[asterisk-commits] bebuild: tag certified-1.8.11-cert8 r375572 - /certified/tags/1.8.11-cert8/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Oct 31 15:34:09 CDT 2012


Author: bebuild
Date: Wed Oct 31 15:34:05 2012
New Revision: 375572

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=375572
Log:
Importing files for 1.8.11-cert8 release.

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    certified/tags/1.8.11-cert8/.version   (with props)
    certified/tags/1.8.11-cert8/ChangeLog   (with props)

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+2012-10-31  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.11-cert8 Released.
+
+2012-10-31 20:26 +0000 [r375568-375570]  Matthew Jordan <mjordan at digium.com>
+
+	* /, UPGRADE.txt: Multiple revisions 375242,375244 ........ r375242
+	  | jrose | 2012-10-18 16:30:13 -0500 (Thu, 18 Oct 2012) | 8 lines
+	  app_queue: add upgrade notes for 375216 Adds notes describing
+	  behavioral changes to rrmemory strategy caused by 375216 (issue
+	  AST-989) Reported by: Thomas Arimont ........ r375244 | jrose |
+	  2012-10-18 16:36:59 -0500 (Thu, 18 Oct 2012) | 4 lines Correct
+	  version number in Upgrade.txt release notes pertaining to queue
+	  order Showed 1.8.17 to 1.8.18, needs to be 1.8.18 to 1.8.19
+	  ........ Merged revisions 375242,375244 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* apps/app_queue.c, /: app_queue: Make ordering of
+	  rrmemory/rrordered persist over add/remove members Prior to this
+	  patch, adding, removing or reloading members to rrmemory would
+	  cause the order to become completely jumbled. Now it behaves more
+	  or less like rrordered other than the fact that it stores the
+	  members on a hash table rather than a linked list. This patch
+	  also prevents removal of members and member reloads from jumbling
+	  rrordered queues. (issue AST-989) Reported by: Thomas Arimont
+	  Review: https://reviewboard.asterisk.org/r/2164/ ........ Merged
+	  revisions 375216 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* channels/sip/include/sip.h, apps/app_voicemail.c, /,
+	  channels/chan_sip.c, funcs/func_math.c: Multiple revisions
+	  369436,369557,369579,369626,369652,372628 for 1.8.11-cert8
+	  ........ r369436 | twilson | 2012-06-27 15:58:51 -0500 (Wed, 27
+	  Jun 2012) | 17 lines AST-2012-010: Clean up after a reinvite that
+	  never gets a final response The basic problem is that if a
+	  re-INVITE is sent by Asterisk and it receives a provisional
+	  response, but no final response, then the dialog is never torn
+	  down. In addition to leaking memory, this also leaks file
+	  descriptors and will eventually lead to Asterisk no longer being
+	  able to process calls. This patch just keeps track of whether
+	  there is an outstanding re-INVITE, and if there is goes ahead and
+	  cleans up everything as though there was no outstanding reinvite.
+	  Review: https://reviewboard.asterisk.org/r/2009/ (closes issue
+	  ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
+	  Davies, Terry Wilson ........ r369557 | twilson | 2012-07-03
+	  09:27:02 -0500 (Tue, 03 Jul 2012) | 11 lines Better handle
+	  re-INVITEs with provisional but no final repsonses A previous
+	  attempt at fixing this issue had negative side effects related to
+	  attended transfers which this patch should resolve. Many thanks
+	  to Steve Davies for all of the good suggestions and testing.
+	  (closes issue ASTERISK-19992) Reported by: Steve Davies Tested
+	  by: Steve Davies, Terry Wilson Review:
+	  https://reviewboard.asterisk.org/r/2009/ ........ r369579 |
+	  twilson | 2012-07-03 11:58:16 -0500 (Tue, 03 Jul 2012) | 8 lines
+	  More improvements to re-INVITEs timing out after a provisional
+	  response There is no need to call check_pendings() on a final
+	  response to an INVITE when destroying the scheduler entry as it
+	  will be done later during normal processing. (issue
+	  ASTERISK-19992) ........ r369626 | mjordan | 2012-07-05 12:01:52
+	  -0500 (Thu, 05 Jul 2012) | 16 lines Do not send a BYE when a
+	  provisional response arrives during a re-INVITE Commits r369557
+	  and r369579 were done to improve handling of re-INVITEs when the
+	  UA that was supposed to receive the re-INVITE fails to respond. A
+	  limitation of those patches occurred when a UA sent a provisional
+	  response to the re-INVITE. This triggered a sending of a BYE in
+	  check_pending. This patch tweaks the handling of the re-INVITE
+	  such that a BYE is not sent in response to those messages. (issue
+	  ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
+	  patches: (reinvite_tweak.diff license #5012 by Steve Davies)
+	  ........ r369652 | kmoore | 2012-07-05 14:01:52 -0500 (Thu, 05
+	  Jul 2012) | 22 lines AST-2012-011: Resolve heap corruption issue
+	  with voicemail The heard and deleted arrays in the voicemail
+	  state structure were not handled properly following the memory
+	  leak fix in r354890 and a fix for an invalid free in r356797.
+	  This could result in accessing and writing into freed memory. The
+	  allocation for these arrays has been reworked to avoid the
+	  possibility of invalid frees, access of freed memory, and crashes
+	  that were occurring as a result of this. Locking around accesses
+	  and modifications of the voicemail state structure members
+	  dh_arraysize, heard, and deleted has been added to prevent
+	  simultaneous modification and access when IMAP storage is in use.
+	  If IMAP storage is not in use, this locking is not compiled in.
+	  Review: https://reviewboard.asterisk.org/r/1994/ (closes issue
+	  ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney,
+	  Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license
+	  6273) ........ r372628 | rmudgett | 2012-09-07 17:06:29 -0500
+	  (Fri, 07 Sep 2012) | 5 lines Remove annoying unconditional debug
+	  message from INC/DEC functions. (closes issue AST-1001) Reported
+	  by: Guenther Kelleter ........ Merged revisions
+	  369436,369557,369579,369626,369652,372628 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-15 14:11 +0000 [r375024]  Matthew Jordan <mjordan at digium.com>
+
+	* configs/sip.conf.sample, channels/sip/include/sip.h, /,
+	  channels/chan_sip.c: Properly handle UAC/UAS roles for SIP
+	  session timers The SIP session timer mechanism contains a
+	  mandatory 'refresher' parameter (included in the Session-Expires
+	  header) which is used in the session timer offer/answer signaling
+	  within a SIP Invite dialog. It looks like asterisk is
+	  interpreting the uac resp. uas role only as the initial role of
+	  client and server (caller is uac, callee is uas). The standard
+	  rfc 4028 however assigns the client role to the ((RE)-Invite)
+	  requester, the server role to the ((RE)-Invite) responder. This
+	  patch has Asterisk track the actual refresher as "us" or "them"
+	  as opposed to relying on just the configured "uas" or "uac"
+	  properties. (closes issue AST-922) Reported by: Thomas Airmont
+	  Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged
+	  revisions 373652 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-11 15:46 +0000 [r374847]  Matthew Jordan <mjordan at digium.com>
+
+	* main/cdr.c, /: Fix incorrect billing duration reported when batch
+	  mode is enabled Similar to r369351, the billing duration can be
+	  skewed when batch mode is enabled. This happened much more rarely
+	  than the duration, as it only occured when the call was answered
+	  (thereby indicating an actual answer time) and immediately hung
+	  up on (indicating a billsec of 0). Since a billing time of '0'
+	  can either mean that the call immediately ended or that the CDR
+	  was improperly answered, we have to use additional information to
+	  know whether or not we can trust the CDR billsec value. Prior to
+	  this patch, we looked to see if we had a valid answer time. If we
+	  did, and billsec was zero, we used the current time to calculate
+	  what billsec value we could from the CDR being written. If batch
+	  mode is enabled, this will incorrectly report a billsec value
+	  being much greater than the actual duration of the call. Instead
+	  of relying on the presence of an answer time to know whether or
+	  not we can re-calculate the billsec for the CDR, we now also use
+	  the presence of the CDR's end time to know if we need to
+	  re-calculate or whether we can trust the billsec value that we
+	  have. This prevents erroneous jumps in the billsec value, while
+	  still making sure that in the worst case, some billing time will
+	  be calculated. (closes issue AST-1016) Reported by: Thomas
+	  Arimont Tested by: Thomas Arimont ........ Merged revisions
+	  374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-10 21:31 +0000 [r374806-374832]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/cel.c, /: Fix compiler warnings. gcc (GCC) 4.2.4 has
+	  problems casting away constness. ........ Merged revisions 370275
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* apps/app_queue.c, /: app_queue: Made pass connected line updates
+	  from the caller to ringing queue members. Party A calls Party B
+	  Party B puts Party A on hold. Party B calls a queue. Ringing
+	  queue member D sees Party B identification. Party B transfers
+	  Party A to the queue. Queue member D does not get a connected
+	  line update for Party A. Queue member D answers the call and
+	  still sees Party B information. However, if Party A later
+	  transfers the call to Party C then queue member D gets a
+	  connected line update for Party C. * Made pass connected line
+	  updates from the caller to queue members while the queue members
+	  are ringing. (closes issue AST-1017) Reported by: Thomas Arimont
+	  (closes issue ABE-2886) Reported by: Thomas Arimont Tested by:
+	  rmudgett ........ Merged revisions 374801 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  ........ Merged revisions 374802 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-05 18:56 +0000 [r374540]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+	  channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
+	  Multiple revisions 370563,374536 ........ r370563 | rmudgett |
+	  2012-07-30 11:47:19 -0500 (Mon, 30 Jul 2012) | 2 lines Release B
+	  channel allocation on error path in chan_misdn. ........ r374536
+	  | rmudgett | 2012-10-05 13:20:01 -0500 (Fri, 05 Oct 2012) | 159
+	  lines Merged revisions 374515-374535 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
+	  (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
+	  Made setup_bc() static. Patches: patch1_unused-code.diff (license
+	  #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
+	  ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
+	  (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
+	  states Patches: patch2_unused-states.diff (license #6372) patch
+	  uploaded by Guenther Kelleter JIRA ABE-2882 ................
+	  r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
+	  | 16 lines chan_misdn: Remove unnecessary null pointer checks and
+	  checks for stack->nt * cleanup_bc() is always called with valid
+	  bc (or it would've crashed before). * Value of stack->nt is known
+	  in advance at some places. * Rename handle_event() to
+	  handle_event_te(), handle_frm() to handle_frm_te(). Patches:
+	  patch3_checks.diff (license #6372) patch uploaded by Guenther
+	  Kelleter Modified JIRA ABE-2882 ................ r374518 |
+	  rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
+	  chan_misdn: Fix spelling in log messages Patches:
+	  patch4_spelling.diff (license #6372) patch uploaded by Guenther
+	  Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
+	  2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
+	  chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
+	  calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
+	  emptied, cleaned and set not in use, although
+	  misdn_lib_send_event() already did the same. This is bad. When
+	  it's not in use we are not allowed to touch it. * Moved log
+	  message in front of the resulting actions and fixed it to match
+	  the case. Patches: patch5_bccleanup.diff (license #6372) patch
+	  uploaded by Guenther Kelleter JIRA ABE-2882 ................
+	  r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
+	  | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
+	  etc., really bad stuff. * Fix return codes of cb_events() for
+	  EVENT_SETUP to use caller's cleanup mechanisms. * Move
+	  cl_queue_chan() call after bearer check. Patches:
+	  patch6_leaks.diff (license #6372) patch uploaded by Guenther
+	  Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
+	  2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
+	  chan_misdn: We must initialize cause on sending a DISCONNECT. We
+	  must initialize cause on sending a DISCONNECT, so it is later
+	  correctly indicated to ast_channel in case the answer
+	  (RELEASE/RELEASE_COMPLETE) does not include one. Patches:
+	  patch7_hangupcause.diff (license #6372) patch uploaded by
+	  Guenther Kelleter JIRA ABE-2882 ................ r374522 |
+	  rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
+	  chan_misdn: Remove unused code for upqueue Patches:
+	  patch8_unused-upqueue.diff (license #6372) patch uploaded by
+	  Guenther Kelleter JIRA ABE-2882 ................ r374523 |
+	  rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
+	  chan_misdn: Improve debugging (port number, messages fixed, dups
+	  removed) Patches: patch9_debug.diff (license #6372) patch
+	  uploaded by Guenther Kelleter JIRA ABE-2882 ................
+	  r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
+	  | 8 lines chan_misdn: Better debug: we can print_bc_info even if
+	  there's no ast leg. Patches: patch10_debug-bc-2.diff (license
+	  #6372) patch uploaded by Guenther Kelleter Modified. JIRA
+	  ABE-2882 ................ r374534 | rmudgett | 2012-10-05
+	  12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
+	  setup_bc() is called too early for an incoming SETUP on TE. This
+	  prevents the B channel from being setup for HDLC mode when
+	  requested by the bearer capability and config option hdlc=yes. It
+	  violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
+	  connect to the channel until a CONNECT ACKNOWLEDGE message has
+	  been received." * Call setup_bc() on receipt of
+	  CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
+	  PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
+	  Guenther Kelleter Modified. JIRA ABE-2881 ................
+	  r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
+	  | 2 lines chan_misdn: Remove some more deadcode. ................
+	  ........ Merged revisions 370563,374536 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-14 20:25 +0000 [r373088]  Matthew Jordan <mjordan at digium.com>
+
+	* include/asterisk/tcptls.h, main/tcptls.c, /, channels/chan_sip.c,
+	  main/ssl.c: Multiple revisions 366052,367002,367416,369731,373061
+	  ........ r366052 | mmichelson | 2012-05-10 11:10:18 -0500 (Thu,
+	  10 May 2012) | 7 lines Close the proper tcptls_session when
+	  session creation fails. (issue AST-998) Reported by: Thomas
+	  Arimont Tested by: Thomas Arimont ........ r367002 | mmichelson |
+	  2012-05-18 11:53:47 -0500 (Fri, 18 May 2012) | 17 lines Fix
+	  memory leak of SSL_CTX structures in TLS core. SSL_CTX structures
+	  were allocated but never freed. This was a bigger issue for
+	  clients than servers since new SSL_CTX structures could be
+	  allocated for each connection. Servers, on the other hand,
+	  typically set up a single SSL_CTX for their lifetime. This is
+	  solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an
+	  ssl_ctx on it, it is freed so that a new one can take its place.
+	  2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
+	  been added so that servers can properly free their SSL_CTXs.
+	  (issue ASTERISK-19278) ........ r367416 | mmichelson | 2012-05-23
+	  15:27:47 -0500 (Wed, 23 May 2012) | 5 lines Only call
+	  SSL_CTX_free if DO_SSL is defined. Thanks to Paul Belanger for
+	  pointing out this error. ........ r369731 | mmichelson |
+	  2012-07-06 13:40:06 -0500 (Fri, 06 Jul 2012) | 19 lines Remove a
+	  superfluous and dangerous freeing of an SSL_CTX. The problem here
+	  is that multiple server sessions share a SSL_CTX. When one
+	  session ended, the SSL_CTX would be freed and set NULL, leaving
+	  the other sessions unable to function. The code being removed is
+	  superfluous because the SSL_CTX structures for servers will be
+	  properly freed when ast_ssl_teardown is called. (closes issue
+	  ASTERISK-20074) Reported by Trevor Helmsley Patches:
+	  ASTERISK-20074.diff uploaded by Mark Michelson (license #5049)
+	  Testers: Trevor Helmsley ........ r373061 | mjordan | 2012-09-14
+	  14:07:20 -0500 (Fri, 14 Sep 2012) | 28 lines Resolve memory leaks
+	  in TLS initialization and TLS client connections This patch
+	  resolves two sources of memory leaks when using TLS in Asterisk:
+	  1) It removes improper initialization (and multiple
+	  re-initializations) of portions of the SSL library. Asterisk
+	  calls SSL_library_init and SSL_load_error_strings during SSL
+	  initialization; collectively this obviates the need for calling
+	  any of the following during initialization or client connection
+	  handling: * ERR_load_crypto_strings (handled by
+	  SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
+	  SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
+	  SSL_library_init) 2) Failure to completely clean up all memory
+	  allocated by Asterisk and by the SSL library for TLS clients.
+	  This included not freeing the SSL_CTX object in the SIP channel
+	  driver, as well as not clearing the error stack when the TLS
+	  client exited. Note that these memory leaks were found by Thomas
+	  Arimont, and this patch was essentially written by him with some
+	  minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
+	  Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
+	  Arimont (license 5525) Review:
+	  https://reviewboard.asterisk.org/r/2105 ........ Merged revisions
+	  366052,367002,367416,369731,373061 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-30  Asterisk Development Team <asteriskteam at digium.com>
+
+        * Asterisk 1.8.11-cert7 Released.
+	
+        * AST-2012-013: Resolve ACL rules being ignored during calls by some
+          IAX2 peers
+			  
+        * AST-2012-012: Resolve AMI User Unauthorized Shell Access through
+          ExternalIVR
+
+2012-08-24  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Certified Asterisk 1.8.11-cert6 Released.
+
+2012-08-24 13:55 +0000 [r371650-371651]  Matthew Jordan <mjordan at digium.com>
+
+	* configs/sip.conf.sample, main/cel.c, channels/sip/include/sip.h,
+	  main/xmldoc.c, main/channel.c, /, channels/chan_sip.c: Merge
+	  patches for 1.8.11-cert6 This includes the following * r369351
+	  for AST-883 * r368807 for AST-884 * r356604, r356650, r364203 for
+	  AST-890 * r370618 for AST-896 * r370205, r370273, r370360 for
+	  AST-916 * r371469 for AST-932
+
+	* main/cdr.c, /: Merge r369351 for AST-883
+
+2012-08-07 15:40 +0000 [r370843]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/sip/config_parser.c, channels/sip/include/sip.h, /,
+	  channels/chan_sip.c: Fix error in the "IPorHost" section of a SIP
+	  dialstring. This is based on the review request posted by Walter
+	  Doekes (referenced lower in the commit message) The main fix here
+	  is to treat the IPorHost portion of the dial string as a
+	  temporary outbound proxy. This ensures requests get sent to the
+	  proper location. Due to the age of the request, some parts were
+	  no longer relevant. For instance, the request moved outbound
+	  proxy parsing code into a single method. This is done in a
+	  previous commit, so it was not necessary to do again. Also, the
+	  review request fixed some errors with regards to request routing
+	  for CANCEL and ACK requests. This has also been fixed in more
+	  recent commits. (closes issue ASTERISK-19677) reported by Walter
+	  Doekes Review https://reviewboard.asterisk.org/r/1859 ........
+	  Merged revisions 370769 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-07-27  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Certified Asterisk 1.8.11-cert5 Released.
+
+2012-07-11  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Certified Asterisk 1.8.11-cert5-rc2 Released.
+
+	* channels/sig_analog.c: Fix bad merge of r368759 in sig_analog
+
+	  The patch for r368759 in Asterisk 1.8 relied upon the methods
+	  analog_unlock_private/analog_lock_private.  In earlier versions of
+	  1.8, including the Certified Asterisk 1.8.11 branch, those two
+	  methods were unused, and hence were undefined out of the source.
+	  When the patch was made for 1.8.11-cert5, those two functions were
+	  not re-defined back in.  This caused linking errors when sig_analog
+	  was loaded.
+
+	  This patch properly restores those two methods, such that the fix
+	  for AST-891 works correctly.
+
+	  (issue AST-891)
+
+2012-07-09  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Certified Asterisk 1.8.11-cert5-rc1 Released.
+
+2012-07-09 19:59 +0000 [r369848]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
+	  channels/sig_analog.c, /, channels/chan_sip.c,
+	  include/asterisk/channel.h: Fix deadlock between bridged channels
+	  that attempt to set the hangup source Calling
+	  ast_set_hangupsource with the channel lock held can result in a
+	  deadlock because the function also locks the bridged channel.
+	  (issue AST-891)
+
+2012-07-09 19:50 +0000 [r369845]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Add support for exposing the received
+	  contact URI and also for setting the request URI in messages.
+	  (closes issue AST-911)
+
+2012-07-09 19:06 +0000 [r369839-369840]  Jason Parker <jparker at digium.com>
+
+	* include/asterisk/app_voicemail.h (removed): Remove file that
+	  should no longer exist.
+
+	* apps/app_mixmonitor.c, apps/app_voicemail.c,
+	  include/asterisk/callerid.h, include/asterisk/app.h,
+	  channels/chan_sip.c, apps/app_voicemail.exports.in,
+	  tests/test_voicemail_api.c, main/callerid.c, main/app.c,
+	  include/asterisk/app_voicemail.h: Re-merge changes that were
+	  reverted.
+	  ------------------------------------------------------------------------
+	  r365395 | qwell | 2012-05-04 16:17:08 -0500 (Fri, 04 May 2012) |
+	  7 lines Add support for folders in MixMonitor 'm' option.
+	  Backport manager actions. The manager actions are needed, so
+	  MixMonitor can be executed on existing channels. (issue DPMA-68)
+	  ------------------------------------------------------------------------
+	  r364761 | qwell | 2012-05-01 12:25:14 -0500 (Tue, 01 May 2012) |
+	  6 lines Remove folder_dir from voicemail snapshots API. It was
+	  both unused (except in tests, where it was fudged) and
+	  unnecessary. (closes issue AST-842)
+	  ------------------------------------------------------------------------
+	  r367161 | mmichelson | 2012-05-21 14:05:52 -0500 (Mon, 21 May
+	  2012) | 21 lines Add "send to voicemail" Digium phone
+	  functionality to Asterisk. This change accommodates two methods
+	  by which calls can be directed to a user's voicemail. * Incoming
+	  calls can be redirected to any user's voicemail. * Established
+	  calls can be blind transferred to any user's voicemail. Digium
+	  phones indicate the desire to direct a call to voicemail by using
+	  a Diversion header with a reason parameter of "send_to_vm". This
+	  patch adds the "send_to_vm" reason as a valid redirecting reason.
+	  In addition, chan_sip.c has been modified to update redirecting
+	  information on the transferred channel by reading a Diversion
+	  header on a REFER request. (closes issue AST-871) Reported by
+	  Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925
+	  ------------------------------------------------------------------------
+	  r368790 | mjordan | 2012-06-12 08:44:36 -0500 (Tue, 12 Jun 2012)
+	  | 18 lines Fix deadlock in SIP transfers that involve a REFER
+	  request In r367163, "send to voicemail" functionality was added
+	  to the SIP channel driver. This required updating the party
+	  redirecting information for the channel based on the headers
+	  provided in the REFER request. When the redirecting party
+	  information is updated on the channel, a call to
+	  ast_indicate_data occurs. Because handle_request_refer still had
+	  the sip_pvt locked, a deadlock could occur between the pbx_thread
+	  and the do_monitor thread servicing the REFER request. This patch
+	  preserves the proper locking order between the channel and the
+	  sip_pvt by ensuring that the sip_pvt is unlocked prior to
+	  updating the party redirecting information on the channel.
+	  (closes issue AST-903) Reported by: Matt Jordan patches:
+	  jira_ast_903_trunk.patch by rmudgett (license 5621)
+	  ------------------------------------------------------------------------
+	  r368962 | qwell | 2012-06-14 13:38:48 -0500 (Thu, 14 Jun 2012) |
+	  11 lines Remove global symbol requirement from app_voicemail.
+	  This uses the existing "function installation" stuff that already
+	  existed for other functions, like getting message counts. (closes
+	  issue AST-807) (issue AST-901) (issue AST-908) Review:
+	  https://reviewboard.asterisk.org/r/1965/
+	  ------------------------------------------------------------------------
+	  r368964 | qwell | 2012-06-14 14:03:24 -0500 (Thu, 14 Jun 2012) |
+	  8 lines These functions that were moved need to be static. Also
+	  wrap test functions in a #ifdef. (issue AST-807) (issue AST-901)
+	  (issue AST-908)
+	  ------------------------------------------------------------------------
+	  r368998 | qwell | 2012-06-15 10:31:43 -0500 (Fri, 15 Jun 2012) |
+	  6 lines Remove some symbol exports that got missed in the removal
+	  of global symbols. (issue AST-807) (issue AST-901) (issue
+	  AST-908)
+	  ------------------------------------------------------------------------
+	  r369024 | qwell | 2012-06-15 11:29:40 -0500 (Fri, 15 Jun 2012) |
+	  2 lines Fix voicemail API tests by using the correct argument
+	  order for create/destroy.
+	  ------------------------------------------------------------------------
+
+2012-07-05  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Certified Asterisk 1.8.11-cert4 Released.
+
+	* AST-2012-010
+
+	* AST-2012-011
+
+2012-05-29  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Certified Asterisk 1.8.11-cert2 Released.
+
+	* AST-2012-007
+
+	* AST-2012-008
+
+2012-04-25  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Certified Asterisk 1.8.11-cert1 Released.
+
+2012-04-25 16:53 +0000 [r363674]  Jason Parker <jparker at digium.com>
+
+	* / (added): Asterisk 1.8-digiumphones branch has become Certified
+	  Asterisk 1.8.11. For more details about Certified Asterisk, see
+	  http://tinyurl.com/7pfp639
+
+2012-04-24 20:57 +0000 [r363374]  Jason Parker <jparker at digium.com>
+
+	* /res/res_smdi.c,
+	  /apps/app_osplookup.c,
+	  /channels/chan_misdn.c,
+	  /channels/chan_skinny.c,
+	  /funcs/func_frame_trace.c,
+	  /cdr/cdr_sqlite.c,
+	  /pbx/pbx_realtime.c,
+	  /apps/app_amd.c,
+	  /pbx/pbx_dundi.c,
+	  /apps/app_url.c,
+	  /channels/chan_nbs.c,
+	  /apps/app_externalivr.c,
+	  /apps/app_zapateller.c,
+	  /cdr/cdr_odbc.c,
+	  /res/res_fax_spandsp.c,
+	  /channels/chan_mgcp.c,
+	  /cel/cel_pgsql.c,
+	  /apps/app_readfile.c,
+	  /apps/app_test.c,
+	  /apps/app_ices.c,
+	  /channels/chan_gtalk.c,
+	  /cdr/cdr_csv.c,
+	  /channels/chan_phone.c,
+	  /funcs/func_pitchshift.c,
+	  /apps/app_waitforring.c,
+	  /formats/format_vox.c,
+	  /res/res_timing_pthread.c,
+	  /apps/app_minivm.c,
+	  /channels/chan_h323.c,
+	  /cel/cel_sqlite3_custom.c,
+	  /apps/app_confbridge.c,
+	  /res/res_config_ldap.c,
+	  /apps/app_nbscat.c,
+	  /cdr/cdr_sqlite3_custom.c,
+	  /res/res_snmp.c,
+	  /apps/app_dictate.c,
+	  /apps/app_waitforsilence.c,
+	  /apps/app_dahdiras.c,
+	  /pbx/pbx_lua.c,
+	  /apps/app_alarmreceiver.c,
+	  /apps/app_image.c,
+	  /res/res_ael_share.c,
+	  /cdr/cdr_tds.c,
+	  /apps/app_setcallerid.c,
+	  /apps/app_mp3.c,
+	  /channels/chan_alsa.c,
+	  /res/res_timing_kqueue.c,
+	  /channels/chan_unistim.c,
+	  /apps/app_dahdibarge.c,
+	  /res/res_config_pgsql.c,
+	  /res/res_adsi.c,
+	  /res/res_phoneprov.c,
+	  /apps/app_morsecode.c,
+	  /cdr/cdr_pgsql.c,
+	  /res/res_config_sqlite.c,
+	  /channels/chan_jingle.c,
+	  /pbx/pbx_ael.c,
+	  /apps/app_sms.c,
+	  /formats/format_jpeg.c,
+	  /apps/app_jack.c,
+	  /apps/app_adsiprog.c,
+	  /cel/cel_radius.c,
+	  /res/res_ais.c,
+	  /cel/cel_tds.c,
+	  /apps/app_festival.c,
+	  /apps/app_chanisavail.c,
+	  /channels/chan_console.c,
+	  /apps/app_talkdetect.c,
+	  /res/res_jabber.c,
+	  /cdr/cdr_radius.c,
+	  /apps/app_getcpeid.c,
+	  /channels/chan_oss.c: Disable extended
+	  and deprecated modules by default. Users can still enable any of
+	  these using menuselect if they so choose. (closes issue AST-873)
+
+2012-04-23 15:17 +0000 [r363161]  Jason Parker <jparker at digium.com>
+
+	* /main/manager.c,
+	  ,
+	  /channels/chan_sip.c,
+	  /channels/chan_skinny.c: Multiple
+	  revisions 363102,363106,363141 ........ r363102 | mjordan |
+	  2012-04-23 08:37:55 -0500 (Mon, 23 Apr 2012) | 16 lines
+	  AST-2012-005: Fix remotely exploitable heap overflow in keypad
+	  button handling When handling a keypad button message event, the
+	  received digit is placed into a fixed length buffer that acts as
+	  a queue. When a new message event is received, the length of that
+	  buffer is not checked before placing the new digit on the end of
+	  the queue. The situation exists where sufficient keypad button
+	  message events would occur that would cause the buffer to be
+	  overrun. This patch explicitly checks that there is sufficient
+	  room in the buffer before appending a new digit. (closes issue
+	  ASTERISK-19592) Reported by: Russell Bryant ........ Merged
+	  revisions 363100 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+	  r363106 | mjordan | 2012-04-23 09:05:02 -0500 (Mon, 23 Apr 2012)
+	  | 17 lines AST-2012-006: Fix crash in UPDATE handling when no
+	  channel owner exists If Asterisk receives a SIP UPDATE request
+	  after a call has been terminated and the channel has been
+	  destroyed but before the SIP dialog has been destroyed, a
+	  condition exists where a connected line update would be attempted
+	  on a non-existing channel. This would cause Asterisk to crash.
+	  The patch resolves this by first ensuring that the SIP dialog has
+	  an owning channel before attempting a connected line update. If
+	  an UPDATE request is received and no channel is associated with
+	  the dialog, a 481 response is sent. (closes issue ASTERISK-19770)
+	  Reported by: Thomas Arimont Tested by: Matt Jordan Patches:
+	  ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license
+	  6283) ........ r363141 | jrose | 2012-04-23 09:33:16 -0500 (Mon,
+	  23 Apr 2012) | 20 lines AST-2012-004: Fix an error that allows
+	  AMI users to run shell commands sans authorization. As detailed
+	  in the advisory, AMI users without write authorization for SYSTEM
+	  class AMI actions were able to run system commands by going
+	  through other AMI commands which did not require that
+	  authorization. Specifically, GetVar and Status allowed users to
+	  do this by setting their variable/s options to the SHELL or EVAL
+	  functions. Also, within 1.8, 10, and trunk there was a similar
+	  flaw with the Originate action that allowed users with originate
+	  permission to run MixMonitor and supply a shell command in the
+	  Data argument. That flaw is fixed in those versions of this
+	  patch. (closes issue ASTERISK-17465) Reported By: David Woolley
+	  Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
+	  (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
+	  (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
+	  (license 6182) ........ Merged revisions 363117 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+	  Merged revisions 363102,363106,363141 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-19 20:31 +0000 [r362673]  Mark Michelson <mmichelson at digium.com>
+
+	* /channels/chan_sip.c: Add a test
+	  application for sending custom SIP INFO messages. When
+	  TEST_FRAMEWORK is enabled, SIPSendCustomInfo is available to test
+	  sending custom INFO requests. Review:
+	  https://reviewboard.asterisk.org/r/1866
+
+2012-04-13 17:19 +0000 [r362042-362132]  Matthew Jordan <mjordan at digium.com>
+
+	* : Rename property branches-1.8-merged to
+	  branch-1.8-merged
+
+	* : Update properties on 1.8-digiumphones
+	  Change the merge property tag from svnmerge-integrated to
+	  branches-1.8-merged. Added merged revisions from r362042.
+
+	* ,
+	  /channels/chan_sip.c,
+	  /main/features.c: Merge of several
+	  needed fixes for 1.8-digiumphones This merges fixes for the
+	  following issues into the 1.8-digiumphones branch: *
+	  ASTERISK-19355 - Call transfer with consultation frequently fails
+	  in cross- linked Asterisk scenario (directmedia & sendrpid
+	  active) * ASTERISK 19365 - Remote SIP Call legs are frequently
+	  not released in a cross-linked Asterisk scenario (directmedia &
+	  sendrpid) * ASTERISK-19183 - Sporadically missing connectedline
+	  event to caller channel in directed pickup app
+
+2012-04-09 20:40 +0000 [r361704]  Mark Michelson <mmichelson at digium.com>
+
+	* /apps/app_voicemail.c,
+	  /apps/app_voicemail.exports.in,
+	  /tests/test_voicemail_api.c (added),
+	  /include/asterisk/app_voicemail.h: Fix
+	  bugs in voicemail APIs and add unit tests. There were several
+	  crashes that could occur due to NULL inputs, invalid inputs, and
+	  the like. This fixes all known ones and adds unit tests to
+	  exercise the APIs.
+
+2012-04-06 19:08 +0000 [r361502]  Richard Mudgett <rmudgett at digium.com>
+
+	* /main/message.c: Update Func MESSAGE()
+	  and AMI MessageSend documentation. * Document
+	  MESSAGE(custom_data) * Update AMI MessageSend documentation *
+	  Eliminate a shadowed variable name in msg_func_write() for
+	  custom_data.
+
+2012-04-05 17:24 +0000 [r361283]  Mark Michelson <mmichelson at digium.com>
+
+	* /funcs/func_presence_state.c,
+	  /tests/test_config.c: Add additional
+	  configuration and presence unit tests. These were originally
+	  written while merging features into trunk, but these tests apply
+	  just as much for the 1.8 version of Digium phones, so might as
+	  well have them here, too.
+
+2012-04-03 21:03 +0000 [r361088]  Jonathan Rose <jrose at digium.com>
+
+	* /apps/app_mixmonitor.c: Make m option
+	  for mixmonitor delete the source file once it is finished copying
+	  to vm. Review: https://reviewboard.asterisk.org/r/1842/
+
+2012-03-29 21:49 +0000 [r360826]  Jason Parker <jparker at digium.com>
+
+	* /main/manager.c,
+	  ,
+	  /main/utils.c,
+	  /include/asterisk/manager.h,
+	  /apps/app_milliwatt.c: Multiple
+	  revisions 359656,359706,359979 ........ r359656 | mjordan |
+	  2012-03-15 13:35:59 -0500 (Thu, 15 Mar 2012) | 22 lines Fix
+	  remotely exploitable stack overrun in Milliwatt Milliwatt is
+	  vulnerable to a remotely exploitable stack overrun when using the
+	  'o' option. This occurs due to the milliwatt_generate function
+	  not accounting for AST_FRIENDLY_OFFSET when calculating the
+	  maximum number of samples it can put in the output buffer. This
+	  patch resolves this issue by taking into account
+	  AST_FRIENDLY_OFFSET when determining the maximum number of
+	  samples allowed. Note that at no point is remote code execution
+	  possible. The data that is written into the buffer is the
+	  pre-defined Milliwatt data, and not custom data. (closes issue
+	  ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt
+	  Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell
+	  Bryant (license 6283) Note that this patch was written by
+	  Russell, even though Matt uploaded it ........ Merged revisions
+	  359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+	  ........ r359706 | mjordan | 2012-03-15 14:01:22 -0500 (Thu, 15
+	  Mar 2012) | 16 lines Fix remotely exploitable stack overflow in
+	  HTTP manager There exists a remotely exploitable stack buffer
+	  overflow in HTTP digest authentication handling in Asterisk. The
+	  particular method in question is only utilized by HTTP AMI. When
+	  parsing the digest information, the length of the string is not
+	  checked when it is copied into temporary buffers allocated on the
+	  stack. This patch fixes this behavior by parsing out pre-defined
+	  key/value pairs and avoiding unnecessary copies to the stack.
+	  (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
+	  by: Matt Jordan ........ r359979 | rmudgett | 2012-03-20 12:21:16
+	  -0500 (Tue, 20 Mar 2012) | 28 lines Allow AMI action callback to
+	  be reentrant. Fix AMI module reload deadlock regression from
+	  ASTERISK-18479 when it tried to fix the race between calling an
+	  AMI action callback and unregistering that action. Refixes
+	  ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2
+	  object guaranteed that there were no active callbacks that
+	  mattered when ast_manager_unregister() was called. Unfortunately,
+	  this causes the deadlock situation. The patch stops locking the

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