[asterisk-commits] lmadsen: tag 1.6.2.12-rc1 r283280 - /tags/1.6.2.12-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Aug 23 13:36:19 CDT 2010
Author: lmadsen
Date: Mon Aug 23 13:36:15 2010
New Revision: 283280
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=283280
Log:
Importing files for 1.6.2.12-rc1 release.
Added:
tags/1.6.2.12-rc1/.lastclean (with props)
tags/1.6.2.12-rc1/.version (with props)
tags/1.6.2.12-rc1/ChangeLog (with props)
Added: tags/1.6.2.12-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.12-rc1/.lastclean?view=auto&rev=283280
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--- tags/1.6.2.12-rc1/ChangeLog (added)
+++ tags/1.6.2.12-rc1/ChangeLog Mon Aug 23 13:36:15 2010
@@ -1,0 +1,26846 @@
+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) ........
+
+2010-08-11 15:17 +0000 [r281722] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_readexten.c: Only set status TIMEOUT, if we have no
+ digits. (closes issue #15188) Reported by: jcovert Patches:
+ app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
+ 551)
+
+2010-08-10 18:04 +0000 [r281567-281574] Russell Bryant <russell at digium.com>
+
+ * main/sched.c: Don't move the time threshold for running scheduled
+ events on every iteration. Instead, only calculate the time
+ threshold each time ast_sched_runq() is called. (closes issue
+ #17742) Reported by: schmidts Patches: sched.c.patch uploaded by
+ schmidts (license 1077)
+
+ * apps/app_dial.c, /: Merged revisions 281566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
+ | 8 lines Reset visible indication after answer. (closes issue
+ #17641) Reported by: klaus3000 Patches:
+ ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+ klaus3000 (license 65) Tested by: schmidts ........
+
+2010-08-09 20:46 +0000 [r281430] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: fixes SIP peers memory leak We zeroed out
+ the peer's addr before it was removed from the peers_by_ip
+ container. This made it impossible to be removed from the
+ container as the addr is the key used by the container to find
+ the peer. (closes issue #17774) Reported by: kkm Patches:
+ 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
+ 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
+
+2010-08-09 20:07 +0000 [r281391] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 281390 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09
+ Aug 2010) | 13 lines Prevent loss of Caller ID information set on
+ local channel after masquerade. Caller ID set on the channel
+ before a masquerade occurs when using a local channel would cause
+ the information to be lost. The problem was that the information
+ was set on a channel destined to be hung up. The somewhat
+ confusing fix is to detect if any Caller ID has been set on the
+ channel and if so preswap the Caller ID data so that basically
+ the masquerade puts the data back. (closes issue #17138) Reported
+ by: kobaz Review: https://reviewboard.asterisk.org/r/847/
+ ........
+
+2010-08-05 13:11 +0000 [r281051] Russell Bryant <russell at digium.com>
+
+ * main/cdr.c: Cleanup default option value handling for cdr.conf
+ [general]. The default values would differ depending on whether
+ or not cdr.conf exists. That is no longer the case. Apply a
+ default value to the unanswered option. Define all default values
+ as named constants.
+
+2010-08-05 07:40 +0000 [r280983] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280982
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
+ | 8 lines Change context lock back to a mutex, because
+ functionality depends upon the lock being recursive. (closes
+ issue #17643) Reported by: zerohalo Patches:
+ 20100726__issue17643.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........
+
+2010-08-03 20:52 +0000 [r280671-280812] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_callerid.c, channels/chan_dahdi.c, /: Merged revisions
+ 280811 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010)
+ | 9 lines Prevent DAHDI channels from overriding the callerid,
+ once it's been set by the user. (closes issue #16661) Reported
+ by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by
+ tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: jstapleton ........
+
+ * doc/asterisk.sgml, doc/asterisk.8, doc/Makefile (added): Document
+ -B and -W flags and regenerate manpage from sgml
+
+ * apps/app_voicemail.c: Allow the pipe, but also allow the comma
+
+2010-08-02 21:14 +0000 [r280669] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: Change SIP NOTIFY requests to expect a
+ response so authentication will work. This changes the request to
+ be sent with the transmit type XMIT_RELIABLE so that sip_ack
+ doesn't return false and cause the 401 to be ignored in cases
+ where authentication is required. (closes issue #14255) Reported
+ by: zktech
+
+2010-07-29 21:07 +0000 [r280556] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_curl.c: Off-by-one error (closes issue #17590)
+ Reported by: atis Patches: 20100729__issue17590.diff.txt uploaded
+ by tilghman (license 14)
+
+2010-07-29 20:42 +0000 [r280449-280551] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: fixes wrong SRV query for TLS connection
+ (closes issue #17612) Reported by: marcelloceschia Patches:
+ chan-sip_srvQuery.patch uploaded by marcelloceschia (license
+ 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
+ chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
+ (license 1079) Tested by: marcelloceschia, st, pabelanger
+
+ * main/channel.c, /: Merged revisions 280448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
+ | 12 lines fixes issue with translator frame not getting freed A
+ translator frame even if it local storage so the translation path
+ can be freed. This issue prevented g729 licenses from being freed
+ up. (closes issue #17630) Reported by: manvirr Patches:
+ encoder_fix.diff uploaded by dvossel (license 671) Tested by:
+ manvirr, dvossel ........
+
+2010-07-29 16:01 +0000 [r280345] Jean Galarneau <jgalarneau at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 280341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
+ 2 lines Fix a dsp structure leak occuring when a local channel is
+ put into a meetme conference, then masquaraded away. ABE-2422
+ ........
+
+2010-07-29 13:45 +0000 [r280306] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_local.c: Implement support for
+ ast_channel_queryoption on local channels. Currently only
+ AST_OPTION_T38_STATE is supported. ABE-2229 Review:
+ https://reviewboard.asterisk.org/r/813/
+
+2010-07-28 20:02 +0000 [r280231] Jason Parker <jparker at digium.com>
+
+ * sounds/Makefile: Work around some silly behavior on BSD. A
+ non-zero exit from a subshell should make the build fail. (closes
+ issue #17621)
+
+2010-07-28 19:57 +0000 [r280229] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Add missing enum value "unknown" to the
+ SS7 called_nai and calling_nai config options.
+
+2010-07-28 19:54 +0000 [r280193-280227] Jason Parker <jparker at digium.com>
+
+ * build_tools/sha1sum-sh (added): Add sha1sum-sh in case there is
+ no util on the system.
+
+ * sounds/Makefile: Remove unnecessary subshells. Attempt to make
+ checksumming work. Also improves readability. (issue #17621)
+ Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
+
+2010-07-28 16:51 +0000 [r280160] Sean Bright <sean at malleable.com>
+
+ * apps/app_queue.c: Plug a reference leak in app_queue when adding
+ members dynamically. (closes issue #17738) Reported by:
+ bobwienholt Patches: issue17738.patch uploaded by bobwienholt
+ (license 950) Tested by: bobwienholt, seanbright
+
+2010-07-28 13:51 +0000 [r280089] Leif Madsen <lmadsen at digium.com>
+
+ * contrib/scripts/live_ast, /: Merged revisions 280088 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
+ Jul 2010) | 1 line Update help text to be less confusing.
+ ........
+
+2010-07-27 20:54 +0000 [r279946] David Vossel <dvossel at digium.com>
+
+ * main/audiohook.c, main/channel.c, /,
+ include/asterisk/audiohook.h: Merged revisions 279945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
+ | 19 lines remove empty audiohook write list on channel If a
+ channel has an audiohook write list created on it, that list
+ stays on the channel until the channel is destroyed. There is no
+ reason to keep that list on the channel if it becomes empty. If
+ it is empty that just means we are doing needless translating for
+ every ast_read and ast_write. This patch removes the audiohook
+ list from the channel once it is detected to be empty on either a
+ read or write. If a audiohook is added back to the channel after
+ this list is destroyed, the list just gets recreated as if it
+ never existed to begin with. (closes issue #17630) Reported by:
+ manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
+
+2010-07-27 17:54 +0000 [r279849-279883] Jason Parker <jparker at digium.com>
+
+ * makeopts.in, configure, configure.ac: Add SHA1SUM to configure,
+ since we require it for sounds/
+
+ * sounds/Makefile: Remove aptly-named EMPTY and BS vars, since they
+ aren't used anymore.
+
+ * sounds/Makefile: Simply sounds/Makefile some more.
+
+2010-07-27 15:13 +0000 [r279784] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix bad behavior of dynamic_exclude_static
+ option in sip.conf. We were attempting to create a contactdeny
+ rule based on the peer's IP address before the peer's IP address
+ had been set. By moving the processing further down in the
+ function, we can ensure stuff works as we expect for it to.
+ (closes issue #17717) Reported by: mmichelson Patches:
+ 17717.patch uploaded by mmichelson (license 60) Tested by:
+ DennisD
+
+2010-07-26 22:59 +0000 [r279657] Jason Parker <jparker at digium.com>
+
+ * sounds/Makefile (added), sounds/Makefile.380 (removed),
+ configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
+ (removed), configure.ac: Really fix sounds Makefile (and make it
+ readableish). There was a rather large syntax error that should
+ have caused ALL versions of GNU make to fail. I don't know how it
+ worked.
+
+2010-07-26 21:18 +0000 [r279609] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, configure.ac: Dunno why this worked on my machine, but
+ it works better this way.
+
+2010-07-26 20:25 +0000 [r279597] Gavin Henry <ghenry at suretecsystems.com>
+
+ * res/res_config_ldap.c: Apply all patches in:
+ https://issues.asterisk.org/view.php?id=13573 (closes issue
+ #13573) Reported by: navkumar Patches:
+ res_config_ldap-category.diff uploaded by navkumar (license 580)
+ res_config_ldap.patch uploaded by bencer (license 961)
+ res_config_ldap uploaded by bencer (license 961) Tested by:
+ suretec
+
+2010-07-26 19:15 +0000 [r279561] Tilghman Lesher <tlesher at digium.com>
+
+ * sounds/Makefile (removed), configure, sounds/Makefile.380
+ (added), sounds/Makefile.381 (added), configure.ac: Use a special
+ Makefile for noobs who still have GNU Make 3.80. (Closes issue
+ #17716) Reported by: farisraouf
+
+2010-07-26 15:41 +0000 [r279501] Sean Bright <sean at malleable.com>
+
+ * autoconf/ast_ext_lib.m4: Expand the correct value within
+ AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
+
+2010-07-24 23:58 +0000 [r279347] Bradley Latus <brad.latus at gmail.com>
+
+ * doc/asterisk.8: Minor update to man page
+
+2010-07-23 22:11 +0000 [r279207] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279206 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
+ | 7 lines SIP promiscuous redirect could fail to dial the
+ redirect. The ast_channel was created with one variable to
+ ast_request() but the call to ast_call() that initiates the
+ outgoing call was using a different variable. The two variables
+ are not equivalent if the call_forward string included a channel
+ technology specifier. e.g., SIP/200 ........
+
+2010-07-23 18:29 +0000 [r279112] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Backport sip_uri_params_cmp() fix from trunk
+ to 1.6.2.
+
+2010-07-23 18:22 +0000 [r279072-279088] Russell Bryant <russell at digium.com>
+
+ * /: remove old properties
+
+ * /: Add branch-1.4-merged and branch-1.4-blocked properties to
+ 1.6.2 branch.
+
+2010-07-23 17:06 +0000 [r278983-278986] Tilghman Lesher <tlesher at digium.com>
+
+ * autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
+ revisions 278985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278985 | tilghman | 2010-07-23 12:05:16 -0500 (Fri, 23 Jul 2010)
+ | 12 lines Merged revisions 278984 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
+ | 5 lines Establish a maximum version for openh323 (i.e. not
+ opal), because chan_h323 will fail to load, even if it links.
+ (issue #17679) Reported by: am ........ ................
+
+ * main/asterisk.c, /: Merged revisions 278982 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010)
+ | 15 lines Merged revisions 278981 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
+ | 8 lines Avoid race with consolethread on shutdown (on parallel
+ processors). (closes issue #17080) Reported by: sybasesql
+ Patches: 20100721__issue17080.diff.txt uploaded by tilghman
+ (license 14) Tested by: sybasesql ........ ................
+
+2010-07-23 15:23 +0000 [r278934] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * channels/chan_dahdi.c: Two more typos to cancell.
+
+2010-07-22 19:52 +0000 [r278709] Jeff Peeler <jpeeler at digium.com>
+
+ * main/xmldoc.c, /: Merged revisions 278708 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r278708 |
+ jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines
+ Add method for finding XML doc files for systems that don't
+ support GLOB_BRACE. In particular, Solaris and perhaps others do
+ not support the above mentioned GNU extension. In this case the
+ paths are simply expanded without the braces and the calls to
+ glob are made separately. Note: I could not explain memory
+ allocation failures that were being reported from within libxml
+ itself when making calls to glob without using GLOB_NOCHECK. This
+ is the only reason why that flag is being used. (closes issue
+ #15402) Reported by: snuffy Patches: bug_xmlpatt-v3.diff uploaded
+ by snuffy (license 35), modified by me ........
+
+2010-07-22 19:32 +0000 [r278703] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: DNID does not get cleard on a new call
+ when using immediate=yes with ISDN signaling. When you are using
+ chan_dahdi ISDN signaling with immediate=yes and a call comes in
+ without a DNID then you get the DNID of a previous call.
+ Chan_dahdi does not touch the DNID field on a new call if it does
+ not have a DNID. Made always copy the DNID from the new call. The
+ patches backport the relevant changes from trunk -r210387.
+ (closes issue #17568) Reported by: wuwu Patches:
+ issue17568_v1.4.patch uploaded by rmudgett (license 664)
+ issue17568_v1.6.2.patch uploaded by rmudgett (license 664)
+
+2010-08-10 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.11 Released.
+
+2010-07-26 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.11-rc2 Released.
+
+2010-07-26 Leif Madsen <lmadsen at digium.com>
+
+ * qwell, asterisk, branch-1.6.2, r279657 ***
+ Really fix sounds Makefile (and make it readableish).
+ There was a rather large syntax error that should have
+ caused ALL versions of GNU make to fail.
+ I don't know how it worked.
+
+ (Closes issue #17716)
+
+2010-07-22 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.11-rc1 Released.
+
+2010-07-22 15:00 +0000 [r278621] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c, /: Merged revisions 278620 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul
+ 2010) | 19 lines Merged revisions 278618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul
+ 2010) | 13 lines Allow PLC to function properly when channels use
+ SLIN for audio. If a channel involved in a bridge was using SLIN
+ audio, then translation paths were not guaranteed to be set up
+ properly since in all likelihood the number of translation steps
+ was only 1. This patch enforces the transcode_via_slin behavior
+ if transcode_via_slin or generic_plc is enabled and one of the
+ formats to make compatible is SLIN. AST-352 ........
+ ................
+
+2010-07-21 18:22 +0000 [r278524] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * channels/chan_dahdi.c: Fix invalid test for rxisoffhook in FXO
+ channels This fixes some cases of no outgoing calls on FXO before
+ an incoming call. Remove an unnecessary testing of an "off-hook"
+ bit from DAHDI for FXO (KS/GS) channels.In some cases the bit
+ would not be initialized properly before the first inbound call
+ and thus prevent an outgoing call. If those tests are actually
+ required by anybody, they should define DAHDI_CHECK_HOOKSTATE in
+ channels/sig_analog.c . (closes issue #14577) Reported by: jkroon
+ Patches: asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd
+ (license 610) Tested by: frawd Review:
+ https://reviewboard.asterisk.org/r/699/
+
+2010-07-21 16:20 +0000 [r278479] Russell Bryant <russell at digium.com>
+
+ * /, res/res_timing_pthread.c: Merged revisions 278465 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010)
+ | 41 lines Use poll() instead of select() in res_timing_pthread
+ to avoid stack corruption. This code did not properly check
+ FD_SETSIZE to ensure that it did not try to select() on fds that
+ were too large. Switching to poll() removes the limitation on the
+ maximum fd value. (closes issue #15915) Reported by: keiron
+ (closes issue #17187) Reported by: Eddie Edwards (closes issue
+ #16494) Reported by: Hubguru (closes issue #15731) Reported by:
+ flop (closes issue #12917) Reported by: falves11 (closes issue
+ #14920) Reported by: vrban (closes issue #17199) Reported by:
+ aleksey2000 (closes issue #15406) Reported by: kowalma (closes
+ issue #17438) Reported by: dcabot (closes issue #17325) Reported
+ by: glwgoes (closes issue #17118) Reported by: erikje possibly
+ other issues, too ... ........
+
+2010-07-21 15:58 +0000 [r278025-278464] Tilghman Lesher <tlesher at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 278463 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 |
+ tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11
+ lines Ensure realtime conferences are treated the same as static
+ conferences when trying to find an empty one. Also, parse the
+ useropts properly, when retrieving from realtime, and add them to
+ the existing flags. (closes issue #17502) Reported by: kenji
+ Patches: 20100720__issue17502.diff.txt uploaded by tilghman
+ (license 14) Tested by: kenji ........
+
+ * apps/app_voicemail.c, /: Merged revisions 278275 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r278275 | tilghman | 2010-07-20 17:40:19 -0500
+ (Tue, 20 Jul 2010) | 14 lines Merged revisions 278261 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010)
+ | 7 lines Delete IMAP messages in reverse order, to ensure
+ reordering after each expunge does not cause deletion of the
+ wrong message. (closes issue #16350) Reported by: noahisaac
+ Patches: 20100623__issue16350.diff.txt uploaded by tilghman
+ (license 14) ........ ................
+
+ * main/autoservice.c, /, main/features.c,
+ include/asterisk/channel.h: Merged revisions 278272 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r278272 | tilghman | 2010-07-20 17:26:23 -0500
+ (Tue, 20 Jul 2010) | 11 lines Merged revisions 278167 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010)
+ | 4 lines Do not queue up DTMF frames while a call is on hold.
+ (Fixes ABE-2110) ........ ................
+
+ * main/manager.c, /: Merged revisions 278024 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010)
+ | 14 lines Merged revisions 278023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010)
+ | 7 lines Off-by-one error (closes issue #16506) Reported by:
+ nik600 Patches: 20100629__issue16506.diff.txt uploaded by
+ tilghman (license 14) ........ ................
+
+2010-07-19 21:21 +0000 [r277966] Jean Galarneau <jgalarneau at digium.com>
+
+ * /, main/features.c: Merged revisions 277945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) |
+ 15 lines Merged revisions 277906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) |
+ 7 lines Avoid trying to pickup a parked extension before the park
+ operation is completed. A crash could occur if the extension is
+ picked up while the parking extension is being announced. Testing
+ pu->notquiteyet while searching for a parked extension resolves
+ this crash. (ABE-2418) ........ ................
+
+2010-07-17 17:52 +0000 [r277774-277777] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_pgsql.c: Merge issues...
+
+ * /, autoconf/ast_func_fork.m4, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 277775 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r277775 | tilghman | 2010-07-17 12:42:32 -0500
+ (Sat, 17 Jul 2010) | 12 lines Merged revisions 277738 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010)
+ | 5 lines Remove uclibc cross-compile triplet, as uclibc has a
+ working fork()... it's only uclinux that does not. (closes issue
+ #17616) Reported by: pprindeville ........ ................
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
+ revisions 277773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010)
+ | 15 lines Merged revisions 277568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010)
+ | 8 lines Since we split values at the semicolon, we should store
+ values with a semicolon as an encoded value. (closes issue
+ #17369) Reported by: gkservice Patches:
+ 20100625__issue17369.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........ ................
+
+2010-07-16 23:37 +0000 [r277666] Tim Ringenbach <tim.ringenbach at gmail.com>
+
+ * /, main/features.c: Merged revisions 277657 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul
+ 2010) | 16 lines Merged revisions 277625 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul
+ 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on
+ attended transfer. ast_bridge_call() clears
+ AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
+ ast_bridge_call() is called for a second bridge on the same
+ channel, and it clears that flag, which still needs to get set
+ for when the original ast_bridge_call() gets control back and
+ checks it. Review: https://reviewboard.asterisk.org/r/741
+ ........ ................
+
+2010-07-16 21:31 +0000 [r277563] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 277530 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul
+ 2010) | 11 lines Merged revisions 277497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul
+ 2010) | 4 lines Default to no udptl error correction so that
+ error correction will be disabled in the event that the remote
+ end indicates that they do not support the error correction mode
+ we requested. FAX-128 ........ ................
+
+2010-07-16 21:16 +0000 [r277489] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 277488 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 |
+ jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines
+ Fix reporting estimated queue hold time. Just say the number of
+ seconds (after minutes) rather than doing some incorrect
+ calculation with respect to minutes. (closes issue #17498)
+ Reported by: corruptor Patches: holdesecs_bug.diff uploaded by
+ corruptor (license 253) ........
+
+2010-07-16 20:35 +0000 [r277485] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 277467 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r277467 | rmudgett | 2010-07-16 15:27:51 -0500
+ (Fri, 16 Jul 2010) | 22 lines Merged revisions 277419 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010)
+ | 15 lines priexclusive in chan_dahdi.conf ignored when reloading
+ dahdi module During a reload, the priexclusive and outsignalling
+ parameters are not read in from the config file as intended.
+ Unfortunately, they get set to defaults as a result. This patch
+ makes sure that they do not get set to defaults during a reload.
+ (closes issue #17441) Reported by: mtryfoss Patches:
+ issue17441_v1.4.patch uploaded by rmudgett (license 664)
+ issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
+ issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
+ by: rmudgett ........ ................
+
+2010-07-16 20:30 +0000 [r277478] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
+ (added), /: Merged revisions 277452 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277452 |
+ tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines
+ Add documentation for MOH realtime fields ........
+
+2010-07-16 19:24 +0000 [r277377] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 277366 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 |
+ jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines
+ Add missing handling for ringing state for use with queue empty
+ options. (closes issue #17471) Reported by: jazzy Patches:
[... 26109 lines stripped ...]
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