[asterisk-commits] lmadsen: tag 1.6.2.14-rc1 r287635 - /tags/1.6.2.14-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Sep 20 13:34:00 CDT 2010
Author: lmadsen
Date: Mon Sep 20 13:33:56 2010
New Revision: 287635
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=287635
Log:
Importing files for 1.6.2.14-rc1 release.
Added:
tags/1.6.2.14-rc1/.lastclean (with props)
tags/1.6.2.14-rc1/.version (with props)
tags/1.6.2.14-rc1/ChangeLog (with props)
Added: tags/1.6.2.14-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.14-rc1/.lastclean?view=auto&rev=287635
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--- tags/1.6.2.14-rc1/ChangeLog (added)
+++ tags/1.6.2.14-rc1/ChangeLog Mon Sep 20 13:33:56 2010
@@ -1,0 +1,27391 @@
+2010-09-20 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.14-rc1 Released.
+
+2010-09-20 15:56 +0000 [r287556-287558] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, /: 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 ........
+
+ * /: Revert r287556.
+
+ * /: 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 16:06 +0000 [r287470] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c, /: 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 [r287387] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c, /: 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 13:36 +0000 [r287308] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, /: 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 ........
+
+2010-09-16 22:12 +0000 [r287198] Jason Parker <jparker at digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 287197 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287197 | qwell | 2010-09-16 17:12:30 -0500 (Thu, 16 Sep 2010) |
+ 7 lines Add LSB headers for Debian init script, since Debian will
+ complain if it isn't there. Headers were taken from trunk.
+ (closes issue #17958) Reported by: javyer ........
+
+2010-09-16 20:06 +0000 [r287115-287119] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, /: 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 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 20:28 +0000 [r286998] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: 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
+ ........
+
+2010-09-14 19:27 +0000 [r286681-286757] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: 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 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 05:06 +0000 [r286527-286587] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/realtime/mysql/voicemail_messages.sql (added),
+ contrib/realtime/mysql/voicemail_data.sql (added): Add
+ documentation on missing backend tables for Voicemail
+
+ * main/features.c: C precedence got me
+
+ * main/features.c: Refactor conversion to ast_poll() to fix
+ callparking regression.
+
+2010-09-13 19:38 +0000 [r286456] Jason Parker <jparker at digium.com>
+
+ * channels/chan_sip.c: Remove "Internal IP" from sip show settings,
+ as it's not at all useful to display. (closes issue #17840)
+ Reported by: oej
+
+2010-09-11 17:05 +0000 [r286268] Olle Johansson <oej at edvina.net>
+
+ * /, main/file.c: 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 22:56 +0000 [r286223] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 286222 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10
+ Sep 2010) | 1 line Return -1 if chan_local doesn't support an
+ option ........
+
+2010-09-10 20:55 +0000 [r286117] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_iax2.c, /: 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 20:42 +0000 [r286116] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: 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 20:35 +0000 [r286115] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/pbx.h, include/asterisk/frame.h,
+ channels/chan_local.c, /, funcs/func_channel.c,
+ include/asterisk/channel.h: 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 18:30 +0000 [r285930-286024] Tilghman Lesher <tlesher at digium.com>
+
+ * tests/test_heap.c, /, main/test.c: 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 ........
+
+ * include/asterisk/select.h: 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 only goes one level deep and then it breaks. Ouch.
+
+ * tests/test_poll.c, include/asterisk/select.h, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 285889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010)
+ | 7 lines Fix Mac OS X build. This also fixes a rather grievous
+ calculation error for the offset of ast_fdset, which was masked
+ on Linux and FreeBSD, because these platforms check the first 256
+ FDs regardless of the bitmask setting (due to backwards
+ compatibility). ........
+
+2010-09-09 22:49 +0000 [r285818] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * /, codecs/gsm/Makefile: Merged revisions 285817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep
+ 2010) | 8 lines GCC 4.2.x optimizations result in improper
+ behavior of GSM codec (closes issue #17688) Reported by:
+ pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by
+ pprindeville (license 347) Tested by: mkeuter, pprindeville
+ ........
+
+2010-09-09 20:09 +0000 [r285744] Jason Parker <jparker at digium.com>
+
+ * main/channel.c, /: Merged revisions 285742 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) |
+ 9 lines Transmit silence when reading DTMF in ast_readstring.
+ Otherwise, you could get issues with DTMF timeouts causing
+ hangups. (closes issue #17370) Reported by: makoto Patches:
+ channel-readstring-silence-generator.patch uploaded by makoto
+ (license 38) ........
+
+2010-09-09 18:50 +0000 [r285639-285710] Brett Bryant <bbryant at digium.com>
+
+ * main/pbx.c: Fixes an issue with dialplan pattern matching where
+ the specificity for pattern ranges and pattern special characters
+ was inconsistent. (closes issue #16903) Reported by: Nick_Lewis
+ Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license
+ 657) Tested by: Nick_Lewis
+
+ * res/res_musiconhold.c, /: Merged revisions 285638 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09
+ Sep 2010) | 7 lines Fixes an issue with MOH where it doesn't
+ recover cleanly when it can't play a file and would just stop,
+ instead of continuing to find the next playable file in the MOH
+ class. (closes issue #17807) Reported by: kshumard Review:
+ https://reviewboard.asterisk.org/r/910/ ........
+
+2010-09-08 22:11 +0000 [r285563-285567] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 285566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010)
+ | 2 lines In retrans_pkt, do not unlock pvt until the end of the
+ function on a transmit failure. ........
+
+ * channels/chan_sip.c: Fixes interoperability problems with session
+ timer behavior in Asterisk. CHANGES: 1. Never put "timer" in
+ "Require" header. This is not to our benefit and RFC 4028 section
+ 7.1 even warns against it. It is possible for one endpoint to
+ perform session-timer refreshes while the other endpoint does not
+ support them. If in this case the end point performing the
+ refreshing puts "timer" in the Require field during a refresh,
+ the dialog will likely get terminated by the other end. 2. Change
+ the behavior of 'session-timer=accept' in sip.conf (which is the
+ default behavior of Asterisk with no session timer configuration
+ specified) to only run session-timers as result of an incoming
+ INVITE request if the INVITE contains an "Session-Expires"
+ header... Asterisk is currently treating having the "timer"
+ option in the "Supported" header as a request for session timers
+ by the UAC. I do not agree with this. Session timers should only
+ be negotiated in "accept" mode when the incoming INVITE supplies
+ a "Session-Expires" header, otherwise RFC 4028 says we should
+ treat a request containing no "Session-Expires" header as a
+ session with no expiration. Below I have outlined some situations
+ and what Asterisk's behavior is. The table reflects the behavior
+ changes implemented by this patch. SITUATIONS: -Asterisk as UAS
+ 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS
+ "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO
+ "Session-Expires". 200 Ok Response HAS "Session-Expires" header
+ 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO
+ "Session-Expires" header 5. Outgoing INVITE: HAS
+ "Session-Expires". Active - Asterisk will have an active refresh
+ timer regardless if the other endpoint does. Inactive - Asterisk
+ does not have an active refresh timer regardless if the other
+ endpoint does. XXXXXXX - Not possible for mode.
+ ______________________________________ |SITUATIONS |
+ 'session-timer' MODES | |___________|________________________| |
+ | originate | accept | |-----------|------------|-----------| |1.
+ | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX |
+ Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX |
+ -------------------------------------- (closes issue #17005)
+ Reported by: alexrecarey
+
+2010-09-08 20:56 +0000 [r285532] Brett Bryant <bbryant at digium.com>
+
+ * apps/app_meetme.c: Fixes a bug with MeetMe where after announcing
+ the amount of time left in a conference, if music on hold was
+ playing, it doesn't restart. (closes issue #17408) Reported by:
+ sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by
+ sysreq (license 1009) Tested by: sysreq
+
+2010-09-08 20:42 +0000 [r285526-285529] Jason Parker <jparker at digium.com>
+
+ * res/res_musiconhold.c, include/asterisk/astobj2.h: Follow coding
+ guidelines in moh rescan fix. Also fix the documentation that got
+ me in trouble.
+
+ * res/res_musiconhold.c: Fixes issue where moh files were no longer
+ rescanned during a reload. (closes issue #16744) Reported by: pj
+ Patches: 16744-reload.diff uploaded by qwell (license 4) Tested
+ by: qwell
+
+2010-09-07 20:31 +0000 [r285267-285366] Tilghman Lesher <tlesher at digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 285365 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010)
+ | 9 lines Catch invalid extensions at the parser, instead of
+ making the core deal with them. (closes issue #17794) Reported
+ by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded
+ by tilghman (license 14) 20100820__issue17794__1.4.diff.txt
+ uploaded by tilghman (license 14) Tested by: PavelL ........
+
+ * main/poll.c, /: Merged revisions 285266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010)
+ | 4 lines Use poll, if indicated to do so, in the ast_poll2
+ implementation. This fixes the unit tests on FreeBSD 8.0.
+ ........
+
+2010-09-07 17:49 +0000 [r285196] Brett Bryant <bbryant at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 285194 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07
+ Sep 2010) | 10 lines Fixes voicemail.conf issues where mailboxes
+ with passwords that don't precede a comma would throw unnecessary
+ error messages. (closes issue #15726) Reported by: 298 Patches:
+ M15726.diff uploaded by junky (license 177) Tested by: junky
+ Review: [full review board URL with trailing slash] ........
+
+2010-09-06 06:55 +0000 [r285089] Tilghman Lesher <tlesher at digium.com>
+
+ * makeopts.in, /, BSDmakefile (added): Merged revisions 285088 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010)
+ | 2 lines Silly convenience script for BSD platforms. ........
+
+2010-09-03 18:15 +0000 [r284958] Brett Bryant <bbryant at digium.com>
+
+ * channels/chan_iax2.c: This is a patch provided for issue #17935
+ to add the ActionID to the IAXregistry AMI response. (closes
+ issue #17935) Reported by: alexkuklin Patches: iaxshowreg
+ uploaded by alexkuklin (license 1115) Tested by: alexkuklin
+
+2010-09-03 16:20 +0000 [r284897] Terry Wilson <twilson at digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 284881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010)
+ | 5 lines Properly detect when a sound file doesn't exist
+ ast_fileexists returns -1 for error and 0 for a non-existant
+ file. The existing code treated missing files as though they
+ existed. ........
+
+2010-09-02 20:54 +0000 [r284778] Brett Bryant <bbryant at digium.com>
+
+ * main/manager.c, /: Merged revisions 284777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010)
+ | 7 lines Fixes a bug in manager.c where the default
+ configuration values weren't reset when the manager configuration
+ was reloaded. (closes issue #17917) Reported by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/883/ ........
+
+2010-09-02 16:48 +0000 [r284704] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284703 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010)
+ | 7 lines Removed relatedpeer code from sip_autodestruct Handling
+ of the relatedpeer structure associated with a sip_pvt should be
+ done during the final sip_destruction function, not in
+ sip_autodestruct. ........
+
+2010-09-02 16:07 +0000 [r284399-284665] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_usbradio.c: Fixing build.
+
+ * apps/app_queue.c: Don't reset queue stats on a module reload.
+ (closes issue #17535) Reported by: raarts Patches:
+ 20100819__issue17535.diff.txt uploaded by tilghman (license 14)
+
+ * configure, include/asterisk/autoconfig.h.in: Failed to rerun
+ bootstrap.sh after last commit
+
+ * res/res_jabber.c, main/rtp.c, main/poll.c,
+ include/asterisk/select.h (added), channels/chan_usbradio.c,
+ channels/chan_phone.c, channels/chan_misdn.c, main/features.c,
+ include/asterisk/poll-compat.h, tests/test_poll.c (added),
+ main/asterisk.c, utils/clicompat.c, res/res_ais.c, /,
+ configure.ac, channels/console_video.c,
+ include/asterisk/channel.h: Merged revisions 284478 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01
+ Sep 2010) | 11 lines Ensure that all areas that previously used
+ select(2) now use poll(2), with implementations that need poll(2)
+ implemented with select(2) safe against 1024-bit overflows. This
+ is a followup to the fix for the pthread timer in 1.6.2 and
+ beyond, fixing a potential crash bug in all supported releases.
+ (closes issue #17678) Reported by: russell Branch:
+ https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
+ Review: https://reviewboard.asterisk.org/r/824/ ........
+
+ * res/res_config_pgsql.c: Don't warn on floats and timestamps
+ (closes issue #17082) Reported by: coolmig
+
+ * /, channels/chan_sip.c: Merged revisions 284393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
+ | 7 lines Don't send a devstate change on poke_noanswer if the
+ state did not change. (closes issue #17741) Reported by: schmidts
+ Patches: chan_sip.c.patch uploaded by schmidts (license 1077)
+ ........
+
+2010-08-31 18:59 +0000 [r284317] Leif Madsen <lmadsen at digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 284316 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31
+ Aug 2010) | 7 lines Update say.conf.sample to match the rules in
+ say.c (closes issue #17835) Reported by: RoadKill Patches:
+ say.conf.sample.patch.rules uploaded by RoadKill (license 933)
+ Tested by: RoadKill ........
+
+2010-08-30 22:27 +0000 [r284280] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_festival.c: Fix 3 coding errors: 1) After we close FD,
+ we should not be trying to write to it. 2) Call _exit(0), not
+ exit(0), to avoid running shutdown routines in a child. 3) Use
+ endian, not processor, detection to ensure bytes are written in
+ the correct order. (closes issue #15706) Reported by: modelnine
+ Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by
+ modelnine (license 865) Tested by: gmartinez
+
+2010-08-27 22:27 +0000 [r284002] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
+ | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests.
+ (closes issue #17758) Reported by: ibc Patches:
+ multiple_accept_headers_1.4.diff uploaded by dvossel (license
+ 671) ........
+
+2010-08-27 20:30 +0000 [r283881] Jason Parker <jparker at digium.com>
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
+ revisions 283880 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
+ 8 lines Fix issue with decoding ^-escaped characters in realtime.
+ (closes issue #17790) Reported by: denzs Patches:
+ 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell,
+ denzs ........
+
+2010-08-26 15:24 +0000 [r283381-283691] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283690 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
+ | 19 lines Fixed how Asterisk destroys a dialog on channel hangup
+ before invite receives a response. If an ast_channel with a SIP
+ tech pvt hangs up before the sip dialog gets a response to its
+ outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is
+ not rfc compliant and results in confusion at the other endpoint.
+ sip_pretend_ack will ack and remove all the packets in the
+ retransmit queue. This means that the INVITE will stop
+ retransmitting, and that any response to that INVITE that comes
+ after the pretend_ack occurs will be ignored. Instead of faking
+ any sort of acknowledgement for an outgoing INVITE during an
+ internal hangup, we should let the protocol stack process the
+ INVITE transaction and terminate the dialog properly. This is
+ achieved by setting the PENDING_BYE flag. When this flag is used,
+ once the dialog proceeds to an escapable state the transaction
+ will either be canceled with a SIP_CANCEL or completed followed
+ immediately by a BYE. Attempting to do this any other way is
+ incorrect. If the endpoint is not responding to the INVITE
+ request, the INVITE must continue to be retransmitted until it
+ times out which will result in the dialog being destroyed.
+ ........
+
+ * channels/chan_sip.c: Add to and from tags to NOTIFY dialog-info
+ xml body so pickup can occur. When pedantic mode is used, the
+ dialog-info xml generated during a ringing event must contain the
+ to and from tag values. Otherwise if a pickup occurs using INVITE
+ with replaces, Astrisk will not be able to locate the
+ subscription.
+
+ * channels/chan_sip.c: Asterisk will not advertise session timers
+ are supported when 'session-timers=refuse' is used. Asterisk now
+ dynamically builds the "Supported" header depending on what is
+ enabled/disabled in sip.conf. Session timers used to always be
+ advertised as being supported even when they were disabled in the
+ configuration. This caused problems with some end points. (issue
+ #17005)
+
+ * /, channels/chan_sip.c: Merged revisions 283380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010)
+ | 11 lines This fix makes sure the ast_channel hangs up correctly
+ when the dialog's PENDING_BYE flag is set. When the pending bye
+ flag is used, it is possible that the dialog will terminate and
+ leave the sip_pvt->owner channel up. This is because we never
+ hangup the ast_channel after sending the SIP_BYE request. When we
+ receive the response for the SIP_BYE we set need_destroy which we
+ would expect to destroy the dialog on the next do_monitor loop,
+ but this is not the case. The dialog will only be destroyed once
+ the owner is hungup even with the need_destroy flag set. This
+ patch sets the softhangup flag on the ast_channel when a SIP_BYE
+ request is sent as a result of the pending bye flag. ........
+
+2010-08-23 21:32 +0000 [r283318] Tilghman Lesher <tlesher at digium.com>
+
+ * cdr/cdr_odbc.c, cdr/cdr_adaptive_odbc.c: CDR drivers depend upon
+ res_odbc, not directly on the ODBC libraries
+
+2010-08-23 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.12-rc1 Released.
+
+2010-08-20 16:48 +0000 [r283049-283124] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 283123 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500
+ (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from
+ https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
+ | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
+ line Reference correct struct member for unlikely event
+ PRI_EVENT_CONFIG_ERR. .......... ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 283048 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20
+ Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending
+ ALERTING is a protocol error The PRI layer in chan_dadhi will
+ check if a PROGRESS message has already been sent, and not allow
+ sending another (although that is technically allowed by the Q931
+ spec), however it does not protect against sending an ALERTING
+ and then sending a PROGRESS message, which is a violation of the
+ specification. Most switches don't seem to care too deeply about
+ this, but some do, and will disconnect the call when receiving
+ this invalid sequence. Protocol specification reference:
+ T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
+ protocol control (network side) point-point (sheet 3 of 8)"
+ (closes issue #17874) Reported by: nic_bellamy Patches:
+ asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
+ nic bellamy (license 299)
+ asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299)
+ asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299) ........
+
+2010-08-19 21:05 +0000 [r282890-282894] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
+ | 11 lines tos_sip option was not being set correctly When
+ tos_sip is used, the tos of the sip socket is only set correctly
+ if the socket binding changes on a reload. If the binding stays
+ the same but the TOS changes, the new tos value would not take
+ into effect. This patch fixes that. (closes issue #17712)
+ Reported by: nickb ........
+
+ * channels/chan_sip.c: fixes sip peer memory leaks in the
+ peer_by_ip table (issue #17798)
+
+2010-08-19 19:44 +0000 [r282859] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Merged revisions 277944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
+ 2010) | 16 lines Regression with T.38 negotiation Prior to
+ 1.4.26.3 T.38 negotiation worked properly, in the case of the
+ reporter. (issue #16852) Reported by: cfc (closes issue #16705)
+ Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
+ by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
+ samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
+
+2010-08-19 02:14 +0000 [r282730] Terry Wilson <twilson at digium.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 282729 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
+ Aug 2010) | 2 lines Add some documentation about codec
+ negotiation to sip.conf ........
+
+2010-08-18 14:28 +0000 [r282668] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: fixes crash with notifycid (closes issue
+ #17868) Reported by: francesco_r Patches: issue_17868.diff
+ uploaded by dvossel (license 671) Tested by: francesco_r
+
+2010-08-18 07:43 +0000 [r282607] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_dahdi.c: Don't warn on callerid when completely
+ text, instead of numeric with localdialplan prefixes. (closes
+ issue #16770) Reported by: jamicque Patches:
+ 20100413__issue16770.diff.txt uploaded by tilghman (license 14)
+ 20100811__issue16770.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamicque
+
+2010-08-17 21:35 +0000 [r282576] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: fixes no default transport for temp peer
+ creation in chan_sip (closes issue #17829) Reported by: falves11
+ Patches: issue_17829.rev1.txt uploaded by russell (license 2)
+ issue_17829.diff uploaded by dvossel (license 671) Tested by:
+ falves11
+
+2010-08-16 18:00 +0000 [r282469] Leif Madsen <lmadsen at digium.com>
+
+ * doc/tex/asterisk.tex, doc/tex/sounds.tex (added): Add information
+ about creating sounds files using the sounds tools publically
+ available so that others can create their own sounds prompts
+ using the same tools we use to generate sounds releases. This
+ allows people creating their own prompts to sound consistent with
+ the prompts available from the open source project. SWP-595
+
+2010-08-16 17:32 +0000 [r282467] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c, /: Merged revisions 282430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
+ | 16 lines Send a SRCCHANGE indication when we masquerade
+ Masquerading a channel means that the src of the audio is
+ potentially changing, so send a SRCCHANGE so that RTP-based media
+ streams can get a new SSRC generated to reflect the change.
+ Original patch by addix (along with lots of testing--thanks!).
+ (closes issue #17007) Reported by: addix Patches:
+ 1001-reset-SSRC-original-channel.diff uploaded by addix (license
+ 1006) srcchange.diff uploaded by twilson (license 396) Tested by:
+ addix, twilson Review: https://reviewboard.asterisk.org/r/862/
+ ........
+
+2010-08-13 18:54 +0000 [r282235] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: only do magic pickup when notifycid is
+ enabled A new way of doing BLF pickup was introduced into 1.6.2.
+ This feature adds a call-id value into the XML of a SIP_NOTIFY
+ message sent to alert a subscriber that a device is ringing. This
+ option should only be enabled when the new 'notifycid' option is
+ set... but this was not the case. Instead the call-id value was
+ included for every RINGING Notify message, which caused a
+ regression for people who used other methods for call pickup.
+ (closes issue #17633) Reported by: urosh Patches: chan_sip.txt
+ uploaded by urosh (license ) blf_cid_issue.diff uploaded by
+ dvossel (license 671) Tested by: dvossel, urosh, okrief,
+ alecdavis
+
+2010-08-12 22:50 +0000 [r282130] Jason Parker <jparker at digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 282129 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) |
+ 1 line Register CLI commands before parsing config, in case there
+ is a config error. ........
+
+2010-08-12 03:01 +0000 [r281912] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, /: Merged revisions 281911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
+ | 20 lines Ensure SSRC is changed when media source is changed to
+ resolve audio delay. This change causes the SSRC to change right
+ before the channels are bridged, which is what used to happen. It
+ seems that fixes were made to attempt limiting SSRC changes,
+ targeted mainly at sending DTMF. DTMF is not affecting the SSRC
+ with this change. There are two other control frames sent in
+ ast_channel_bridge that probably should also be changed to
+ AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
+ up to the discretion of resolving issue #17007. For reference -
+ old review implementing new control frame SRCCHANGE:
+ https://reviewboard.asterisk.org/r/540 (closes issue #17404)
+ Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
+ (license 325) Tested by: sdolloff ........
+
+2010-08-11 21:09 +0000 [r281763-281873] Leif Madsen <lmadsen at digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 281819 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11
+ Aug 2010) | 6 lines Add Danish support to say.conf.sample (closes
+ issue #17836) Reported by: RoadKill Patches:
+ say.conf.sample.patch.dk uploaded by RoadKill (license 933)
+ ........
+
+ * configs/say.conf.sample, /: Merged revisions 281762 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11
+ Aug 2010) | 6 lines Allow say.conf to handle large numbers ending
+ with multiple zeros. (closes issue #17833) Reported by: RoadKill
+ Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
+ (license 933) ........
+
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