[asterisk-commits] lmadsen: tag 10.0.0-beta1 r329324 - /tags/10.0.0-beta1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jul 22 14:46:54 CDT 2011
Author: lmadsen
Date: Fri Jul 22 14:46:50 2011
New Revision: 329324
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=329324
Log:
Importing files for 10.0.0-beta1 release.
Added:
tags/10.0.0-beta1/.lastclean (with props)
tags/10.0.0-beta1/.version (with props)
tags/10.0.0-beta1/ChangeLog (with props)
Added: tags/10.0.0-beta1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/10.0.0-beta1/.lastclean?view=auto&rev=329324
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+
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URL: http://svnview.digium.com/svn/asterisk/tags/10.0.0-beta1/.version?view=auto&rev=329324
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Added: tags/10.0.0-beta1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/10.0.0-beta1/ChangeLog?view=auto&rev=329324
==============================================================================
--- tags/10.0.0-beta1/ChangeLog (added)
+++ tags/10.0.0-beta1/ChangeLog Fri Jul 22 14:46:50 2011
@@ -1,0 +1,15582 @@
+2011-07-22 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 10.0.0-beta1 Released.
+
+2011-07-21 20:22 +0000 [r329257] Russell Bryant <russell at digium.com>
+
+ * channels/chan_dahdi.c, main/features.c,
+ include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c,
+ include/asterisk/rtp_engine.h: s/1.10/10.0/
+
+2011-07-21 18:05 +0000 [r329200-329204] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 329203 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011)
+ | 6 lines Document parkinglot in chan_dahdi.conf.sample. *
+ Document existing feature in chan_dahdi.conf.sample. * Remove
+ some dead code related to the parkinglot option. ........
+
+ * /, apps/app_directed_pickup.c: Merged revisions 329199 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011)
+ | 17 lines Update PickupChan documentation. The PickupChan uses
+ the ampersand as the argument separator. Was documented as:
+ PickupChan(channel[,channel2[,...][,options]]) Fixed
+ documentation to:
+ PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
+ This is a continuation of ASTERISK-17494 for v1.8 and later.
+ (closes issue ASTERISK-18144) Reported by: Erik Smith Patches:
+ pickupchan_ducumentation-v2.patch (License #6263) patch uploaded
+ by Erik Smith Tested by: Erik Smith ........
+
+2011-07-21 17:27 +0000 [r329188] Jason Parker <jparker at digium.com>
+
+ * UPGRADE.txt: Fix version number in UPGRADE.txt.
+
+2011-07-21 16:52 +0000 [r329145] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/features.c: Merged revisions 329144 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011)
+ | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed
+ more times than we've locked! This appears to be a leftover from
+ when ast_channel was converted to ao2 objects. Simply removed the
+ extraneous unlock. (closes issue ASTERISK-17772) ........
+
+2011-07-21 16:04 +0000 [r329104] Russell Bryant <russell at digium.com>
+
+ * / (added): Change Asterisk 2.0 to 2.0 in binary
+
+2011-07-20 21:31 +0000 [r329056] Paul Belanger <pabelanger at digium.com>
+
+ * /, UPGRADE-1.8.txt: Merged revisions 329055 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/2.0
+ ................ r329055 | pabelanger | 2011-07-20 17:27:50 -0400
+ (Wed, 20 Jul 2011) | 9 lines Merged revisions 329027 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed,
+ 20 Jul 2011) | 2 lines Asterisk now requires libpri 1.4.11+ for
+ PRI support. ........ ................
+
+2011-07-20 20:19 +0000 [r328996] Terry Wilson <twilson at digium.com>
+
+ * /, tests/test_netsock2.c: Merged revisions 328992 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/2.0
+ ................ r328992 | twilson | 2011-07-20 15:18:25 -0500
+ (Wed, 20 Jul 2011) | 12 lines Merged revisions 328987 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011)
+ | 5 lines We can't guarantee an eth0 is present FreeBSD test
+ fails on this case presumably because there is no eth0 on the
+ test machine. Better to just remove this test for now. ........
+ ................
+
+2011-07-20 19:03 +0000 [r328937] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 328936 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/2.0
+ ................ r328936 | kmoore | 2011-07-20 14:01:37 -0500
+ (Wed, 20 Jul 2011) | 15 lines Merged revisions 328935 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) |
+ 8 lines Inband DTMF regression The functionality of inband DTMF
+ in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not
+ working properly to avoid calling ast_rtp_instance_dtmf_begin/end
+ on RTP streams with inband DTMF. According to documentation,
+ ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
+ never inband. This fixes the regression introduced in revision
+ 328823. ........ ................
+
+2011-07-19 21:32 +0000 [r328880-328881] Kevin P. Fleming <kpfleming at digium.com>
+
+ * Makefile, /, Makefile.moddir_rules, sounds/Makefile: Merged
+ revisions 328879 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/2.0
+ ................ r328879 | kpfleming | 2011-07-19 16:31:16 -0500
+ (Tue, 19 Jul 2011) | 23 lines Merged revisions 328878 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul
+ 2011) | 17 lines Revert partial attempt at handling pathnames
+ with spaces. Revision 299794 attempted to improve the build
+ system to be able to handle pathnames (primarily DESTDIR) with
+ spaces in them, since this is common on some platforms (including
+ Mac OSX). Unfortunately, the changes were incomplete and did not
+ actually provide the desired behavior, and as a side effect the
+ functionality that ensured that stale headers in the Asterisk
+ 'include' directory were removed got broken. In addition, the
+ check for stale (and possibly incompatible) modules in the
+ Asterisk 'modules' directory also got broken, and would never
+ report any stale modules. Users upgrading to this version or
+ later versions would then see unexpected module load errors.
+ Since there are few users who actually want to install Asterisk
+ into paths that contain spaces, and a proper fix for the build
+ system would take many hours, the best solution for now is to
+ just revert the partial solution. ........ ................
+
+ * /: Edit the merge properties to match their names.
+
+2011-07-19 21:21 +0000 [r328877] Russell Bryant <russell at digium.com>
+
+ * /: Fix properties after twilson's change so merges still work
+
+2011-07-19 18:07 +0000 [r328772-328825] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_rtp_asterisk.c, main/rtp_engine.c, /,
+ channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged
+ revisions 328824 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328824 | kmoore | 2011-07-19 13:05:21 -0500
+ (Tue, 19 Jul 2011) | 18 lines Merged revisions 328823 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) |
+ 11 lines RTP bridge away with inband DTMF and feature detection
+ When deciding whether Asterisk was allowed to bridge the call
+ away from the core, chan_sip did not take into account the usage
+ of features on dialed channels that require monitoring of DTMF on
+ channels utilizing inband DTMF. This would cause Asterisk to
+ allow the call to be locally or remotely bridged, preventing
+ access to the data required to detect activations of such
+ features. (closes 17237) Review:
+ https://reviewboard.asterisk.org/r/1302/ ........
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 328771 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328771 | kmoore | 2011-07-19 10:46:54 -0500
+ (Tue, 19 Jul 2011) | 18 lines Merged revisions 328770 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) |
+ 11 lines MeetMe requests a PIN twice in some circumstances If a
+ call to MeetMe includes both the dynamic(D) and always request
+ PIN(P) options, MeetMe will ask for the PIN two times: once for
+ creating the conference and once for entering the conference.
+ This behavior was introduced in rev 311616 when adding the
+ CONFFLAG_ALWAYSPROMPT option to the logic branch controlling PIN
+ entry for joining a conference. (closes AST-601) Review:
+ https://reviewboard.asterisk.org/r/1305/ ........
+ ................
+
+2011-07-19 02:00 +0000 [r328718] Terry Wilson <twilson at digium.com>
+
+ * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c
+ (added): Merged revisions 328717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328717 | twilson | 2011-07-18 20:55:32 -0500
+ (Mon, 18 Jul 2011) | 14 lines Merged revisions 328716 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011)
+ | 7 lines Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't
+ modify the element passed in if it isn't found. This commit also
+ adds linked list unit tests. Review:
+ https://reviewboard.asterisk.org/r/1321/ ........
+ ................
+
+2011-07-18 20:51 +0000 [r328610-328665] Mark Murawki <markm at intellasoft.net>
+
+ * apps/app_dial.c, /: Merged revisions 328664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328664 | markm | 2011-07-18 16:50:13 -0400
+ (Mon, 18 Jul 2011) | 15 lines Merged revisions 328663 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) |
+ 9 lines app_dial may double free a channel datastore When
+ starting a call with originate, and having the callee channel run
+ Bridge() on pickup, we will double free the dialed_interface_info
+ datastore, causing a crash. Make sure to check if the datastore
+ still exists before trying to free it. (closes issue
+ ASTERISK-17917) Reported by: Mark Murawski Tested by: Mark
+ Murawski ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 328611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328611 | markm | 2011-07-18 08:56:49 -0400
+ (Mon, 18 Jul 2011) | 15 lines Merged revisions 328608 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) |
+ 9 lines If the sip private structure is null, sip_setoption()
+ will defref the null pointer and crash. Ideally, sip_setoption
+ shouldn't be called if there is a lack of a sip private
+ structure. But this will fix a crash. (closes issue
+ ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark
+ Murawski ........ ................
+
+ * /, main/asterisk.c: Merged revisions 328609 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328609 | markm | 2011-07-18 08:37:53 -0400
+ (Mon, 18 Jul 2011) | 15 lines Merged revisions 328593 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) |
+ 8 lines Fixed invalid read and null pointer deref on asterisk
+ shutdown. In some cases when starting asterisk with -c and
+ hitting control-c to shutdown, there will be an invalid read and
+ null pointer deref causing a crash. (closes issue ASTERISK-17927)
+ Reported by: Mark Murawski Tested by: Mark Murawski, Kinsey
+ Moore, Tilghman Lesher ........ ................
+
+2011-07-18 07:12 +0000 [r328542] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * /, funcs/func_odbc.c: Merged revisions 328541 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328541 | tilghman | 2011-07-18 02:11:26 -0500
+ (Mon, 18 Jul 2011) | 9 lines Merged revisions 328540 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18
+ Jul 2011) | 2 lines Typo ........ ................
+
+2011-07-15 21:41 +0000 [r328502] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c, /: Merged revisions
+ 328428-328429 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328428 | may | 2011-07-15 23:31:09 +0400 (Fri,
+ 15 Jul 2011) | 13 lines Merged revisions 328427 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7
+ lines small gk processing fixes: - decrease for 1 second
+ registration ttl for very low expirations (some providers expire
+ few earlier than TTL) - delete rrq and registration expire timers
+ on URQ received as we make new registration. ........
+ ................ r328429 | may | 2011-07-15 23:35:50 +0400 (Fri,
+ 15 Jul 2011) | 2 lines delete unproperly changed svn props
+ ................
+
+2011-07-15 21:19 +0000 [r328449-328459] Leif Madsen <lmadsen at digium.com>
+
+ * /, apps/app_macro.c: Merged revisions 328451 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10 ........
+ r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011)
+ | 1 line Build app_macro by default because things depend on it.
+ ........
+
+ * /, UPGRADE-1.10.txt, UPGRADE.txt, CHANGES: Merged revisions
+ 328448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10 ........
+ r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011)
+ | 2 lines Update UPGRADE.txt and CHANGES files. Update
+ documentation files stating that deprecated modules are no longer
+ built by default. ........
+
+2011-07-15 08:19 +0000 [r328381] Damien Wedhorn <voip at facts.com.au>
+
+ * channels/chan_skinny.c: Add SLA to skinny. Adds sublines to
+ skinny lines. Each subline can be attached to an SLA
+ station/trunk combo. Includes the following functionality: Callid
+ is persistent for both in/out calls on all skinny devices. Can
+ join, hold, resume. All sublines appear under a single line
+ button. See:
+ https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for
+ doc. (closes issue ASTERISK-17947) Review:
+ https://reviewboard.asterisk.org/r/1239/
+
+2011-07-15 00:23 +0000 [r328318-328344] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c,
+ include/asterisk/extconf.h, include/asterisk/pbx.h,
+ apps/app_queue.c: Merged revisions 328329 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10 ........
+ r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011)
+ | 2 lines Make hint watcher callback take const strings for
+ context and exten parameters. ........
+
+ * /, channels/chan_sip.c: Merged revisions 328317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328317 | rmudgett | 2011-07-14 18:28:49 -0500
+ (Thu, 14 Jul 2011) | 13 lines Merged revisions 328302 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011)
+ | 6 lines Missing SIP pvt and channel unlock in
+ sip_set_rtp_peer(). Regression introduced by -r326144. Add
+ missing SIP pvt and channel unlock in sip_set_rtp_peer().
+ ........ ................
+
+2011-07-14 20:28 +0000 [r328259] Leif Madsen <lmadsen at digium.com>
+
+ * funcs/func_speex.c, apps/app_playtones.c,
+ bridges/bridge_softmix.c, apps/app_alarmreceiver.c,
+ res/res_calendar_caldav.c, apps/app_ices.c, apps/app_exec.c,
+ channels/chan_iax2.c, res/res_pktccops.c, channels/chan_skinny.c,
+ pbx/pbx_ael.c, formats/format_h263.c, cdr/cdr_odbc.c,
+ cdr/cdr_manager.c, utils/refcounter.c, funcs/func_timeout.c,
+ formats/format_wav.c, apps/app_softhangup.c, codecs/codec_g726.c,
+ bridges/bridge_simple.c, funcs/func_cut.c, apps/app_talkdetect.c,
+ apps/app_db.c, funcs/func_callcompletion.c, funcs/func_channel.c,
+ funcs/func_iconv.c, pbx/pbx_config.c, res/res_odbc.c,
+ apps/app_voicemail.c, formats/format_sln.c,
+ apps/app_authenticate.c, apps/app_readexten.c,
+ res/res_phoneprov.c, apps/app_userevent.c, codecs/codec_gsm.c,
+ tests/test_func_file.c, apps/app_setcallerid.c,
+ res/res_config_odbc.c, funcs/func_audiohookinherit.c,
+ apps/app_osplookup.c, funcs/func_odbc.c, cel/cel_custom.c,
+ tests/test_utils.c, apps/app_mp3.c, res/res_timing_timerfd.c,
+ codecs/codec_resample.c, formats/format_h264.c,
+ apps/app_directory.c, formats/format_siren14.c,
+ tests/test_amihooks.c, res/res_config_pgsql.c,
+ funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c,
+ res/res_crypto.c, apps/app_originate.c, channels/chan_jingle.c,
+ apps/app_forkcdr.c, funcs/func_blacklist.c, apps/app_sms.c,
+ formats/format_g723.c, utils/extconf.c, tests/test_poll.c,
+ apps/app_stack.c, apps/app_verbose.c, utils/check_expr.c,
+ funcs/func_module.c, codecs/codec_adpcm.c, tests/test_event.c,
+ cdr/cdr_adaptive_odbc.c, apps/app_image.c,
+ formats/format_wav_gsm.c, utils/stereorize.c, pbx/pbx_loopback.c,
+ tests/test_time.c, funcs/func_shell.c, apps/app_skel.c,
+ channels/chan_alsa.c, apps/app_externalivr.c,
+ apps/app_milliwatt.c, formats/format_gsm.c, res/res_speech.c,
+ apps/app_dial.c, apps/app_page.c, apps/app_fax.c, utils/astman.c,
+ apps/app_disa.c, res/res_monitor.c, apps/app_waitforring.c,
+ addons/cdr_mysql.c, res/res_fax_spandsp.c,
+ res/res_timing_kqueue.c, apps/app_chanspy.c, apps/app_cdr.c,
+ channels/chan_unistim.c, funcs/func_base64.c,
+ channels/chan_multicast_rtp.c, funcs/func_md5.c,
+ apps/app_meetme.c, tests/test_gosub.c, funcs/func_sysinfo.c,
+ funcs/func_vmcount.c, res/res_musiconhold.c, cdr/cdr_radius.c,
+ apps/app_followme.c, res/res_config_sqlite.c,
+ apps/app_controlplayback.c, cdr/cdr_csv.c, formats/format_ilbc.c,
+ channels/chan_phone.c, funcs/func_enum.c, main/manager.c,
+ funcs/func_groupcount.c, tests/test_stringfields.c,
+ tests/test_locale.c, tests/test_devicestate.c,
+ funcs/func_frame_trace.c, funcs/func_aes.c, cdr/cdr_sqlite.c,
+ tests/test_astobj2.c, apps/app_ivrdemo.c, res/res_clioriginate.c,
+ apps/app_jack.c, apps/app_nbscat.c, res/res_calendar_icalendar.c,
+ codecs/codec_a_mu.c, tests/test_ast_format_str_reduce.c,
+ tests/test_dlinklists.c, res/res_convert.c, apps/app_waituntil.c,
+ pbx/pbx_lua.c, utils/astcanary.c, apps/app_queue.c,
+ channels/chan_oss.c, cdr/cdr_tds.c, channels/chan_usbradio.c,
+ apps/app_flash.c, apps/app_senddtmf.c, funcs/func_callerid.c,
+ addons/app_saycountpl.c, cel/cel_pgsql.c, apps/app_dahdibarge.c,
+ channels/chan_local.c, funcs/func_dialgroup.c,
+ tests/test_logger.c, apps/app_record.c, funcs/func_env.c,
+ funcs/func_strings.c, res/res_timing_dahdi.c,
+ apps/app_chanisavail.c, bridges/bridge_multiplexed.c,
+ res/res_rtp_multicast.c, cel/cel_odbc.c, channels/chan_dahdi.c,
+ pbx/pbx_spool.c, funcs/func_pitchshift.c, formats/format_pcm.c,
+ apps/app_dumpchan.c, main/http.c, res/res_clialiases.c,
+ res/res_calendar_exchange.c, res/res_ais.c, funcs/func_sprintf.c,
+ codecs/codec_g722.c, tests/test_expr.c, cel/cel_tds.c,
+ tests/test_app.c, utils/smsq.c, apps/app_morsecode.c,
+ formats/format_ogg_vorbis.c, tests/test_sched.c,
+ res/res_calendar_ews.c, apps/app_speech_utils.c,
+ tests/test_acl.c, apps/app_sendtext.c, funcs/func_cdr.c,
+ utils/hashtest2.c, utils/ael_main.c, apps/app_mixmonitor.c,
+ formats/format_g726.c, utils/streamplayer.c, res/res_calendar.c,
+ cel/cel_radius.c, channels/chan_vpb.cc, tests/test_heap.c,
+ addons/format_mp3.c, res/res_snmp.c, apps/app_dictate.c,
+ channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c,
+ res/res_jabber.c, funcs/func_uri.c, cel/cel_manager.c,
+ apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
+ codecs/codec_lpc10.c, apps/app_read.c, cdr/cdr_syslog.c,
+ codecs/codec_alaw.c, res/res_adsi.c, agi/eagi-test.c,
+ utils/conf2ael.c, tests/test_pbx.c, apps/app_channelredirect.c,
+ formats/format_vox.c, res/res_stun_monitor.c, tests/test_aoc.c,
+ pbx/pbx_dundi.c, funcs/func_devstate.c,
+ addons/res_config_mysql.c, funcs/func_rand.c,
+ apps/app_readfile.c, addons/chan_ooh323.c,
+ cdr/cdr_sqlite3_custom.c, /, apps/app_sayunixtime.c,
+ apps/app_test.c, res/res_http_post.c, res/res_smdi.c,
+ main/features.c, funcs/func_srv.c, apps/app_amd.c,
+ pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c,
+ formats/format_g719.c, channels/chan_bridge.c,
+ apps/app_privacy.c, apps/app_echo.c, codecs/codec_speex.c,
+ apps/app_saycounted.c, apps/app_dahdiras.c,
+ channels/chan_agent.c, funcs/func_math.c, res/res_ael_share.c,
+ apps/app_transfer.c, res/res_mutestream.c, apps/app_playback.c,
+ res/res_config_curl.c, channels/chan_misdn.c, funcs/func_curl.c,
+ tests/test_skel.c, apps/app_macro.c, apps/app_zapateller.c,
+ codecs/codec_ilbc.c, addons/app_mysql.c,
+ tests/test_substitution.c, utils/muted.c, utils/hashtest.c,
+ funcs/func_sha1.c, formats/format_siren7.c,
+ tests/test_security_events.c, funcs/func_config.c,
+ bridges/bridge_builtin_features.c, funcs/func_volume.c,
+ res/res_agi.c, apps/app_confbridge.c, addons/chan_mobile.c,
+ apps/app_parkandannounce.c, res/res_security_log.c,
+ cdr/cdr_custom.c, apps/app_while.c, res/res_rtp_asterisk.c,
+ funcs/func_dialplan.c, funcs/func_db.c, apps/app_festival.c,
+ res/res_limit.c, res/res_fax.c, apps/app_waitforsilence.c,
+ channels/chan_console.c, apps/app_getcpeid.c,
+ funcs/func_global.c, res/res_srtp.c, funcs/func_extstate.c,
+ tests/test_strings.c, res/res_timing_pthread.c,
+ apps/app_directed_pickup.c, channels/chan_h323.c,
+ cel/cel_sqlite3_custom.c, codecs/codec_ulaw.c,
+ channels/chan_nbs.c, formats/format_g729.c: Merged revisions
+ 328247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400
+ (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011)
+ | 6 lines Introduce <support_level> tags in MODULEINFO. This
+ change introduces MODULEINFO into many modules in Asterisk in
+ order to show the community support level for those modules. This
+ is used by changes committed to menuselect by Russell Bryant
+ recently (r917 in menuselect). More information about the support
+ level types and what they mean is available on the wiki at
+ https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
+ ........ ................
+
+2011-07-14 19:56 +0000 [r328208] Jonathan Rose <jrose at digium.com>
+
+ * /, res/res_monitor.c: Merged revisions 328207 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ................ r328207 | jrose | 2011-07-14 14:45:18 -0500
+ (Thu, 14 Jul 2011) | 13 lines Merged revisions 328205 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) |
+ 6 lines Monitor application arguments requirements fixed. Monitor
+ was requiring options in spite of no individual option on Monitor
+ being required. Review: https://reviewboard.asterisk.org/r/1320/
+ ........ ................
+
+2011-07-14 17:47 +0000 [r328163] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, main/dsp.c: Merged revisions 328162 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.10 ........
+ r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul
+ 2011) | 3 lines tune the v21 preamble detector to properly detect
+ the desired sequence of hits and misses ........
+
+2011-07-13 22:10 +0000 [r328121] David Vossel <dvossel at digium.com>
+
+ * /, apps/app_mixmonitor.c: Merged revisions 328120 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.10
+ ........ r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13
+ Jul 2011) | 15 lines Preserve sample rate quality of wideband
+ mixmonitor recordings. MixMonitor has the ability to record in
+ any file format Asterisk supports, but the quality of wideband
+ audio is not preserved. This is because regardless of the sample
+ rate the call is being recorded in, the audio is always
+ downsampled to 8khz and then upsampled to whatever wideband
+ format it is being written as. This patch resolves this by
+ requesting the audio from the audiohook in the signed linear
+ format closest to the sample rate of the format we are writing.
+ This fix is only possible for Asterisk 1.10 because audio hooks
+ in 1.8 are not capable of wideband audio. Review:
+ https://reviewboard.asterisk.org/r/1314/ ........
+
+2011-07-13 21:06 +0000 [r328079] Leif Madsen <lmadsen at digium.com>
+
+ * BUGS, UPGRADE-1.10.txt (added), UPGRADE.txt: Add UPGRADE-1.10.txt
+ file from UPGRADE.txt.
+
+2011-07-13 20:40 +0000 [r328075-328076] Russell Bryant <russell at digium.com>
+
+ * /: set 1.10 merge properties
+
+ * /: remove 1.8 merge properties
+
+2011-07-13 18:47 +0000 [r328016] Richard Mudgett <rmudgett at digium.com>
+
+ * /, configs/features.conf.sample: Merged revisions 328014 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r328014 | rmudgett | 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011)
+ | 1 line Add ATXFER_NULL_TECH note in features.conf.sample.
+ ........
+
+2011-07-12 23:02 +0000 [r327953] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/manager.c, /: Merged revisions 327950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul
+ 2011) | 14 lines Correct double-free situation in manager output
+ processing. The process_output() function calls ast_str_append()
+ and xml_translate() on its 'out' parameter, which is a pointer to
+ an ast_str buffer. If either of these functions need to
+ reallocate the ast_str so it will have more space, they will free
+ the existing buffer and allocate a new one, returning the address
+ of the new one. However, because process_output only receives a
+ pointer to the ast_str, not a pointer to its caller's variable
+ holding the pointer, if the original ast_str is freed, the caller
+ will not know, and will continue to use it (and later attempt to
+ free it). (reported by jkroon on #asterisk-dev) ........
+
+2011-07-12 20:08 +0000 [r327891] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, apps/app_directory.c: Merged revisions 327890 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r327890 | mnicholson | 2011-07-12 15:07:20 -0500 (Tue,
+ 12 Jul 2011) | 2 lines search in the current context for 'a' and
+ 'o' instead of 'default' ........
+
+2011-07-12 19:39 +0000 [r327889] Jason Parker <jparker at digium.com>
+
+ * Makefile, /: Merged revisions 327888 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327888 | qwell | 2011-07-12 14:38:44 -0500 (Tue, 12 Jul 2011) |
+ 1 line Fix uninstall target, so that modules dir gets cleared
+ again. ........
+
+2011-07-12 19:18 +0000 [r327856] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 327852 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12
+ Jul 2011) | 12 lines Added additional checks for mailbox /
+ password beginning with '*' character A bug existed such that if
+ a user entered a password with '*', and the extension 'a' did not
+ exist, an invalid mailbox would be created and the user
+ authenticated. The code was changed to prevent this from
+ occurring, and to prevent users from having mailboxes or
+ passwords defined that begin with the '*' character. (closes
+ issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/
+ ........
+
+2011-07-12 15:38 +0000 [r327794] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * tests/test_substitution.c, /: Merged revisions 327793 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011)
+ | 14 lines Use 'printf' (POSIX issue 4) instead of 'echo -n', for
+ portability. The problem with using 'echo -n' is that it is not
+ portable. While BSD systems required that the '-n' option be
+ removed and interpreted, System V required that all strings
+ should be echoed with no interpretation of options. This
+ fundamental difference of behavior means that it is never
+ possible to use the '-n' flag to echo in tests which are meant to
+ be portable. In this case, on Mac OS X 10.6, the /bin/sh shell
+ builtin 'echo' uses the System V semantics of the command, and
+ thus the SHELL test failed on that platform.
+ http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16
+ ........
+
+2011-07-12 15:23 +0000 [r327769] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_fax.c, include/asterisk/dsp.h, main/dsp.c: do v21
+ detection instead of CED detection for the fax gateway
+
+2011-07-12 14:55 +0000 [r327749] David Vossel <dvossel at digium.com>
+
+ * main/bridging.c: Send video update frame to new video source in
+ follow_talker correctly.
+
+2011-07-12 14:40 +0000 [r327748] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_confbridge.c: Segfault on shutdown when confbridge is
+ active When undergoing a shutdown and channels are kicked out of
+ a bridge, a segfault occurs because ConfBridge tries to play
+ sounds on the bridge after the underlying channels have been
+ blown away due to the shutdown. (closes ASTERISK-18040) Review:
+ https://reviewboard.asterisk.org/r/1283/
+
+2011-07-11 20:06 +0000 [r327684] Matthew Nicholson <mnicholson at digium.com>
+
+ * tests/test_substitution.c: use printf instead of echo -n in
+ substitution test
+
+2011-07-11 19:49 +0000 [r327683] Terry Wilson <twilson at digium.com>
+
+ * /, include/asterisk/jingle.h, channels/chan_gtalk.c: Merged
+ revisions 327682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011)
+ | 9 lines Update chan_gtalk to work with changed GMail-based
+ calls The messages sent by the GMail client have changed, but
+ include the old-style messages as well. This patch checks for
+ this case and uses the old-style offer. (closes issue
+ ASTERISK-18084) Review: https://reviewboard.asterisk.org/r/1312/
+ ........
+
+2011-07-11 18:44 +0000 [r327640] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/bridging.h, bridges/bridge_softmix.c,
+ main/bridging.c: Updates follow_talker video_mode in confbridge
+ application. follow_talker mode originally echoed the same video
+ stream to all participants. As the primary talker switched
+ around, the video stream would result in the talker seeing
+ themselves. Now the primary talker sees the last person who was
+ talking rather than themselves.
+
+2011-07-11 17:23 +0000 [r327469-327598] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_fax.c: renamed fax_gateway_send_ced() to
+ fax_gateway_request_t38()
+
+ * res/res_fax.c: actually do something with the ced timeout, also
+ added more debug output
+
+ * res/res_fax.c: write silence on the channel during t.38
+ negotiation
+
+ * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327512
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327512 | mnicholson | 2011-07-11 08:53:59 -0500 (Mon, 11 Jul
+ 2011) | 2 lines reset our buffer each iteration when doing
+ variable substitution ........
+
+ * res/res_fax.c: Delay sending an CED tone generated T.38 reinvite
+ to give the CED tone generating party time to send its own T.38
+ reinvite. Also don't forward frames through the gateway if we are
+ negotiating T.38.
+
+ * res/res_fax.c: fixed wording in a comment
+
+2011-07-11 10:57 +0000 [r327413] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * /, main/Makefile: Merged revisions 327411 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327411 | tzafrir | 2011-07-11 13:46:34 +0300 (ב', 11 יול 2011) |
+ 5 lines fix building the Debian armhf (HardFloat) port Fixes
+ http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385
+ (Missing pthreads) ........
+
+2011-07-10 01:37 +0000 [r327359] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c, configs/chan_ooh323.conf.sample: Full T.38
+ handshaking and fax detection Add full t.38 handshaking for
+ OOH323 that are required for newest T.38 gateway codes. Add fax
+ detection (cng tone, t38) and dialplan redirection to fax ext on
+ fax event detected. Add OOH323() function to set/get t38support
+ and faxdetect parameters. (closes issue ASTERISK-17754) Reported
+ by: irroot Patches: ooh323_faxdetect.patch uploaded by irroot
+ (license 52) issue19183-final.patch uploaded by may213 (license
+ 454) Tested by: may213, irroot Review:
+ https://reviewboard.asterisk.org/r/1174/
+
+2011-07-08 22:25 +0000 [r327246] Jason Parker <jparker at digium.com>
+
+ * main/stdtime, utils, codecs, utils/db1-ast/recno, apps, cel,
+ apps/confbridge, cdr, formats, codecs/gsm/src,
+ utils/db1-ast/hash, funcs, bridges, codecs/lpc10,
+ utils/db1-ast/db, codecs/g722, utils/db1-ast/mpool, main,
+ codecs/speex, channels/sip, pbx, res, res/ael, channels,
+ utils/db1-ast/btree: Add .o files to svn:ignore property, since
+ it's only ignored if locally configured to do so.
+
+2011-07-08 21:43 +0000 [r327212] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 327211 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011)
+ | 9 lines INVITE 403 Forbidden response always retransmits the
+ maximum times. Asterisk sends a 403 Forbidden response if
+ authentication fails for an INVITE as required. However, it
+ ignores the ACK and keeps retransmitting the response. * Made not
+ delete the to-tag in the dialog so the expected ACK can be
+ matched with the dialog and stop the retransmissions. ........
+
+2011-07-08 20:33 +0000 [r327116-327168] David Vossel <dvossel at digium.com>
+
+ * UPGRADE.txt, CHANGES: Adds entry in UPDATES.txt for removal of
+ formats/format_sln16.c. Fixes typo in CHANGES as well.
+
+ * CHANGES: Updates CHANGES log to reflect new slinear read/write
+ file interpreters.
+
+ * formats/format_sln.c, formats/format_sln16.c (removed): Support
+ for writing and reading raw slin files 8khz-192khz.
+
+ * formats/format_attr_silk.c (removed), formats/format_attr_celt.c
+ (removed), res/res_format_attr_silk.c (added),
+ res/res_format_attr_celt.c (added): Moves celt and silk format
+ attribute files into res folder. It was inconsistent to have the
+ silk and celt format attribute modules in the format file
+ interpreter folder.
+
+2011-07-08 19:54 +0000 [r327107] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327106
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul
+ 2011) | 11 lines Reset our ast_str before passing it on to
+ dialplan function backends. It is possible for a dialplan backend
+ to not modify the given buffer or ast_str and still return
+ success. This causes any previous value stored in the buffer to
+ be used as if the new function call provided it. Some functions
+ also append to the given buffer assuming it is empty. The
+ test_substitution unit test has also been modified to detect this
+ problem. (closes issue ASTERISK-17878) ........
+
+2011-07-08 16:00 +0000 [r327045-327047] Russell Bryant <russell at digium.com>
+
+ * /, tests/test_netsock2.c: Merged revisions 327046 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r327046 | russell | 2011-07-08 11:00:05 -0500 (Fri, 08
+ Jul 2011) | 2 lines Fix an error and add more log message info to
+ help see why this fails on FreeBSD. ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 327044 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r327044 | russell | 2011-07-08 10:28:44 -0500 (Fri, 08
+ Jul 2011) | 2 lines Resolve some set-but-unused-variable
+ warnings. ........
+
+2011-07-08 01:26 +0000 [r327000] Richard Mudgett <rmudgett at digium.com>
+
+ * main/pbx.c, /: Merged revisions 326985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011)
+ | 12 lines Some code cleanup in pbx.c * Mostly comment and format
+ changes. * ast_context_remove_extension_callerid() and
+ ast_add_extension_nolock() will write lock the found specific
[... 14866 lines stripped ...]
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