Importing release summary for 11.6-cert7 release.

git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/tags/11.6-cert7@426069 65c4cc65-6c06-0410-ace0-fbb531ad65f3

.version

@@ -1 +1 @@
-11.6.0
+11.6-cert7

ChangeLog

@@ -1,3 +1,704 @@
+2014-10-20  Asterisk Development Team <asteriskteam@digium.com>
+
+	* Certified Asterisk 11.6-cert7 Released.
+
+	* AST-2014-011: Fix POODLE security issues
+
+	  There are two aspects to the vulnerability:
+	  (1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module
+	      to use TLSv1+. At this time, it does not refactor res_jabber/
+	      res_xmpp to use the TCP/TLS core, which should be done as an
+	      improvement at a latter date.
+	  (2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left
+	      unspecified, will default to the OpenSSL SSLv23_method. This
+	      method allows for all encryption methods, including SSLv2/SSLv3.
+	      A MITM can exploit this by forcing a fallback to SSLv3, which
+	      leaves the server vulnerable to POODLE. This patch adds WARNINGS
+	      if a user uses SSLv2/SSLv3 in their configuration, and explicitly
+	      disables SSLv2/SSLv3 if using SSLv23_method.
+
+	  For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or
+	  SSLv3 is explicitly chosen. For TLS servers, Asterisk will no longer
+	  support SSLv2 or SSLv3.
+
+	  Much thanks to abelbeck for reporting the vulnerability and providing
+	  a patch for the res_jabber/res_xmpp modules.
+
+2014-09-18  Asterisk Development Team <asteriskteam@digium.com>
+
+	* Certified Asterisk 11.6-cert6 Released.
+
+	* AST-2014-010: Resolve crash when the Message channel technology
+	  enters into the ReceiveFax application using res_fax_spandsp
+
+	  If faxing fails at a very early stage, then it is possible for
+	  us to pass a NULL t30 state pointer to spandsp, which spandsp
+	  is none too pleased with.
+
+	  This patch ensures that we pass the correct pointer to spandsp
+	  in the situation where we have not yet set our local t30 state
+	  pointer.
+
+	  An advisory was made for this issue due to the likelihood of
+	  it occurring in some Asterisk configurations.
+
+	  ASTERISK-24301 #close
+	  Reported by Matt Jordan, Philippe Lindheimer
+
+2014-09-05  Asterisk Development Team <asteriskteam@digium.com>
+
+	* Certified Asterisk 11.6-cert5 Released.
+
+2014-08-17 01:54 +0000 [r421209]  Kinsey Moore <kmoore@digium.com>
+
+	* res/res_snmp.c, apps/app_dictate.c, apps/app_test.c,
+	  apps/app_ices.c, res/res_http_websocket.c, cdr/cdr_radius.c,
+	  build_tools/cflags.xml, funcs/func_pitchshift.c,
+	  apps/app_osplookup.c, funcs/func_frame_trace.c,
+	  channels/console_gui.c, apps/app_mp3.c, pbx/pbx_ael.c,
+	  channels/console_board.c, formats/format_jpeg.c,
+	  channels/chan_mgcp.c, res/res_config_pgsql.c, cel/cel_tds.c,
+	  apps/app_dahdiras.c, res/res_ael_share.c, apps/app_talkdetect.c,
+	  utils/conf2ael.c, channels/chan_jingle.c, channels/chan_misdn.c,
+	  formats/format_vox.c, res/res_timing_pthread.c,
+	  res/res_corosync.c, cel/cel_sqlite3_custom.c, apps/app_sms.c,
+	  apps/app_zapateller.c, res/res_fax_spandsp.c,
+	  res/res_timing_kqueue.c, utils/check_expr.c,
+	  channels/chan_unistim.c, build_tools/cflags-devmode.xml,
+	  utils/muted.c, cdr/cdr_sqlite3_custom.c, res/res_phoneprov.c,
+	  channels/console_video.c, apps/app_alarmreceiver.c,
+	  apps/app_chanisavail.c, apps/app_image.c, channels/chan_gtalk.c,
+	  cdr/cdr_pgsql.c, res/res_config_sqlite.c, res/res_pktccops.c,
+	  cdr/cdr_csv.c, utils/stereorize.c, channels/chan_phone.c,
+	  channels/chan_skinny.c, build_tools/embed_modules.xml,
+	  apps/app_minivm.c, pbx/pbx_realtime.c, apps/app_amd.c,
+	  channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c,
+	  cdr/cdr_odbc.c, res/res_config_ldap.c, apps/app_jack.c,
+	  apps/app_adsiprog.c, utils/refcounter.c, apps/app_nbscat.c,
+	  apps/app_festival.c, apps/app_waitforsilence.c, utils/astman.c,
+	  apps/app_morsecode.c, utils/smsq.c, pbx/pbx_lua.c,
+	  channels/chan_console.c, apps/app_getcpeid.c,
+	  channels/chan_oss.c, cdr/cdr_tds.c, apps/app_waitforring.c,
+	  pbx/pbx_dundi.c, utils/ael_main.c, utils/extconf.c,
+	  channels/chan_nbs.c, utils/streamplayer.c, cel/cel_pgsql.c,
+	  cel/cel_radius.c: Add missing commit from 11.2-cert This disables
+	  building by default for all extended modules for Certified
+	  Asterisk 11.6. This commit was missed from 11.2-cert when
+	  creating the 11.6-cert branch. ASTERISK-24104 #close Reported by:
+	  Rusty Newton
+
+2014-08-08 17:18 +0000 [r420559]  Richard Mudgett <rmudgett@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 ........ Merged
+	  revisions 420435 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-25 23:27 +0000 [r419662]  Richard Mudgett <rmudgett@digium.com>
+
+	* main/features.c, /: 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 ........ Merged
+	  revisions 419631 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-23 14:34 +0000 [r419308]  Scott Griepentrog <sgriepentrog@digium.com>
+
+	* /, 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/ ........
+	  Merged revisions 419284 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-11 16:39 +0000 [r418368]  Scott Griepentrog <sgriepentrog@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/
+	  ........ Merged revisions 418366 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-07-01 15:37 +0000 [r417724]  Joshua Colp <jcolp@digium.com>
+
+	* res/res_rtp_asterisk.c, main/rtp_engine.c, /,
+	  channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample,
+	  include/asterisk/rtp_engine.h, channels/sip/include/sip.h:
+	  Multiple revisions
+	  402345,405234,409129-409130,409565,413008,417141,417677 ........
+	  r402345 | kmoore | 2013-11-01 05:31:49 -0700 (Fri, 01 Nov 2013) |
+	  11 lines chan_sip: Fix RTCP port for SRFLX ICE candidates This
+	  corrects one-way audio between Asterisk and Chrome/jssip as a
+	  result of Asterisk inserting the incorrect RTCP port into RTCP
+	  SRFLX ICE candidates. This also exposes an ICE component
+	  enumeration to extract further details from candidates. (closes
+	  issue ASTERISK-21383) Reported by: Shaun Clark Review:
+	  https://reviewboard.asterisk.org/r/2967/ ........ r405234 |
+	  kharwell | 2014-01-09 08:49:55 -0800 (Thu, 09 Jan 2014) | 19
+	  lines res_rtp_asterisk: Fails to resume WebRTC call from hold In
+	  ast_rtp_ice_start if the ice session create check list failed,
+	  start check was never initiated and ice_started was never set to
+	  true. Upon re-entering the function (for instance, [un]hold) it
+	  would try to create the check list again with duplicate remote
+	  candidates. Fixed so that if the create check list fails the
+	  necessary data structures are properly re-initialized for any
+	  subsequent retries. Note, it was decided to not stop ice support
+	  (by calling ast_rtp_ice_stop) on a check list failure because it
+	  possible things might still work. However, a debug message was
+	  added to help with any future troubleshooting. (closes issue
+	  ASTERISK-22911) Reported by: Vytis Valentinavičius Patches:
+	  works_on_my_machine.patch uploaded by xytis (license 6558)
+	  ........ r409129 | jrose | 2014-02-27 11:19:02 -0800 (Thu, 27 Feb
+	  2014) | 15 lines res_rtp_asterisk: Fix checklist creating
+	  problems in ICE sessions Prior to this patch, local candidate
+	  lists including SRFLX would fail to start properly when building
+	  ICE candidate check lists. This patch fixes that problem by
+	  making sure that each SRFLX candidate is associated with the
+	  proper base address so that the check list can create matches
+	  properly. This patch was written by jcolp. The issue will be left
+	  open to await testing by the issue participants. (issue
+	  ASTERISK-23213) Reported by: Andrea Suisani Review:
+	  https://reviewboard.asterisk.org/r/3256/ ........ r409130 | jrose
+	  | 2014-02-27 11:38:10 -0800 (Thu, 27 Feb 2014) | 8 lines
+	  res_rtp_asterisk: correct build error from r409129 Accidentally
+	  placed a declaration below functional code (issue ASTERISK-23213)
+	  Reported by: Andrea Suisani Review:
+	  https://reviewboard.asterisk.org/r/3256/ ........ r409565 | jrose
+	  | 2014-03-04 08:40:39 -0800 (Tue, 04 Mar 2014) | 9 lines
+	  res_rtp_asterisk: Fix one way audio problems with hold/unhold
+	  when using ICE ICE sessions will now be restarted if sessions are
+	  changed to use new sets of remote candidates. (closes issue
+	  ASTERISK-22911) Reported by: Vytis Valentinavičius Review:
+	  https://reviewboard.asterisk.org/r/3275/ ........ r413008 |
+	  mjordan | 2014-04-25 10:47:21 -0700 (Fri, 25 Apr 2014) | 14 lines
+	  res_rtp_asterisk: Add support for DTLS handshake retransmissions
+	  On congested networks, it is possible for the DTLS handshake
+	  messages to get lost. This patch adds a timer to res_rtp_asterisk
+	  that will periodically check to see if the handshake has
+	  succeeded. If not, it will retransmit the DTLS handshake. Review:
+	  https://reviewboard.asterisk.org/r/3337 ASTERISK-23649 #close
+	  Reported by: Nitesh Bansal patches: dtls_retransmission.patch
+	  uploaded by Nitesh Bansal (License 6418) ........ r417141 | file
+	  | 2014-06-23 11:49:14 -0700 (Mon, 23 Jun 2014) | 5 lines
+	  res_rtp_asterisk: Return the length of data written when sending
+	  via ICE instead of 0. ASTERISK-23834 #close Reported by: Richard
+	  Kenner ........ r417677 | file | 2014-06-30 12:42:18 -0700 (Mon,
+	  30 Jun 2014) | 12 lines res_rtp_asterisk: Add SHA-256 support for
+	  DTLS and perform DTLS negotiation on RTCP. This change fixes up
+	  DTLS support in res_rtp_asterisk so it can accept and provide a
+	  SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after
+	  ICE negotiation completes. Configuration options to chan_sip have
+	  also been added to allow behavior to be tweaked (such as forcing
+	  the AVP type media transports in SDP). ASTERISK-22961 #close
+	  Reported by: Jay Jideliov Review:
+	  https://reviewboard.asterisk.org/r/3679/ ........ Merged
+	  revisions 402345,405234,409129-409130,409565,413008,417141,417677
+	  from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-13 05:29 +0000 [r415977-416106]  Richard Mudgett <rmudgett@digium.com>
+
+	* main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
+	  main/http.c, include/asterisk/tcptls.h: 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 ........ Merged
+	  revisions 416067 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* main/http.c, UPGRADE.txt, main/utils.c,
+	  include/asterisk/tcptls.h, res/res_http_websocket.c,
+	  configs/http.conf.sample, include/asterisk/utils.h,
+	  main/tcptls.c, main/manager.c, /, channels/chan_sip.c:
+	  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 ........ Merged
+	  revisions 415854 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-06-12 16:27 +0000 [r415867]  Scott Griepentrog <sgriepentrog@digium.com>
+
+	* /, apps/app_queue.c: app_queue: delayed state can cause early
+	  leavewhenempty ringing In app_queue, device state changes arrive
+	  in event messages and update the queue member status value. That
+	  value is checked in get_member_status() to decide that the caller
+	  should leave when there are no available members. Although event
+	  messages can be delayed by other activity, there is no adverse
+	  affect by lagged status except in one specific case: there is
+	  only one available member, it was just rung, and leavewhenempty
+	  is enabled set for ringing members. This change adds a direct
+	  check of the device state only under this condition where the
+	  caller may be dropped incorrectly, resolving this issue without
+	  affecting performance of app_queue normally. AST-1248 #close
+	  Review: https://reviewboard.asterisk.org/r/3595/ Reported by:
+	  Thomas Arimont ........ Merged revisions 415833 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-12 16:06 +0000 [r415842]  Jonathan Rose <jrose@digium.com>
+
+	* /, UPGRADE.txt, apps/app_mixmonitor.c: MixMonitor: Add class
+	  authorization requirements to MixMonitor AMI commands MixMonitor
+	  AMI commands StartMixMonitor and StopMixMonitor lacked class
+	  authorization. StopMixMonitor now requires that the manager user
+	  either have the call or system class authorization.
+	  StartMixMonitor is a slightly larger issue since it can execute
+	  shell commands if the right arguments are passed into it, and we
+	  consider this a permission escalation. A security release will be
+	  issued for problem this shortly. ASTERISK-23609 #close Reported
+	  by: Corey Farrell ........ Merged revisions 415837 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-05-13 00:48 +0000 [r413773]  Richard Mudgett <rmudgett@digium.com>
+
+	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  channels/sig_pri.c: chan_dahdi/sig_pri: Prevent unnecessary
+	  PROGRESS events when overlap dialing is enabled. When overlap
+	  dialing is enabled, the lack of inband audio available
+	  information in the SETUP_ACKNOWLEDGE events causes an
+	  interoperability problem with SIP. sig_pri doesn't know if there
+	  is dialtone present when a SETUP_ACKNOWLEDGE is received so it
+	  assumes it is there and posts an AST_CONTROL_PROGRESS frame. The
+	  SIP channel driver then sends out a 183 Session Progress and
+	  blocks the desired 180 Ringing message when the ALERTING message
+	  comes in. * Made the configure script detect if the installed
+	  version of libpri supports the SETUP_ACKNOWLEDGE enhancements. *
+	  Using the new API, made generate an AST_CONTROL_PROGRESS frame on
+	  an incoming SETUP_ACKNOWLEDGE message when the message indicates
+	  inband audio is present instead of assuming that dialtone is
+	  present. * Using the new API, made SETUP_ACKNOWLEDGE send out an
+	  inband audio available indication only if dialtone is expected.
+	  The change also makes the fallback behaviour of sending the
+	  PROGRESS message better by sending it only if dialtone is
+	  expected. * Changed receiving a PROCEEDING message to not
+	  generate an AST_CONTROL_PROGRESS frame if the progress indication
+	  ie indicates non-end-to-end-ISDN. This helps interoperability
+	  with SIP. * Changed sending a PROCEEDING message in response to
+	  an AST_CONTROL_PROCEEDING frame to not indicate inband audio
+	  available. It was silly to do so anyway because the channel
+	  driver doesn't know if inband audio is even available. This helps
+	  interoperability with SIP. This patch and a corresponding change
+	  in libpri work together to allow Asterisk to control the inband
+	  audio available progress indication ie on the SETUP_ACKNOWLEDGE
+	  message when dialtone is present. AST-1338 #close Reported by:
+	  Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/
+	  ........ Merged revisions 413714 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 413765 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-04-11 17:27 +0000 [r412212]  Kevin Harwell <kharwell@digium.com>
+
+	* main/asterisk.c, /: asterisk.c: suppress live_dangerously warning
+	  on rasterisk Even since the fixes of AST-2013-007, Asterisk
+	  prints the following warning on startup if the user decided to
+	  live dangerously: Privilege escalation protection disabled! See
+	  https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This
+	  message is intended for the logs and interactive startup. No need
+	  for it to appear on a remote console. This commit removes it from
+	  there. (closes issue ASTERISK-23084) Review:
+	  https://reviewboard.asterisk.org/r/3101/ ........ Merged
+	  revisions 404861 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 404888 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-10 17:34 +0000 [r410429]  Richard Mudgett <rmudgett@digium.com>
+
+	* /, main/http.c: AST-2014-001: Stack overflow in HTTP processing
+	  of Cookie headers. Sending a HTTP request that is handled by
+	  Asterisk with a large number of Cookie headers could overflow the
+	  stack. Another vulnerability along similar lines is any HTTP
+	  request with a ridiculous number of headers in the request could
+	  exhaust system memory. (closes issue ASTERISK-23340) Reported by:
+	  Lucas Molas, researcher at Programa STIC, Fundacion; and Dr.
+	  Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions
+	  410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ Merged revisions 410381 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-03-10 14:04 +0000 [r410359]  Kinsey Moore <kmoore@digium.com>
+
+	* /, channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad
+	  session timers request This change allows chan_sip to avoid
+	  creation of the channel and consumption of associated file
+	  descriptors altogether if the inbound request is going to be
+	  rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey
+	  Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey
+	  Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by
+	  Corey Farrell (license 5909) ........ Merged revisions 410308
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  Merged revisions 410311 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-02-19 19:17 +0000 [r408392]  Richard Mudgett <rmudgett@digium.com>
+
+	* main/config.c, /: config: Add file size and nanosecond resolution
+	  fields to the cached modified config file information. Repeatedly
+	  modifying config files and reloading too fast sometimes fails to
+	  reload the configuration because the cached modification
+	  timestamp has one second resolution. * Added file size and
+	  nanosecond resolution fields to the cached config file
+	  modification timestamp information. Now if the file size changes
+	  or the file system supports nanosecond resolution the modified
+	  file has a better chance of being detected for reload. * Added a
+	  missing unlock in an off-nominal code path. (closes issue
+	  AST-1303) Review: https://reviewboard.asterisk.org/r/3235/
+	  ........ Merged revisions 408387 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 408388 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-02-07 19:30 +0000 [r407746]  Richard Mudgett <rmudgett@digium.com>
+
+	* channels/chan_iax2.c, include/asterisk/frame.h,
+	  configs/iax.conf.sample, /: chan_iax2: Block unnecessary control
+	  frames to/from the wire. Establishing an IAX2 call between
+	  Asterisk v1.4 and v1.8 (or later) results in an unexpected call
+	  disconnect. The problem happens because newer values in the enum
+	  ast_control_frame_type are not consistent between the branch
+	  versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later)
+	  using IAX2 2) v1.8 answers and sends a connected line update
+	  control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4
+	  receives the control frame as an end-of-q (on v1.4
+	  AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the
+	  receive queue becomes empty. Several things are done by this
+	  patch to fix the problem and attempt to prevent it from happening
+	  again in the future: * Added a warning at the definition of enum
+	  ast_control_frame_type about how to add new control frame values.
+	  * Made block sending and receiving control frames that have no
+	  reason to go over the wire. * Extended the connectedline iax.conf
+	  parameter to also include the redirecting information updates. *
+	  Updated the connectedline iax.conf parameter documentation to
+	  include a notice that the parameter must be "no" when the peer is
+	  an Asterisk v1.4 instance. (closes issue AST-1302) Review:
+	  https://reviewboard.asterisk.org/r/3174/ ........ Merged
+	  revisions 407678 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 407727 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-25 00:13 +0000 [r406358-406469]  Richard Mudgett <rmudgett@digium.com>
+
+	* /, main/cel.c: CEL: Protect data structures during reload and
+	  shutdown. The CEL data structures need to be protected during a
+	  configuration reload and shutdown. Asterisk crashed during a
+	  shutdown because CEL events were still in flight and the CEL data
+	  structures were already destroyed. * Protected the appset and
+	  linkedids ao2 containers using the reload_lock. As a result
+	  appset, linkedids, and held objects don't need a lock. * Added
+	  NULL checks before use of the appset and linkedids ao2 containers
+	  in case the CEL module is already shutdown. * Fixed overloading
+	  of the linkedids held objects reference count. During shutdown
+	  any held objects would be leaked. * Fixed memory leak of
+	  linkedids held objects if the LINKEDID_END is not being tracked.
+	  The objects in the linkedids container were not removed if the
+	  LINKEDID_END event is not used. * Added access protection to the
+	  appset container during the CLI "cel show status" command. * Made
+	  CEL config reload not set defaults if the cel.conf file is
+	  invalid. (closes issue AST-1253) Reported by: Guenther Kelleter
+	  Review: https://reviewboard.asterisk.org/r/3127/ ........ Merged
+	  revisions 406417 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 406418 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* main/manager.c, /: manager: Protect data structures during
+	  shutdown. Occasionally, the manager module would get an
+	  "INTERNAL_OBJ: bad magic number" error on a "core restart
+	  gracefully" command if an AMI connection is established. * Added
+	  ao2_global_obj protection to the sessions global container. *
+	  Fixed the order of unreferencing a session object in
+	  session_destroy(). * Removed unnecessary container traversals of
+	  the white/black filters during session_destructor(). (closes
+	  issue AST-1242) Reported by: Guenther Kelleter Review:
+	  https://reviewboard.asterisk.org/r/3144/ ........ Merged
+	  revisions 406341 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-15 15:27 +0000 [r405536-405578]  Matthew Jordan <mjordan@digium.com>
+
+	* main/pbx.c, /: pbx.c: put copy of ast_exten.data on stack to
+	  prevent memory corruption During dialplan execution in
+	  pbx_extension_helper(), the contexts global read lock prevents
+	  link list corruption, but was released with a pointer to the
+	  ast_exten and data later used in variable substitution. Instead,
+	  this patch removes pbx_substitute_variables() and locates a copy
+	  of the ast_exten data on the stack before releasing the lock,
+	  where ast_exten could get free'd by another thread performing a
+	  module reload. (issue AST-1179) Reported by: Thomas Arimont
+	  (issue AST-1246) Reported by: Alexander Hömig Review:
+	  https://reviewboard.asterisk.org/r/3055/ ........ Merged
+	  revisions 403862 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 403863 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* /, channels/chan_sip.c: chan_sip: Hangup transferer/transferee
+	  when transfer to Parking fails When performing a SIP transfer to
+	  a Park extension, if the Park fails, chan_sip will currently not
+	  hang up either the transferer or the transfer target. This
+	  results in the channels being orphaned with no thread to service
+	  frames, resulting in stuck channels. This patch immediately hangs
+	  up the two channels if a Park fails. (closes issue
+	  ASTERISK-22834) Reported by: rsw686 Tested by: rsw686 (closes
+	  issue ASTERISK-23047) Reported by: Tommy Thompson Tested by:
+	  Tommy Thomspon Review: https://reviewboard.asterisk.org/r/3107
+	  ........ Merged revisions 405380 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-14 18:50 +0000 [r405488]  Richard Mudgett <rmudgett@digium.com>
+
+	* apps/app_verbose.c, main/asterisk.c, configs/logger.conf.sample,
+	  main/cli.c, include/asterisk/logger.h, main/pbx.c,
+	  main/manager.c, /, funcs/func_timeout.c, apps/app_dumpchan.c,
+	  main/logger.c, UPGRADE.txt: verbosity: Fix performance of console
+	  verbose messages. The per console verbose level feature as
+	  previously implemented caused a large performance penalty. The
+	  fix required some minor incompatibilities if the new rasterisk is
+	  used to connect to an earlier version. If the new rasterisk
+	  connects to an older Asterisk version then the root console
+	  verbose level is always affected by the "core set verbose"
+	  command of the remote console even though it may appear to only
+	  affect the current console. If an older version of rasterisk
+	  connects to the new version then the "core set verbose" command
+	  will have no effect. * Fixed the verbose performance by not
+	  generating a verbose message if nothing is going to use it and
+	  then filtered any generated verbose messages before actually
+	  sending them to the remote consoles. * Split the "core set debug"
+	  and "core set verbose" CLI commands to remove the per module
+	  verbose support that cannot work with the per console verbose
+	  level. * Added a silent option to the "core set verbose" command.
+	  * Fixed "core set debug off" tab completion. * Made "core show
+	  settings" list the current console verbosity in addition to the
+	  root console verbosity. * Changed the default verbose level of
+	  the 'verbose' setting in the logger.conf [logfiles] section. The
+	  default is now to once again follow the current root console
+	  level. As a result, using the AMI Command action with "core set
+	  verbose" could again set the root console verbose level and
+	  affect the verbose level logged. (closes issue AST-1252) Reported
+	  by: Guenther Kelleter Review:
+	  https://reviewboard.asterisk.org/r/3114/ ........ Merged
+	  revisions 405431 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2014-01-09 16:34 +0000 [r405233]  Matthew Jordan <mjordan@digium.com>
+
+	* /, apps/app_confbridge.c,
+	  apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix
+	  crash caused when waitmarked/marked users leave together When
+	  waitmarked users join a ConfBridge, the conference state is
+	  transitioned from EMPTY -> INACTIVE. In this state, the users are
+	  maintined in a waiting users list. When a marked user joins, the
+	  ConfBridge conference transitions from INACTIVE -> MULTI_MARKED,
+	  and all users are put onto the active list of users. This process
+	  works correctly. When the marked user leaves, if they are the
+	  last marked user, the MULTI_MARKED state does the following: (1)
+	  It plays back a message to the bridge stating that the leader has
+	  left the conference. This requires an unlocking of the bridge.
+	  (2) It moves waitmarked users back to the waiting list (3) It
+	  transitions to the appropriate state: in this case, INACTIVE
+	  However, because it plays the prompt back to the bridge before
+	  moving the users and before finishing the state transition, this
+	  creates a race condition: with the bridge unlocked, waitmarked
+	  users who leave the conference (or are kicked from it) can cause
+	  a state transition of the bridge to another state before the
+	  conference is transitioned to the INACTIVE state. This causes the
+	  state machine to get a bit wonky, often leading to a crash when
+	  the MULTI_MARKED state attempts to conclude its processing. This
+	  patch fixes this problem: (1) It prevents kicked users from being
+	  kicked again. That's just a nicety. (2) More importantly, it
+	  fixes the race condition by only playing the prompt once the
+	  state has transitioned correctly to INACTIVE. If waitmarked users
+	  sneak out during the prompt being played, no harm no foul.
+	  Review: https://reviewboard.asterisk.org/r/3108/ (closes issue
+	  AST-1258) Reported by: Steve Pitts ........ Merged revisions
+	  405215 from http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-12-19 16:38 +0000 [r404349]  Scott Griepentrog <sgriepentrog@digium.com>
+
+	* main/db.c, /: astdb: crash in sqlite3 during shutdown When
+	  Asterisk is shut down, the astdb_atexit() function releases
+	  (finalize) the previously initiated (prepared) SQL statements in
+	  sqlite3. Another thread making a subsequent request can cause a
+	  crash in sqlite3. This patch eliminates that issue by resetting
+	  the statement pointer after it is released/cleared. The sqlite3
+	  code detects the null pointer, and aborts the operation cleanly.
+	  (closes issue AST-1265) Reported by: Alexander Hömig (closes
+	  issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter
+	  Review: https://reviewboard.asterisk.org/r/3078/ ........ Merged
+	  revisions 404344 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-12-16 17:29 +0000 [r403956]  David M. Lee <dlee@digium.com>
+
+	* funcs/func_realtime.c, main/pbx.c, main/tcptls.c,
+	  funcs/func_db.c, /, README-SERIOUSLY.bestpractices.txt,
+	  configs/asterisk.conf.sample, funcs/func_shell.c,
+	  funcs/func_env.c, funcs/func_lock.c, UPGRADE.txt,
+	  include/asterisk/pbx.h, main/asterisk.c: security: Inhibit
+	  execution of privilege escalating functions This patch allows
+	  individual dialplan functions to be marked as 'dangerous', to
+	  inhibit their execution from external sources. A 'dangerous'
+	  function is one which results in a privilege escalation. For
+	  example, if one were to read the channel variable SHELL(rm -rf /)
+	  Bad Things(TM) could happen; even if the external source has only
+	  read permissions. Execution from external sources may be enabled
+	  by setting 'live_dangerously' to 'yes' in the [options] section
+	  of asterisk.conf. Although doing so is not recommended. (closes
+	  issue ASTERISK-22905) Review:
+	  http://reviewboard.digium.internal/r/432/ ........ Merged
+	  revisions 403913 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 403917 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-12-16 15:38 +0000 [r403860]  Scott Griepentrog <sgriepentrog@digium.com>
+
+	* apps/app_sms.c: app_sms: BufferOverflow when receiving odd length
+	  16 bit message This patch prevents an infinite loop overwriting
+	  memory when a message is received into the unpacksms16()
+	  function, where the length of the message is an odd number of
+	  bytes. (closes issue ASTERISK-22590) Reported by: Jan Juergens
+	  Tested by: Jan Juergens
+
+2013-11-04 21:20 +0000 [r402463]  Kevin Harwell <kharwell@digium.com>
+
+	* /, channels/chan_sip.c: chan_sip: notify dialog info ignores
+	  presentation indicator in callerid The presentation indicator in
+	  a callerid (e.g. set by dialplan function
+	  Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog
+	  Info Notifies are generated during extension monitoring. Added a
+	  check to make sure the name and/or number presentations on the
+	  callee (remote identity) are set to allow. If they are restricted
+	  then "anonymous" is used instead. (closes issue AST-1175)
+	  Reported by: Thomas Arimont Review:
+	  https://reviewboard.asterisk.org/r/2976/ ........ Merged
+	  revisions 402450 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+2013-11-01 20:39 +0000 [r402377-402383]  Matthew Jordan <mjordan@digium.com>
+
+	* asterisk-11.6.0-summary.html (removed),
+	  asterisk-11.6.0-summary.txt (removed): Remove old summaries
+
+	* include/asterisk/pbx.h, res/res_rtp_asterisk.c, main/pbx.c, /,
+	  configure, configure.ac: Multiple revisions
+	  396884,400075,400093,401446,401960 ........ r396884 | jbigelow |
+	  2013-08-16 17:45:10 -0500 (Fri, 16 Aug 2013) | 8 lines Add test
+	  suite events to indicate when a feature is detected or not These
+	  are needed by the bridge test suite tests for them to be able to
+	  run against Asterisk 11. Review:
+	  https://reviewboard.asterisk.org/r/2751/ ........ r400075 |
+	  mjordan | 2013-09-28 16:59:12 -0500 (Sat, 28 Sep 2013) | 16 lines
+	  Add check for openSUSE when detecting bfd library In
+	  ASTERISK-17842, some additional library checks were added to the
+	  configure script so that the bfd library could be found on CentOS
+	  and Fedora systems. As it turns out, openSUSE requires an
+	  additional library. This patch adds another check to the
+	  configure script for openSUSE that will add that library. Review:
+	  https://reviewboard.asterisk.org/r/2885/ (closes issue AST-1169)
+	  Reported by: Guenther Kelleter ........ Merged revisions 400073
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r400093 | mjordan | 2013-09-28 17:21:37 -0500 (Sat, 28 Sep 2013)
+	  | 23 lines res_rtp_asterisk: Correct erroneous lost packet
+	  information in RTCP reports RTCP's calculation of the number of
+	  lost packets in an RTP stream is based on that stream's sequence
+	  number count, the number of received packets, and how many
+	  packets we expect to receive. When the SSRC for an RTP stream
+	  changes, there can - and almost always will be - a large jump in
+	  the next packet's timestamp and sequence number. If we don't
+	  reset the number of received packets, sequence number count, and
+	  other metrics used by RTCP, the next RR/SR report will use the
+	  previous SSRC's values to calculate the lost packet count for the
+	  new SSRC - resulting in a very large number of lost packets. This
+	  patch modifies res_rtp_asterisk such that, if it detects a SSRC
+	  change, it will reset the various values used by the RTCP
+	  calculations. From the perspective of RTCP, this appears as a new
+	  media stream - which is what it is. Review:
+	  https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174)
+	  Reported by: Thomas Arimont ........ Merged revisions 400089 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r401446 | mjordan | 2013-10-22 17:42:24 -0500 (Tue, 22 Oct 2013)
+	  | 15 lines res_rtp_asterisk: Fix crash when RTCP is not available
+	  during SSRC change In r400089, a patch was put in to correct
+	  erroneous RTCP statistic resets. Unfortunately, ast_rtp_read can
+	  be called on an RTP instance that does not have RTCP information.
+	  This patch prevents that crash by only resetting the statistics
+	  if we do actually have an RTCP instance. (issue AST-1174) (closes
+	  issue ASTERISK-22667) Reported by: John Bigelow ........ Merged
+	  revisions 401445 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r401960 | sgriepentrog | 2013-10-25 15:44:40 -0500 (Fri, 25 Oct
+	  2013) | 15 lines pbx.c: fix confused match caller id that deleted
+	  exten still in hash This fixes a bug where a zero length callerid
+	  match adjacent to a no match callerid extension entry would be
+	  deleted together, which then resulted in hashtable references to
+	  free'd memory. A third state of the matchcid value has been added
+	  to indicate match to any extension which allows enforcing
+	  comparison of matchcid on/off without errors. (closes issue
+	  AST-1235) Reported by: Guenther Kelleter Review:
+	  https://reviewboard.asterisk.org/r/2930/ ........ Merged
+	  revisions 401959 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 396884,400075,400093,401446,401960 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11
+
+	* /: SVN properties: Add svnmerge properties for 11
+
+2013-10-22 16:10 +0000 [r401416]  bebuild <bebuild@localhost>:
+
+	* / (added): Create branch for Certified Asterisk 11.6.
+
 2013-10-21  Asterisk Development Team <asteriskteam@digium.com>
 
 	* Asterisk 11.6.0 Released.

certified-asterisk-11.6-cert7-summary.html

@@ -0,0 +1,64 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
+<html xmlns="http://www.w3.org/1999/xhtml">
+<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - certified-asterisk-11.6-cert7</title></head>
+<body>
+<h1 align="center"><a name="top">Release Summary</a></h1>
+<h3 align="center">certified-asterisk-11.6-cert7</h3>
+<h3 align="center">Date: 2014-10-20</h3>
+<h3 align="center">&lt;asteriskteam@digium.com&gt;</h3>
+<hr/>
+<h2 align="center">Table of Contents</h2>
+<ol>
+   <li><a href="#summary">Summary</a></li>
+   <li><a href="#contributors">Contributors</a></li>
+   <li><a href="#commits">Other Changes</a></li>
+   <li><a href="#diffstat">Diffstat</a></li>
+</ol>
+<hr/>
+<a name="summary"><h2 align="center">Summary</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This release has been made to address one or more security vulnerabilities that have been identified.  A security advisory document has been published for each vulnerability that includes additional information.  Users of versions of Asterisk that are affected are strongly encouraged to review the advisories and determine what action they should take to protect their systems from these issues.</p>
+<p>Security Advisories: <a href="http://downloads.asterisk.org/pub/security/AST-2014-011.html">AST-2014-011</a></p>
+<p>The data in this summary reflects changes that have been made since the previous release, certified-asterisk-11.6-cert6.</p>
+<hr/>
+<a name="contributors"><h2 align="center">Contributors</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release.  For coders, the number is how many of their patches (of any size) were committed into this release.  For testers, the number is the number of times their name was listed as assisting with testing a patch.  Finally, for reporters, the number is the number of issues that they reported that were closed by commits that went into this release.</p>
+<table width="100%" border="0">
+<tr>
+<td width="33%"><h3>Coders</h3></td>
+<td width="33%"><h3>Testers</h3></td>
+<td width="33%"><h3>Reporters</h3></td>
+</tr>
+<tr valign="top">
+<td>
+2 bebuild<br/>
+</td>
+<td>
+</td>
+<td>
+</td>
+</tr>
+</table>
+<hr/>
+<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker.  The commits may have been marked as being related to an issue.  If that is the case, the issue numbers are listed here, as well.</p>
+<table width="100%" border="1">
+<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/tags/11.6-cert7?view=revision&revision=426030">426030</a></td><td>bebuild</td><td>Create 11.6-cert7</td>
+<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/tags/11.6-cert7?view=revision&revision=426055">426055</a></td><td>bebuild</td><td>Merge 426053</td>
+<td></td></tr></table>
+<hr/>
+<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
+<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
+<pre>
+.version                                   |    2
+ChangeLog                                  |   26 ++++++++
+UPGRADE.txt                                |   12 +++
+certified-asterisk-11.6-cert6-summary.html |   62 -------------------
+certified-asterisk-11.6-cert6-summary.txt  |   93 -----------------------------
+main/tcptls.c                              |   22 +++++-
+res/res_jabber.c                           |    5 +
+res/res_xmpp.c                             |    6 +
+8 files changed, 65 insertions(+), 163 deletions(-)
+</pre><br/>
+<hr/>
+</body>
+</html>

certified-asterisk-11.6-cert7-summary.txt

@@ -0,0 +1,92 @@
+                                Release Summary
+
+                         certified-asterisk-11.6-cert7
+
+                                Date: 2014-10-20
+
+                           <asteriskteam@digium.com>
+
+     ----------------------------------------------------------------------
+
+                               Table of Contents
+
+    1. Summary
+    2. Contributors
+    3. Other Changes
+    4. Diffstat
+
+     ----------------------------------------------------------------------
+
+                                    Summary
+
+                                 [Back to Top]
+
+   This release has been made to address one or more security vulnerabilities
+   that have been identified. A security advisory document has been published
+   for each vulnerability that includes additional information. Users of
+   versions of Asterisk that are affected are strongly encouraged to review
+   the advisories and determine what action they should take to protect their
+   systems from these issues.
+
+   Security Advisories: AST-2014-011
+
+   The data in this summary reflects changes that have been made since the
+   previous release, certified-asterisk-11.6-cert6.
+
+     ----------------------------------------------------------------------
+
+                                  Contributors
+
+                                 [Back to Top]
+
+   This table lists the people who have submitted code, those that have
+   tested patches, as well as those that reported issues on the issue tracker
+   that were resolved in this release. For coders, the number is how many of
+   their patches (of any size) were committed into this release. For testers,
+   the number is the number of times their name was listed as assisting with
+   testing a patch. Finally, for reporters, the number is the number of
+   issues that they reported that were closed by commits that went into this
+   release.
+
+     Coders                   Testers                  Reporters              
+   2 bebuild                
+
+     ----------------------------------------------------------------------
+
+                      Commits Not Associated with an Issue
+
+                                 [Back to Top]
+
+   This is a list of all changes that went into this release that did not
+   directly close an issue from the issue tracker. The commits may have been
+   marked as being related to an issue. If that is the case, the issue
+   numbers are listed here, as well.
+
+   +------------------------------------------------------------------------+
+   | Revision    | Author     | Summary              | Issues Referenced    |
+   |-------------+------------+----------------------+----------------------|
+   | 426030      | bebuild    | Create 11.6-cert7    |                      |
+   |-------------+------------+----------------------+----------------------|
+   | 426055      | bebuild    | Merge 426053         |                      |
+   +------------------------------------------------------------------------+
+
+     ----------------------------------------------------------------------
+
+                                Diffstat Results
+
+                                 [Back to Top]
+
+   This is a summary of the changes to the source code that went into this
+   release that was generated using the diffstat utility.
+
+ .version                                   |    2
+ ChangeLog                                  |   26 ++++++++
+ UPGRADE.txt                                |   12 +++
+ certified-asterisk-11.6-cert6-summary.html |   62 -------------------
+ certified-asterisk-11.6-cert6-summary.txt  |   93 -----------------------------
+ main/tcptls.c                              |   22 +++++-
+ res/res_jabber.c                           |    5 +
+ res/res_xmpp.c                             |    6 +
+ 8 files changed, 65 insertions(+), 163 deletions(-)
+
+     ----------------------------------------------------------------------