[asterisk-commits] bebuild: tag certified-1.8.28-cert1 r422739 - /certified/tags/1.8.28-cert1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Sep 5 19:23:12 CDT 2014
Author: bebuild
Date: Fri Sep 5 19:23:10 2014
New Revision: 422739
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=422739
Log:
Importing files for 1.8.28-cert1 release.
Modified:
certified/tags/1.8.28-cert1/.version
certified/tags/1.8.28-cert1/ChangeLog
Modified: certified/tags/1.8.28-cert1/.version
URL: http://svnview.digium.com/svn/asterisk/certified/tags/1.8.28-cert1/.version?view=diff&rev=422739&r1=422738&r2=422739
==============================================================================
--- certified/tags/1.8.28-cert1/.version (original)
+++ certified/tags/1.8.28-cert1/.version Fri Sep 5 19:23:10 2014
@@ -1,1 +1,1 @@
-1.8.28.0
+1.8.28-cert1
Modified: certified/tags/1.8.28-cert1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/certified/tags/1.8.28-cert1/ChangeLog?view=diff&rev=422739&r1=422738&r2=422739
==============================================================================
--- certified/tags/1.8.28-cert1/ChangeLog (original)
+++ certified/tags/1.8.28-cert1/ChangeLog Fri Sep 5 19:23:10 2014
@@ -1,3 +1,164 @@
+2014-09-05 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.28-cert1 Released.
+
+2014-09-05 17:37 +0000 [r422662] Kinsey Moore <kmoore at digium.com>
+
+ * funcs/func_presence_state.c: Add missing support level to
+ func_presence_state
+
+2014-09-04 14:41 +0000 [r422582] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_vpb.cc: Set chan_vpb to be disabled by default
+
+2014-08-15 19:39 +0000 [r421208] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_voicemail.c, apps/app_meetme.c,
+ channels/sip/reqresp_parser.c, main/test.c: Fix build in
+ dev/TEST_FRAMEWORK mode
+
+2014-08-14 17:29 +0000 [r421032-421033] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * /, main/config.c: config: inform config hook of change when
+ writing file When updated configuration is written back to the
+ conf file - for example when a user changes their voicemail pin,
+ make sure that any config hook that wants to know of changes is
+ informed. Review: https://reviewboard.asterisk.org/r/3708/
+
+ * apps/app_voicemail.c: app_voicemail: use a consistent generator
+ string When updating voicemail.conf when a user changes their
+ pin, change the generator string to be the same as the module
+ name when reading so that the same config_hook will be called.
+ Review: https://reviewboard.asterisk.org/r/3837/
+
+2014-08-08 17:29 +0000 [r420560] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Replace sip_tls_read() and
+ resolve the large SDP poll issue. Replace sip_tls_read() and
+ sip_tcp_read() with a single function and resolve the poll/wait
+ issue with large SDP payloads. ASTERISK-18345 #close Reported by:
+ Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835)
+ patch uploaded by Elazar Broad Review:
+ https://reviewboard.asterisk.org/r/3882/ ........ Merged
+ revisions 420434 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-25 23:47 +0000 [r419678] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/features.c, main/app.c, apps/app_stack.c,
+ include/asterisk/app.h: features.c: Allow appliationmap to use
+ Gosub. Using DYNAMIC_FEATURES with a Gosub application as the
+ mapped application does not work. It does not work because Gosub
+ just pushes the current dialplan context, exten, and priority
+ onto a stack and sets the specified Gosub location. Gosub does
+ not have a dialplan execution loop to run dialplan like Macro. *
+ Made the DYNAMIC_FEATURES application mapping feature call
+ ast_app_exec_macro() and ast_app_exec_sub() for the Macro and
+ Gosub applications respectively. * Backported
+ ast_app_exec_macro() and ast_app_exec_sub() from v11 to execute
+ dialplan routines from the DYNAMIC_FEATURES application mapping
+ feature. NOTE: This issue does not affect v12+ because it already
+ does what this patch implements. AST-1391 #close Reported by:
+ Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/3844/ ........ Merged
+ revisions 419630 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-13 05:25 +0000 [r415975-416095] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/tcptls.h, main/tcptls.c, main/manager.c, /,
+ channels/chan_sip.c, main/http.c: AST-2014-007: Fix of fix to
+ allow AMI and SIP TCP to send messages. ASTERISK-23673 #close
+ Reported by: Richard Mudgett Review:
+ https://reviewboard.asterisk.org/r/3617/ ........ Merged
+ revisions 416066 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configs/http.conf.sample, include/asterisk/utils.h,
+ main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
+ main/http.c, UPGRADE.txt, main/utils.c,
+ include/asterisk/tcptls.h: AST-2014-007: Fix DOS by consuming the
+ number of allowed HTTP connections. Simply establishing a TCP
+ connection and never sending anything to the configured HTTP port
+ in http.conf will tie up a HTTP connection. Since there is a
+ maximum number of open HTTP sessions allowed at a time you can
+ block legitimate connections. A similar problem exists if a HTTP
+ request is started but never finished. * Added http.conf
+ session_inactivity timer option to close HTTP connections that
+ aren't doing anything. Defaults to 30000 ms. * Removed the
+ undocumented manager.conf block-sockets option. It interferes
+ with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections
+ now have better authentication timeout protection. Though I
+ didn't remove the bizzare TLS timeout polling code from chan_sip.
+ * chan_sip can now handle SSL certificate renegotiations in the
+ middle of a session. It couldn't do that before because the
+ socket was non-blocking and the SSL calls were not restarted as
+ documented by the OpenSSL documentation. * Fixed an off nominal
+ leak of the ssl struct in handle_tcptls_connection() if the FILE
+ stream failed to open and the SSL certificate negotiations
+ failed. The patch creates a custom FILE stream handler to give
+ the created FILE streams inactivity timeout and timeout after a
+ specific moment in time capability. This approach eliminates the
+ need for code using the FILE stream to be redesigned to deal with
+ the timeouts. This patch indirectly fixes most of ASTERISK-18345
+ by fixing the usage of the SSL_read/SSL_write operations.
+ ASTERISK-23673 #close Reported by: Richard Mudgett ........
+ Merged revisions 415841 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-05 19:17 +0000 [r415246-415290] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_ices.c, channels/chan_gtalk.c, cdr/cdr_csv.c,
+ channels/chan_jingle.c, channels/chan_phone.c,
+ funcs/func_pitchshift.c, include/asterisk/custom_control_frame.h
+ (added), apps/app_minivm.c, main/features.c,
+ apps/app_confbridge.c, apps/app_sms.c, configs/sip.conf.sample,
+ include/asterisk/pbx.h, res/res_config_ldap.c,
+ apps/app_voicemail.c, cel/cel_radius.c, apps/app_nbscat.c,
+ apps/app_waitforsilence.c, include/asterisk/config.h,
+ apps/app_dahdiras.c, pbx/pbx_lua.c, res/res_ael_share.c,
+ apps/app_chanisavail.c, cdr/cdr_radius.c, res/res_jabber.c,
+ main/presencestate.c (added), cdr/cdr_tds.c,
+ apps/app_osplookup.c, channels/chan_skinny.c,
+ funcs/func_frame_trace.c, cdr/cdr_sqlite.c, apps/app_amd.c,
+ pbx/pbx_realtime.c, apps/app_url.c, apps/app_externalivr.c,
+ tests/test_config.c, cdr/cdr_odbc.c, include/asterisk/frame.h,
+ res/res_timing_kqueue.c, main/custom_control_frame.c (added),
+ channels/sip/include/sip.h, main/asterisk.c,
+ channels/chan_mgcp.c, tests/test_custom_control.c (added),
+ channels/chan_unistim.c, apps/app_dahdibarge.c, main/pbx.c,
+ res/res_phoneprov.c, include/asterisk/channel.h, cdr/cdr_pgsql.c,
+ include/asterisk/manager.h, apps/app_queue.c,
+ res/res_config_sqlite.c, main/config.c,
+ include/asterisk/callerid.h, include/asterisk/file.h,
+ include/asterisk/app.h, apps/app_waitforring.c,
+ include/asterisk/event_defs.h, formats/format_vox.c, configure,
+ configs/jabber.conf.sample, res/res_timing_pthread.c,
+ include/asterisk/message.h (added), pbx/pbx_ael.c,
+ channels/chan_h323.c, cel/cel_sqlite3_custom.c, main/event.c,
+ formats/format_jpeg.c, apps/app_adsiprog.c, apps/app_jack.c,
+ res/res_ais.c, cdr/cdr_sqlite3_custom.c, res/res_snmp.c,
+ channels/chan_sip.c, apps/app_dictate.c, apps/app_festival.c,
+ cel/cel_tds.c, apps/app_alarmreceiver.c,
+ configs/manager.conf.sample, apps/app_image.c,
+ channels/chan_console.c, include/asterisk/_private.h,
+ apps/app_getcpeid.c, apps/app_talkdetect.c, channels/chan_iax2.c,
+ channels/chan_oss.c, main/channel.c, funcs/func_presence_state.c
+ (added), main/manager.c, apps/app_setcallerid.c,
+ channels/chan_misdn.c, tests/test_voicemail_api.c (added),
+ apps/app_mp3.c, include/asterisk/jabber.h, main/file.c,
+ main/callerid.c, channels/chan_alsa.c, main/app.c,
+ pbx/pbx_dundi.c, channels/chan_nbs.c, apps/app_zapateller.c,
+ main/message.c (added), apps/app_mixmonitor.c,
+ res/res_fax_spandsp.c, cel/cel_pgsql.c, res/res_config_pgsql.c,
+ apps/app_readfile.c, /, apps/app_test.c,
+ include/asterisk/presencestate.h (added), apps/app_morsecode.c:
+ Merge changes for Digium phone support, and default module
+ building. All of these changes were merged from
+ certified/branches/1.8.15/
+
+ * / (added): Create branch for Certified Asterisk 1.8.28
+
2014-05-29 Asterisk Development Team <asteriskteam at digium.com>
* Asterisk 1.8.28.0 Released.
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