell. Two hours later, I
+ backed out, and just removed the offending code. ast_inline_api
[... 23842 lines stripped ...]
More information about the asterisk-commits
mailing list