[asterisk-commits] lmadsen: tag 1.8.3-rc1 r301838 - /tags/1.8.3-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jan 14 12:32:50 CST 2011
Author: lmadsen
Date: Fri Jan 14 12:32:46 2011
New Revision: 301838
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=301838
Log:
Importing files for 1.8.3-rc1 release.
Added:
tags/1.8.3-rc1/.lastclean (with props)
tags/1.8.3-rc1/.version (with props)
tags/1.8.3-rc1/ChangeLog (with props)
Added: tags/1.8.3-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.3-rc1/.lastclean?view=auto&rev=301838
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==============================================================================
--- tags/1.8.3-rc1/ChangeLog (added)
+++ tags/1.8.3-rc1/ChangeLog Fri Jan 14 12:32:46 2011
@@ -1,0 +1,27088 @@
+2011-01-14 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.3-rc1 Released.
+
+2011-01-14 17:32 +0000 [r301790] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: Resolve deadlock involving REFER. Two fixes:
+ 1) One must always have the private unlocked before calling
+ pbx_builtin_setvar_helper to not invalidate locking order since
+ it locks the channel. 2) Unlock the channel before calling
+ pbx_find_extension, which starts and stops autoservice during the
+ lookup. The problem scenario as illustrated by the reporter:
+ Thread: do_monitor ----------------------- handle_request_do
+ handle_incoming handle_request_refer ast_parking_ext_valid
+ pbx_find_extension ast_autoservice_stop while (chan_list_state ==
+ as_chan_list_state) { usleep(1000); } Thread: autoservice_run
+ ----------------------- autoservice_run chan = ast_waitfor_n
+ ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple /
+ complex (depending on your system) ast_channel_lock(c[x]);
+ handle_request_do and schedule_process_request_queue locks the
+ owner if it exists. The autoservice thread is waiting for the
+ channel lock, which wasn't ever released since the do_monitor
+ thread was waiting for autoservice operations to complete. Solved
+ by unlocking the channel but keeping a reference to guarantee
+ safety. (closes issue #18403) Reported by: jthurman Patches:
+ 20110103-blind_deadlock.diff uploaded by jthurman (license 614)
+ issue18403.patch uploaded by jpeeler (license 325) Tested by:
+ jthurman
+
+2011-01-13 17:01 +0000 [r301731] Leif Madsen <lmadsen at digium.com>
+
+ * configs/phoneprov.conf.sample, /: Merged revisions 301730 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011)
+ | 7 lines Add static entry for split Polycom 332 firmware.
+ (closes issue #18607) Reported by: cjacobsen Patches:
+ polycom_331.diff uploaded by cjacobsen (license 1029) Tested by:
+ lathama ........
+
+2011-01-12 21:19 +0000 [r301683] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 301682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011)
+ | 9 lines Don't reject all SUBSCRIBE auth requests When merging
+ another SUBSCRIBE fix from 1.4, some braces were put in the wrong
+ place. This patch fixes that. (closes issue #18597) Reported by:
+ thsgmbh ........
+
+2011-01-12 18:51 +0000 [r301595] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/manager.c, /: Merged revisions 301594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600
+ (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't
+ be necessary in session_do, and removed the ms_t member from the
+ mansession_session structure. Merged revisions 301591 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan
+ 2011) | 5 lines Don't store the thread id for the manager session
+ in the structure we pass to the thread for the manager session.
+ ABE-2543 ........ ................
+
+2011-01-12 18:12 +0000 [r301504] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, /: Merged revisions 301503 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600
+ (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011)
+ | 12 lines Fix CPU spike when pressing DTMF after agent login.
+ The problem here is that DTMF was being continuously deferred and
+ requeued since ast_safe_sleep is called in a loop. There are
+ serveral other places in the code that sleeps and then loops in a
+ similar fashion. Because of this fact I opted to not defer DTMF
+ any more, which will not affect the original fix:
+ https://reviewboard.asterisk.org/r/674 (closes issue #18130)
+ Reported by: rgj ........ ................
+
+2011-01-12 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.2 Released.
+
+ * Merge in a change in the configure script to fix an issue for
+ Debian packagers.
+
+ ------------------------------------------------------------------------
+ r301221 | pabelanger | 2011-01-09 15:40:35 -0600 (Sun, 09 Jan 2011)
+ | 21 lines
+
+ Merged revisions 301220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 [^]
+
+ ........
+ r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan
+ 2011) | 14 lines
+ SOUND_CACHE_DIR now defaults to empty
+
+ Sounds files included in the Asterisk tarball were being
+ ignored and
+ re-downloaded. Users wanting to cache the files can
+ still override the setting
+ using the --with-sounds-cache option.
+
+ (closes issue 0018589)
+ Reported by: pabelanger
+ Patches:
+ issue18589.patch uploaded by
+ pabelanger (license 224)
+ Tested by: pabelanger
+
+ Review:
+ https://reviewboard.asterisk.org/r/1074/
+
+ ------------------------------------------------------------------------
+
+2010-12-13 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.2-rc1 Released.
+
+2010-12-11 21:45 +0000 [r298099] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c: Correction to work with
+ gatekeeper which don't send GK ID Don't use GK ID if it's not
+ presented in GK replies Extract GK ID not only in GK confirm but
+ in GK register confirm also (issue #18401) Reported by: MrHanMan
+ Patches: no-gkid-2.patch uploaded by may213 (license 454) Tested
+ by: may213, MrHanMan
+
+2010-12-10 16:52 +0000 [r298054] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_fax.c: Prevent a memcpy overlap in
+ GENERIC_FAX_EXEC_SET_VARS
+
+2010-12-10 16:26 +0000 [r298051] Tilghman Lesher <tlesher at digium.com>
+
+ * main/netsock.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Merged revisions 298050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010)
+ | 11 lines Portability issue on OpenSolaris. Also detect the
+ required structure element, because OpenSolaris defines
+ SIOCGIFHWADDR, but without support for IP sockets. (closes issue
+ #18442) Reported by: ranjtech Patches:
+ 20101209__issue18442.diff.txt uploaded by tilghman (license 14)
+ Tested by: ranjtech ........
+
+2010-12-09 22:18 +0000 [r297965] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297960 | twilson | 2010-12-09 16:10:31 -0600
+ (Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010)
+ | 14 lines Ignore spurious REGISTER requests If a REGISTER
+ request with a Call-ID matching an existing transaction is
+ received it was possible that the REGISTER request would
+ overwrite the initreq of the private structure. This info is used
+ to generate messages for other responses in the transaction. This
+ patch ignores REGISTER requests that match non-REGISTER
+ transactions. (closes issue #18051) Reported by: eeman Tested by:
+ twilson Review: https://reviewboard.asterisk.org/r/1050/ ........
+ ................
+
+2010-12-09 21:32 +0000 [r297957] David Vossel <dvossel at digium.com>
+
+ * channels/chan_gtalk.c: Fixes issue with outbound google voice
+ calls not working. Thanks to az1234 and nevermind_quack for their
+ input in helping debug the issue. (closes issue #18412) Reported
+ by: nevermind_quack Patches: fix uploaded by dvossel (license
+ 671)
+
+2010-12-09 20:48 +0000 [r297952] Terry Wilson <twilson at digium.com>
+
+ * main/features.c: Don't crash after Set(CDR(userfield)=...) in
+ ast_bridge_call Instead of setting peer->cdr = NULL, set it to
+ not post. (closes issue #18415) Reported by: macbrody Patches:
+ patch-18415 uploaded by jsolares (license 1167) Tested by:
+ jsolares, twilson
+
+2010-12-08 18:06 +0000 [r297909] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 297908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r297908 | tilghman | 2010-12-08 12:04:38 -0600 (Wed, 08 Dec 2010)
+ | 4 lines Use inheritance to get correct results for
+ SIPFROMDOMAIN. (from an internal Digium discussion) ........
+
+2010-12-08 16:12 +0000 [r297905] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_fax.c: Display the capabilities requested when requesting
+ a fax session fails instead of displaying a hex value. Tweak the
+ way fax stats are calculated so that all fax attempts and
+ faliures are logged. Also make ensure faxes are either counted as
+ completed or falied and never both. FAX-210
+
+2010-12-07 22:59 +0000 [r297825] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, /: Merged revisions 297824 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600
+ (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010)
+ | 12 lines Revert code that changed SSRC for DTMF. Some previous
+ behavior was attempted to be restored, but mistakingly I did not
+ realize that the previous behavior was incorrect. This fixes DTMF
+ not being detected since DTMF shouldn't cause the SSRC to change.
+ (related to issue #17404) (closes issue #18189) (closes issue
+ #18352) Reported by: marcbou Tested by: cmbaker82 ........
+ ................
+
+2010-12-07 22:51 +0000 [r297733-297821] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/org.asterisk.muted.plist (added), Makefile,
+ contrib/init.d/org.asterisk.asterisk.plist, utils/muted.c, /:
+ Merged revisions 297819 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297819 | tilghman | 2010-12-07 16:40:45 -0600
+ (Tue, 07 Dec 2010) | 11 lines Merged revisions 297818 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010)
+ | 4 lines Use non-deprecated APIs for CoreAudio Review:
+ https://reviewboard.asterisk.org/r/1040/ ........
+ ................
+
+ * apps/app_followme.c, /: Merged revisions 297713 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297713 | tilghman | 2010-12-06 18:21:50 -0600
+ (Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010)
+ | 8 lines Don't create a Local channel if the target extension
+ does not exist. (closes issue #18126) Reported by: junky Patches:
+ followme.diff uploaded by junky (license 177) (partially
+ restructured by me to avoid a possible memory leak) ........
+ ................
+
+2010-12-06 22:06 +0000 [r297607] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297605 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600
+ (Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010)
+ | 12 lines Improve handling of REGISTER requests with multiple
+ contact headers. The changes here attempt to more strictly follow
+ RFC 3261 section 10.3. Basically the following will now cause a
+ 400 Bad Response to be returned, if: - multiple Contact headers
+ are present with one set to expire all bindings ("*") - wildcard
+ parameter is specified for Contact without Expires header or
+ Expires header is not set to zero. ABE-2442 ABE-2443 ........
+ ................
+
+2010-12-03 17:41 +0000 [r297535] Sean Bright <sean at malleable.com>
+
+ * channels/chan_console.c, /: Merged revisions 297534 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r297534 | seanbright | 2010-12-03 12:40:52 -0500 (Fri,
+ 03 Dec 2010) | 3 lines The CLI command should not contain
+ <placeholder>s, these are for descriptions. ........
+
+2010-12-03 15:21 +0000 [r297486-297495] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_fax.c: Print a DEBUG message instead of a WARNING message
+ when the selected fax tech does not support reserving sessions.
+ Answer the channel before quering it for t.38 support. This is
+ necessary for the query to work properly over local channels.
+
+ * include/asterisk/res_fax.h, res/res_fax.c: Add support for
+ reserving a fax session before answering the channel. Note: this
+ change breaks ABI compatibility. FAX-217
+
+2010-12-02 20:09 +0000 [r297406] Paul Belanger <pabelanger at digium.com>
+
+ * Makefile, /: Merged revisions 297405 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297405 | pabelanger | 2010-12-02 15:06:43 -0500
+ (Thu, 02 Dec 2010) | 14 lines Merged revisions 297404 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec
+ 2010) | 7 lines Resolve compile error under FreeBSD We now set
+ _ASTCFLAGS+=-march=i686 for i386 processors, still allowing
+ ASTCFLAGS to override the setting. Review:
+ https://reviewboard.asterisk.org/r/1043/ ........
+ ................
+
+2010-12-02 18:13 +0000 [r297312] Terry Wilson <twilson at digium.com>
+
+ * /, main/abstract_jb.c: Merged revisions 297311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297311 | twilson | 2010-12-02 12:07:39 -0600
+ (Thu, 02 Dec 2010) | 21 lines Merged revisions 297310 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010)
+ | 12 lines Initialize offset for adaptive jitter buffer When the
+ adaptive jitter buffer is enabled in sip.conf, the first frame
+ placed in the jitter buffer fails with something like:
+ jb_warning_output: Resyncing the jb. last_delay 0, this delay
+ -215886466, threshold 1000, new offset 215886466 This happens
+ because the offset is not initialized before calling jb_put().
+ This patch modifies jb_put_first_adaptive() to set the offset to
+ the frame's timestamp. Review:
+ https://reviewboard.asterisk.org/r/1041/ ........
+ ................
+
+2010-12-02 13:20 +0000 [r297245] Russell Bryant <russell at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 297229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297229 | russell | 2010-12-02 07:16:47 -0600
+ (Thu, 02 Dec 2010) | 13 lines Merged revisions 297228 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010)
+ | 6 lines Add "DAHDI" to a couple of app_meetme error messages.
+ This is in response to some questions on IRC. To the user, there
+ was nothing that made it obvious that this error had anything to
+ do with DAHDI not being loaded. ........ ................
+
+2010-12-01 19:47 +0000 [r297157] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_fax.c: Changed some NOTICE and WARNING messages to DEBUG
+ messages.
+
+2010-12-01 17:53 +0000 [r297075] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 297073 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600
+ (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010)
+ | 23 lines Fix not stopping MOH when transfered local channel
+ queue member is answered. The problem here is only present when
+ local channels are used with the MOH passthru option as well as
+ no optimization (/nm). I will describe the slightly bizarre
+ scenario that was used to test, where phones B and C are queue
+ members: Phone A dials into a queue with two members using local
+ channels and the above options. Phone B answers. Phone A blind
+ transfers phone B into the same queue. Phone A hangs up. Phone C
+ answers, but phone B didn't stop playing MOH. In this scenario,
+ the unhold frame that should have gotten to phone B never arrived
+ due to the masquerade from the blind transfer. This is usually
+ fine since app_queue manages the starting and stopping of MOH.
+ However, with the passthrough option enabled when app_queue
+ attempts to stop MOH it tries to do so on the local channel
+ rather than the real channel. The easiest solution was to just
+ make sure to send an unhold frame during the transfer since it
+ wouldn't make sense to have MOH playing after a transfer anyway.
+ This only modifies SIP transfers, but the other transfers did not
+ seem to be a problem. If DTMF based transfers were a problem it
+ might be okay to add ast_moh_stop to finishup, but I didn't want
+ to have to add that unless required. ABE-2624 ........
+ ................
+
+2010-12-01 17:01 +0000 [r296951-296992] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/frame.h, /: Merged revisions 296991 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296991 | tilghman | 2010-12-01 11:01:00 -0600
+ (Wed, 01 Dec 2010) | 12 lines Merged revisions 296990 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010)
+ | 5 lines Clarify documentation on how we store codec preference
+ lists. (closes issue #18397) Reported by: birgita ........
+ ................
+
+ * channels/chan_iax2.c, /: Merged revisions 296950 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r296950 | tilghman | 2010-11-30 19:38:19 -0600 (Tue, 30
+ Nov 2010) | 2 lines Missed initializations caused startup errors
+ on Mac OS X (and possibly others, too). ........
+
+2010-12-01 00:28 +0000 [r296870] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 296869 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296869 | jpeeler | 2010-11-30 18:24:58 -0600
+ (Tue, 30 Nov 2010) | 11 lines Merged revisions 296868 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010)
+ | 4 lines Properly restore backup information file when hanging
+ up during message prepending. ABE-2654 ........ ................
+
+2010-11-30 19:12 +0000 [r296787] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_meetme.c: DOC: Conference number can be omitted; if
+ omitted, all users in a meetme are listed.
+
+2010-11-29 23:05 +0000 [r296673] Paul Belanger <pabelanger at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 296671 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296671 | pabelanger | 2010-11-29 17:54:14 -0500
+ (Mon, 29 Nov 2010) | 12 lines Merged revisions 296670 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov
+ 2010) | 5 lines Make sure nothing else is needed before
+ destroying the scheduler. (closes issue #18398) Reported by:
+ pabelanger ........ ................
+
+2010-11-29 21:26 +0000 [r296628] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Complete some error handling in
+ transmit_publish() in chan_sip.c. This error handling block
+ caught my eye. It was missing a couple of things, but it should
+ be safe now. Thanks to mmichelson for the quick peer review on
+ IRC.
+
+2010-11-29 20:46 +0000 [r296582] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_msg_parser.c, channels/chan_misdn.c: Merged
+ revision 296575 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon,
+ 29 Nov 2010) | 13 lines Invalid mISDN PTMP redirecting signaling
+ as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY
+ redirecting number and notification code, SETUP redirecting
+ number) is also sent in PTMP/TE mode. It should only apply in
+ PTMP/NT mode. The call setup proceeds but the network (Deutsche
+ Telekom) reacts with ugly ISDN STATUS messages. Also don't send
+ the redirecting number ie when PTP is also sending the
+ DivertingLegInformation2 facility. The redirecting number ie is
+ redundant and the network (Deutsche Telekom) complains about it.
+ Patches: abe_2651_v4.patch uploaded by rmudgett (license 664)
+ JIRA ABE-2651 JIRA SWP-2537 ..........
+
+2010-11-29 07:28 +0000 [r296534] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Merged revisions 296533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010)
+ | 13 lines I love standards. There are so many to choose from.
+ Except when there isn't one. Linux and *BSD disagree on the
+ elements within the ucred structure. Detect which one is in use
+ on the system. (closes issue #18384) Reported by: bjm Patches:
+ cred-diffs uploaded by bjm (license 473)
+ 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman
+ (license 14) 20101127__issue18384__1.8.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman, bjm ........
+
+2010-11-27 10:40 +0000 [r296429-296467] Tilghman Lesher <tlesher at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 296466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010)
+ | 5 lines 18 characters is too short for most date/times (20 is
+ the usual, but we add more in case of greater precision). (closes
+ issue #18369) Reported by: tnakonz ........
+
+ * include/asterisk.h: Also don't build DEBUG_FD_LEAKS when
+ STANDALONE2 is defined. (closes issue #18385) Reported by: cmaj
+
+2010-11-26 21:37 +0000 [r296391] Olle Johansson <oej at edvina.net>
+
+ * main/say.c: Merged revisions 296351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296351 | oej | 2010-11-26 13:23:03 +0100 (Fre,
+ 26 Nov 2010) | 17 lines Merged revisions 296309 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11
+ lines Fix bugs in saying numbers using the Swedish language
+ syntax (closes issue #18355) Reported by: oej Patch by: oej Much
+ help from Peter Lindahl. Testing by the ClearIT team during a
+ coffee break. Review: https://reviewboard.asterisk.org/r/1033/
+ ........ ................
+
+2010-11-26 18:31 +0000 [r296352-296354] Brad Watkins <Marquis42 at gmail.com>
+
+ * res/res_jabber.c: Fix XMPP PubSub-based distributed device state.
+ Initialize pubsubflags to 0 so res_jabber doesn't think there is
+ already an XMPP connection sending device state. Also clean up
+ CLI commands a bit. (closes issue #18272) Reported by: klaus3000
+ Patches: res_jabber_fix_pubsubflags_and_CLI-patch.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000, Marquis Review:
+ https://reviewboard.asterisk.org/r/1030/
+
+ * channels/chan_sip.c: Fix reloading of peer when a user is
+ requested. Prevent peer reloading from causing multiple MWI
+ subscriptions to be created when using realtime. This had the
+ effect of sending one NOTIFY for every time a sip peer made a
+ call, in one case eventually overwhelming the phone and causing
+ it to reboot. (closes issue #18342) Reported by: nivek Patches:
+ issue0018342p1.patch uploaded by nivek (license 636) Tested by:
+ nivek Review: https://reviewboard.asterisk.org/r/1029/
+
+2010-11-24 23:29 +0000 [r296230] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, /: Merged revisions 296221 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296221 | russell | 2010-11-24 17:28:19 -0600
+ (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010)
+ | 6 lines Make Asterisk less crashy. Since we might not put a new
+ translation path on the channel, go ahead and set it to NULL
+ right after destroying the old one to ensure we don't try to free
+ an invalid translation path later on. ........ ................
+
+2010-11-24 22:49 +0000 [r296167] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
+ /, channels/sig_analog.h: Merged revisions 296166 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296166 | rmudgett | 2010-11-24 16:42:45 -0600
+ (Wed, 24 Nov 2010) | 50 lines Merged revisions 296165 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010)
+ | 43 lines Oneway audio to SIP phone from FXS port after FXS port
+ gets a CallWaiting pip. The FXS connected phone has to have
+ CW/CID support to fail, as it will send back a DTMF 'A' or 'D'
+ when it's ready to receive CallerID. A normal phone with no CID
+ never fails. Also the SIP phone does not hear MOH when the CW
+ call is answered. The DTMF end frame is suppressed when the phone
+ acknowledges the CW signal for CID. The problem is the DTMF begin
+ frame needs to be suppressed as well. The DTMF begin frame is
+ causing SIP to start sending the DTMF RTP frames. Since the DTMF
+ end frame is suppressed, SIP will not stop sending those DTMF RTP
+ packets. * Suppress the DTMF begin and end frames when the
+ channel driver is looking for DTMF digits. * Fixed a couple
+ issues caused by not cleaning up the CID spill if you answer the
+ CW call while it is sending the CID spill. * Fixed not sending
+ CW/CID spill to the phone when the call is natively bridged.
+ (Fixed by not using native bridge if CW/CID is possible.) *
+ Suppress received audio when sending CW/CID spills. The other
+ parties involved do not need to hear the CW/CID spills and may be
+ confused if the CW call is for them. (closes issue #18129)
+ Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ NOTE: * v1.4 does not have the main problem fixed by suppressing
+ the DTMF start frames. The other three items fixed are relevant.
+ * If you really must restore native bridging between analog
+ ports, you need to disable CW/CID either by configuring
+ chan_dahdi.conf callwaitingcallerid=no or dialing *70 before
+ dialing the number to temporarily disable CW. ........
+ ................
+
+2010-11-24 20:23 +0000 [r296002-296084] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, /: Merged revisions 296083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296083 | russell | 2010-11-24 14:23:11 -0600
+ (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010)
+ | 12 lines Fix false reporting of an error by set_format(). In
+ the case that the native format was able to be changed to match
+ the new requested format, the code proceeded to attempt to build
+ a translation path, anyway. The result would be NULL, since no
+ translation path is necessary and resulted in this function
+ thinking an error has occurred. This case is now specifically
+ caught and no attempt to build a translation path is attempted.
+ Thanks to our automated tests and bamboo.asterisk.org for
+ catching this problem and making a whole lot of noise when things
+ started failing. :-) ........ ................
+
+ * apps/app_dial.c, main/channel.c, /: Merged revisions 296001 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r296001 | russell | 2010-11-24 11:03:16 -0600
+ (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010)
+ | 38 lines Handle failures building translation paths more
+ effectively. The problem scenario occurred on a heavily loaded
+ system that was using the codec_dahdi module and exceeded the
+ hardware transcoding capacity. The failure mode at that point was
+ not good. The report came in to us as an Asterisk lock-up. The
+ "core show locks" shows a ton of threads locked up (but no
+ obvious deadlock). Upon deeper investigation, when the system is
+ in this state, the CPU was maxed out. The CPU was being consumed
+ by the Asterisk logger spewing messages on every audio frame for
+ calls set up after transcoder capacity was reached. The purpose
+ of this patch is to make Asterisk handle failures to create a
+ translation path in a more graceful manner. If we can't
+ translate, then the call just needs to be dropped, as it's not
+ going to work. These are the changes: 1) In set_format() of
+ channel.c (which is called by set_read_format() and
+ set_write_format()), it was ignoring if
+ ast_translator_build_path() failed and returned NULL. It now pays
+ attention to that case and returns a result reflecting failure.
+ With this change in place, the bridging code will immediately
+ detect a failure and end the bridge instead of proceeding to try
+ to bridge frames that can't be translated and making channel
+ drivers freak out by sending them frames in a format they weren't
+ expecting. 2) In ast_indicate_data() of channel.c, failure of
+ ast_playtones_start() was ignored. It is now reflected in the
+ return value of the function. This didn't turn out to have any
+ affect on the bug, but seemed like a good change to leave in. 3)
+ In app_dial(), when only sending a call to a single endpoint, it
+ will attempt to do some bridging of its own of early audio. It
+ uses make_compatible() when it's going to do this. However, it
+ ignored failure from make compatible. So, even with the fix from
+ #1, if there was early audio going through app_dial, there would
+ still be a period of invalid frames passing through. After
+ detecting failure here, Dial() exits. ABE-2658 ........
+ ................
+
+2010-11-23 10:30 +0000 [r295949] Olle Johansson <oej at edvina.net>
+
+ * /, main/say.c: Merged revisions 295907 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295907 | oej | 2010-11-23 10:36:38 +0100 (Tis,
+ 23 Nov 2010) | 14 lines Merged revisions 295906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8
+ lines Fix support of saynumber(1,n) in the Swedish language
+ (closes issue #18353) Reported by: oej Review:
+ https://reviewboard.asterisk.org/r/1031/ ........
+ ................
+
+2010-11-22 20:03 +0000 [r295869] Sean Bright <sean at malleable.com>
+
+ * configs/chan_dahdi.conf.sample, /: Merged revisions 295868 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, 22 Nov
+ 2010) | 2 lines Change some documentation to suggest
+ dahdi_monitor instead of ztmonitor. ........
+
+2010-11-22 19:36 +0000 [r295866] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_macro.c, include/asterisk/channel.h,
+ include/asterisk/frame.h, main/channel.c, main/pbx.c, /: Merged
+ revisions 295843 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600
+ (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010)
+ | 46 lines The channel redirect function (CLI or AMI) hangs up
+ the call instead of redirecting the call. To recreate the
+ problem: 1) Party A calls Party B 2) Invoke CLI "channel
+ redirect" command to redirect channel call leg associated with A.
+ 3) All associated channels are hung up. Note that if the CLI
+ command were done on the channel call leg associated with B it
+ works. This regression was a result of the fix for issue #16946
+ (https://reviewboard.asterisk.org/r/740/). The regression affects
+ all features that use an async goto to execute the dialplan
+ because of an external event: Channel redirect, AMI redirect, SIP
+ REFER, and FAX detection. The struct ast_channel._softhangup code
+ is a mess. The variable is used for several purposes that do not
+ necessarily result in the call being hung up. I have added
+ doxygen comments to describe how the various _softhangup bits are
+ used. I have corrected all the places where the variable was
+ tested in a non-bit oriented manner. The primary fix is the new
+ AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so
+ the soft hangup requests that do not normally result in a hangup
+ do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171)
+ Reported by: SantaFox (closes issue #18185) Reported by:
+ kwemheuer (closes issue #18211) Reported by: zahir_koradia
+ (closes issue #18230) Reported by: vmarrone (closes issue #18299)
+ Reported by: mbrevda (closes issue #18322) Reported by: nerbos
+ Review: https://reviewboard.asterisk.org/r/1013/ ........
+ ................
+
+2010-11-20 03:11 +0000 [r295747] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: One way audio before answering call
+ waiting call on analog port. * Analog call waiting Caller ID
+ spills could get stuck resulting in one way audio until the
+ waiting call is answered. This only happens on the second (and
+ later) call waiting call if the active call is not the first
+ call. * The CLI/AMI "dahdi show channel" command could report the
+ wrong channel information. Must keep the struct analog_pvt.owner
+ and struct dahdi_pvt.owner pointer in sync.
+
+2010-11-20 00:50 +0000 [r295711] Russell Bryant <russell at digium.com>
+
+ * main/event.c, include/asterisk/event.h, /: Merged revisions
+ 295710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010)
+ | 29 lines Fix cache of device state changes for multiple
+ servers. This patch addresses a regression where device states
+ across multiple servers were not being processing completely
+ correctly. The code works to determine the overall state by
+ looking at the last known state of a device on each server.
+ However, there was a regression due to some invasive rewrites of
+ how the cache works that led to the cache only storing the last
+ device state change for a device, regardless of which server it
+ was on. The code is set up to cache device state change events by
+ ensuring that each event in the cache has a unique device name +
+ entity ID (server ID). The code that was responsible for
+ comparing raw information elements (which EID is) always returned
+ a match due to a memcmp() with a length of 0. There isn't much
+ code to fix the actual bug. This patch also introduces a new CLI
+ command that was very useful for debugging this problem. The
+ command allows you to dump the contents of the event cache.
+ (closes issue #18284) Reported by: klaus3000 Patches:
+ issue18284.rev1.txt uploaded by russell (license 2) Tested by:
+ russell, klaus3000 (closes issue #18280) Reported by: klaus3000
+ Review: https://reviewboard.asterisk.org/r/1012/ ........
+
+2010-11-19 22:06 +0000 [r295673] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 295672 via svnmerge from
[... 26378 lines stripped ...]
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