[asterisk-commits] bebuild: tag certified-1.8.15-cert2 r384098 - /certified/tags/1.8.15-cert2/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Mar 27 13:14:56 CDT 2013


Author: bebuild
Date: Wed Mar 27 13:14:52 2013
New Revision: 384098

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=384098
Log:
Importing files for 1.8.15-cert2 release.

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+2013-03-27  Asterisk Development Team
+
+	* Asterisk 1.8.15-cert2 Released.
+
+2013-03-27 15:19 +0000 [r383991]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/sip/include/sip.h, channels/chan_sip.c: AST-2013-003:
+	  Prevent username disclosure in SIP channel driver When
+	  authenticating a SIP request with alwaysauthreject enabled,
+	  allowguest disabled, and autocreatepeer disabled, Asterisk
+	  discloses whether a user exists for INVITE, SUBSCRIBE, and
+	  REGISTER transactions in multiple ways. The information is
+	  disclosed when: * A "407 Proxy Authentication Required" response
+	  is sent instead of a "401 Unauthorized" response * The presence
+	  or absence of additional tags occurs at the end of "403
+	  Forbidden" (such as "(Bad Auth)") * A "401 Unauthorized" response
+	  is sent instead of "403 Forbidden" response after a
+	  retransmission * Retransmission are sent when a matching peer did
+	  not exist, but not when a matching peer did exist. This patch
+	  resolves these various vectors by ensuring that the responses
+	  sent in all scenarios is the same, regardless of the presence of
+	  a matching peer. This issue was reported by Walter Doekes, OSSO
+	  B.V. A substantial portion of the testing and the solution to
+	  this problem was done by Walter as well - a huge thanks to his
+	  tireless efforts in finding all the ways in which this setting
+	  didn't work, providing automated tests, and working with Kinsey
+	  on getting this fixed. (closes issue ASTERISK-21013) Reported by:
+	  wdoekes Tested by: wdoekes, kmoore patches: AST-2013-003-1.8
+	  uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-10
+	  uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-11
+	  uploaded by kmoore, wdoekes (License 6273, 5674)
+
+2013-03-04 20:54 +0000 [r382389]  Jason Parker <jparker at digium.com>
+
+	* main/event.c: Fix comparison of presence state in event
+	  subsystem. Several new IEs were not given types (or names),
+	  causing the comparison function to improperly succeed. This adds
+	  those. (closes issue AST-1128)
+
+2013-02-28 16:49 +0000 [r382228]  Matthew Jordan <mjordan at digium.com>
+
+	* UPGRADE.txt, /, apps/app_meetme.c: Let channels joining a MeetMe
+	  conference opt out of the denoiser For some channel drivers,
+	  specifically those that have a varying rate in the number of
+	  audio samples, the audio quality for a MeetMe conference can be
+	  exceedingly poor. This is due to a unilateral application of the
+	  DENOISE function in func_speex to channels joining the
+	  conference. The denoiser function in the speex library is
+	  initialized with the number of audio samples in each sample that
+	  will be provided to it. If the number of audio samples changes,
+	  the denoiser has to be thrown away and re-initialized. While this
+	  could be worked around by removing func_speex, that doesn't help
+	  if you actually use the denoiser with other channels on the
+	  system. This patches does the following: * Checks for the
+	  presence of func_speex as opposed to codec_speex when determining
+	  if the DENOISE function is present (which is where the function
+	  is actually implemented) * Adds an option to MeetMe 'n' that
+	  causes the denoiser to not be applied to a channel when it joins.
+	  This keeps the current behavior the default, but let's users
+	  disable the denoiser if it causes problems on their system.
+	  Review: https://reviewboard.asterisk.org/r/2358 (closes issue
+	  AST-1062) Reported by: Thomas Arimont ........ Merged revisions
+	  382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-20 19:10 +0000 [r381834]  Matthew Jordan <mjordan at digium.com>
+
+	* apps/app_voicemail.c: Let vm_mailbox_snapshot_create's combine
+	  option apply to "Urgent" as well The vm_mailbox_snapshot_create
+	  function has an option that combines the contents of INBOX and
+	  Old into a single snapshot. The intent of this is that both 'new'
+	  messages and 'deleted' messages are given in a single snapshot,
+	  as some applications prefer this view of the voicemail world.
+	  Unfortunately, the initial implementation ignored the "Urgent"
+	  folder. The "Urgent" folder is a pseudo-INBOX, in that new
+	  messages left with the 'U' flag will be placed in that folder as
+	  opposed to INBOX. Thus, the option failed the intent with which
+	  it was added. This patch makes it so that the "Urgent" folder is
+	  included in the snapshot when that option is used.
+
+2013-01-14 15:17 +0000 [r379002]  David M. Lee <dlee at digium.com>
+
+	* /, channels/chan_sip.c: Fix XML encoding of 'identity display' in
+	  NOTIFY messages, continued. When r378933 was merged into 1.8, it
+	  should have also escaped remote_display, since it will have the
+	  same XML encoding problem when the caller/callee roles are
+	  reversed. (closes issue ABE-2902) Reported by: Guenther Kelleter
+	  ........ Merged revisions 379001 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-12 07:02 +0000 [r378936]  David M. Lee <dlee at digium.com>
+
+	* main/utils.c, include/asterisk/utils.h, /, channels/chan_sip.c,
+	  tests/test_xml_escape.c (added): Fix XML encoding of 'identity
+	  display' in NOTIFY messages. XML encoding in chan_sip is
+	  accomplished by naively building the XML directly from strings.
+	  While this usually works, it fails to take into account escaping
+	  the reserved characters in XML. This patch adds an
+	  'ast_xml_escape' function, which works similarly to
+	  'ast_uri_encode'. This is used to properly escape the
+	  local_display attribute in XML formatted NOTIFY messages. Several
+	  things to note: * The Right Thing(TM) to do would probably be to
+	  replace the ast_build_string stuff with building an ast_xml_doc.
+	  That's a much bigger change, and out of scope for the original
+	  ticket, so I refrained myself. * It is with great sadness that I
+	  wrote my own ast_xml_escape function. There's one in libxml2, but
+	  it's knee-deep in libxml2-ness, and not easily used to one-off
+	  escape a string. * I only escaped the string we know is causing
+	  problems (local_display). At least some of the other strings are
+	  URI-encoded, which should be XML safe. Rather than figuring out
+	  what's safe and escaping what's not, it would be much cleaner to
+	  simply build an ast_xml_doc for the messages and let the XML
+	  library do the XML escaping. Like I said, that's out of scope.
+	  (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by:
+	  Guenther Kelleter Review:
+	  http://reviewboard.digium.internal/r/365/ ........ Merged
+	  revision 378919 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  ........ Merged revisions 378933 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-09  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.15-cert1 Released.
+
+2013-01-02  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.15-cert1-rc3 Released.
+
+	* AST-2012-015: Resolve crashes due to large stack allocations when
+	  using TCP
+
+	  Asterisk had several places where messages received over various
+	  network transports may be copied in a single stack allocation. In
+	  the case of TCP, since multiple packets in a stream may be
+	  concatenated together, this can lead to large allocations that
+	  overflow the stack.
+
+	  This patch modifies those portions of Asterisk using TCP to either
+	  favor heap allocations or use an upper bound to ensure that the
+	  stack will not overflow:
+	   * For SIP, the allocation now has an upper limit
+	   * For HTTP, the allocation is now a heap allocation instead of a
+	     stack allocation
+	   * For XMPP (in res_jabber), the allocation has been eliminated
+	     since it was unnecesary.
+
+	* AST-2012-014: Prevent exhaustion of system resources through
+	  exploitation of event cache
+
+	  Asterisk maintains an internal cache for devices in the event
+	  subsystem. The device state cache holds the state of each device
+	  known to Asterisk, such that consumers of device state information
+	  can query for the last known state for a particular device, even if
+	  it is not part of an active call. The concept of a device in Asterisk
+	  can include entities that do not have a physical representation. One
+	  way that this occurred was when anonymous calls are allowed in
+	  Asterisk. A device was automatically created and stored in the cache
+	  for each anonymous call that occurred; this was possible in the SIP
+	  and IAX2 channel drivers and through channel drivers that utilized
+	  the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif).
+	  These devices are never removed from the system, allowing anonymous
+	  calls to potentially exhaust a system's resources.
+
+	  This patch changes the event cache subsystem and device state
+	  management to no longer cache devices that are not associated with a
+	  physical entity.
+
+2012-11-19   Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.15-cert1-rc2 Released.
+
+	* Fix SendDTMF crash and channel reference leak using channel name
+	  parameter.
+
+	  The SendDTMF channel name parameter has two issues.
+	  1) Crashes if the channel name does not exist.
+	  2) Leaks a channel reference if the channel is the current channel.
+	  Problem introduced by ASTERISK-15956.
+
+	  * Updated SendDTMF documentation.
+
+	  * Renamed app to senddtmf_name and tweaked the type.
+
+	* Fix Incrementing Sequence Number For Retransmitted DTMF End Packets
+
+	  In Asterisk 1.4+, a fix was put in place to increment the sequence
+	  number for retransmitted DTMF end packets.  With the introduction
+	  of the RTP engine API in 1.8, the sequence number was no longer being
+	  incremented.  This patch fixes this regression as well as cleans up a
+	  few lines that were not doing anything.
+
+	* app_queue: Fix a lock that was being held forever caused by a merging
+	  mistake
+
+    	  r375591 merged with conflicts and an oversight resulted in an unlock
+	  being missed which resulted in a deadlock when updating realtime
+	  members in queues.  This patch adds that unlock back in.
+
+2012-11-02  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.15-cert1-rc1 Released.
+
+2012-11-02 18:59 +0000 [r375630-375631]  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
+
+	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Multiple
+	  revisions 375519-375524 ........ r375519 | rmudgett | 2012-10-30
+	  16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines chan_misdn: Timer
+	  primitives must be handled first. The frm->addr is a different
+	  "address space" than the stack/instance address of other Lx
+	  primitives. The test for B channel instance address could fail.
+	  Patches: patch01_timers.diff (license #6372) patch uploaded by
+	  Guenther Kelleter JIRA ABE-2888 ........ r375520 | rmudgett |
+	  2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines
+	  chan_misdn: Free memory in error paths and other memory leaks.
+	  The one line commented with BUG is not easily fixable because
+	  there is no de-init function one can call. Patches:
+	  patch02_memory.diff (license #6372) patch uploaded by Guenther
+	  Kelleter JIRA ABE-2888 ........ r375521 | rmudgett | 2012-10-30
+	  16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines chan_misdn: ISDN NT
+	  L2 de-establish/establish * An NT-PTMP cannot de/establish L2
+	  since it doesn't know the TEIs. * On NT-PTP L2 is started when L1
+	  is finally active in handle_l1. * L2 deactivation logging
+	  cleanup. * L2 aggregate link status is unknown for NT-PTMP, show
+	  as "UNKN". * Removed unused functions and code for L2 handling.
+	  Patches: patch03_L2estab.diff (license #6372) patch uploaded by
+	  Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 |
+	  rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22
+	  lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH
+	  prim via lower_id layer (3 or 1) simply does not work. For TE (3)
+	  it returns an error (len=-6) which is not evaluated by
+	  handle_l1(), so the L1 layer status ends up wrong. Instead PH
+	  must be sent via L4, only then does it reach L1 without an error
+	  message. And NT PH prims only reach L1 when they are sent to
+	  layer 2 id. --> use upper_id to send PH primitives. * Check for
+	  errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are
+	  improved. * The lower_id is now not used for anything, except:
+	  Why is lower_id layer deleted when it wasn't created? I removed
+	  this code since it looks very wrong. Patches:
+	  patch04_l1activation.diff (license #6372) patch uploaded by
+	  Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett |
+	  2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
+	  chan_misdn: Fix loss of B channels if L1 is down. If you make 2
+	  calls out an NT PTMP port which is not connected to any phone,
+	  the B channel associated with that call becomes unusable until
+	  Asterisk is restarted. The problem is the EVENT_SETUP is queued
+	  when L1 is not up in misdn_lib_send_event(). If L1 cannot be
+	  activated the event won't be dequeued. It gets even worse when
+	  the call is hung up. The queued EVENT_SETUP will be overwritten
+	  by an EVENT_DISCONNECT. The reserved B channel then will never be
+	  freed. If later someone connects a phone to the port, L1 will
+	  eventually activate and the queued EVENT_DISCONNECT is sent down
+	  the stack. However, it is ignored because it is the wrong call
+	  state. The real fix would be that activation and queueing for a
+	  new SETUP is done by the NT stack. But since it doesn't, the
+	  workaround must be removed because it doesn't always work. Fix:
+	  The event is no longer queued but immediately sent to the stack.
+	  If L1 cannot be activated, the L3 state machine that was started
+	  by the EVENT_SETUP will do its work, i.e. a timeout will release
+	  the B channel properly. The SETUP possibly cannot be sent the
+	  first time but is resent by T303 in case L1 could be activated.
+	  Patches: patch05_bchan-loss.diff (license #6372) patch uploaded
+	  by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 |
+	  rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13
+	  lines chan_misdn: Remove some calls to exit(). Try proper cleanup
+	  when something goes wrong in misdn_lib_init(). Especially do not
+	  call exit()! * Fix memory leak because stack_destroy() does not
+	  free the stack struct. Patches: patch06_cleanup-init.diff
+	  (license #6372) patch uploaded by Guenther Kelleter Modified JIRA
+	  ABE-2888 ........ Merged revisions 375519-375524 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  ........ Merged revisions 375625 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-02 17:38 +0000 [r375584-375615]  Matthew Jordan <mjordan at digium.com>
+
+	* main/app.c, /: core: Fix a memory leak in app.c from an early
+	  return ast_app_group_match_get_count allocates memory with the
+	  regcomp function and we previously forgot to free it when bailing
+	  out due to a regex compilation failure against category. (closes
+	  issue AST-1018) Reported by: Guenther Kelleter Patches:
+	  regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372)
+	  ........ Merged revisions 375299 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* channels/chan_dahdi.c, /: chan_dahdi: Fix segfault dereferencing
+	  a NULL tech_pvt. The tech support customer was using the AMI
+	  Redirect action shortly after a call was placed. While the
+	  channel tried to do an ast_read(), the masquerade resulting from
+	  the channel redirect took place. The masquerade in the middle of
+	  the ast_read() resulted in the segfault. (closes issue AST-1025)
+	  Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch
+	  (license #5621) patch uploaded by rmudgett ........ Merged
+	  revisions 375361 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* UPGRADE.txt, apps/app_queue.c, /: Multiple revisions
+	  375216,375242 ........ r375216 | jrose | 2012-10-18 15:58:07
+	  -0500 (Thu, 18 Oct 2012) | 12 lines 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/ ........ 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 ........ Merged revisions
+	  375216,375242 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* configs/sip.conf.sample, channels/sip/include/sip.h,
+	  channels/sip/sdp_crypto.c, /, channels/chan_sip.c: Multiple
+	  revisions 372709,373165,373532,373652,374456 ........ r372709 |
+	  mjordan | 2012-09-08 20:19:21 -0500 (Sat, 08 Sep 2012) | 38 lines
+	  Only re-create an SRTP session when needed; respond with correct
+	  crypto policy In r356604, SRTP handling was fixed to accomodate
+	  multiple crypto keys in an SDP offer and the ability to re-create
+	  an SRTP session when the crypto keys changed. In certain
+	  circumstances - most notably when a phone is put on hold after
+	  having been bridged for a significant amount of time - the act of
+	  re-creating the SRTP session causes problems for certain models
+	  of phones. The patch committed in r356604 always re-created the
+	  SRTP session regardless of whether or not the cryptographic keys
+	  changed. Since this is technically not necessary, this patch
+	  modifies the behavior to only re-create the SRTP session if
+	  Asterisk detects that the remote key has changed. This allows
+	  models of phones that do not handle the SRTP session changing to
+	  continue to work, while also providing the behavior needed for
+	  those phones that do re-negotiate cryptographic keys. In
+	  addition, in Asterisk 1.8 only, it was found that phones that
+	  offer AES_CM_128_HMAC_SHA1_32 will end up with no audio if the
+	  phone is the initiator of the call. The phone will send an INVITE
+	  request specifying that AES_CM_128_HMAC_SHA1_32 be used for the
+	  cryptographic policy; Asterisk will set its policy to that value.
+	  Unfortunately, when the call is Answered and a 200 OK is sent
+	  back to the UA, the policy sent in the response's SDP will be the
+	  hard coded value AES_CM_128_HMAC_SHA1_80. This potentially
+	  results in Asterisk using the INVITE request's policy of
+	  AES_CM_128_HMAC_SHA1_32, while the phone uses Asterisk's response
+	  of AES_CM_128_HMAC_SHA1_80. Hilarity ensues as both endpoints
+	  think the other is crazy. This patch fixes that by caching the
+	  policy from the request and responding with it. Note that this is
+	  not a problem in Asterisk 10 and later, as the ability to
+	  configure the policy was added in that version. (issue
+	  ASTERISK-20194) Reported by: Nicolo Mazzon Tested by: Nicolo
+	  Mazzon Review: https://reviewboard.asterisk.org/r/2099 ........
+	  r373165 | file | 2012-09-19 11:02:18 -0500 (Wed, 19 Sep 2012) |
+	  10 lines Fix a regression where direct media was not permitted
+	  for calls using SIP INFO DTMF. A change was committed to fix
+	  direct media ACL support. This change wrongly assumed that only a
+	  single channel technology structure exists for chan_sip. This is
+	  in fact false as a second exists for calls using SIP INFO DTMF.
+	  The code which performs direct media ACL checking now checks for
+	  both the non-INFO DTMF and INFO DTMF channel technology
+	  structures. (closes issue ASTERISK-20409) Reported by: michele
+	  cicciotti privatewave ........ r373532 | file | 2012-09-24
+	  19:09:46 -0500 (Mon, 24 Sep 2012) | 5 lines Add missing checks
+	  that I neglected. The SIP technology and SIP info technology
+	  should be considered equal. (closes issue ASTERISK-20409)
+	  Reported by: michele cicciotti privatewave ........ r373652 |
+	  twilson | 2012-09-25 12:21:19 -0500 (Tue, 25 Sep 2012) | 18 lines
+	  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/ ........ r374456 | file
+	  | 2012-10-04 12:39:18 -0500 (Thu, 04 Oct 2012) | 14 lines Fix a
+	  regression from direct media ACLs where the directrtpsetup option
+	  no longer works. A check was added for direct media ACLs that
+	  immediately forbid remote bridging if there was no bridged
+	  channel. This caused directrtpsetup to no longer function as it
+	  needs this information before bridging actually occurs. Logic has
+	  now been adjusted so if there is no bridged channel a remote
+	  bridge will still be attempted. (closes issue ASTERISK-20511)
+	  Reported by: kristoff Review:
+	  https://reviewboard.asterisk.org/r/2146/ ........ Merged
+	  revisions 372709,373165,373532,373652,374456 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* funcs/func_math.c, main/xmldoc.c, apps/app_dial.c, /,
+	  channels/chan_sip.c: Multiple revisions
+	  371357,371469,371860,372628 ........ r371357 | jrose | 2012-08-16
+	  13:57:27 -0500 (Thu, 16 Aug 2012) | 8 lines chan_sip: Use pvt
+	  outgoing_call variable to set Remote-Party-ID Header Previously
+	  the pvt SIP_OUTGOING flag was used instead, which will frequently
+	  flip during reinvites. (closes issue AST-897) Reported by: Thomas
+	  Arimont ........ r371469 | mjordan | 2012-08-17 13:51:43 -0500
+	  (Fri, 17 Aug 2012) | 14 lines Fix memory leak in XML
+	  documentation When formatting documentation fields, the XML
+	  documentation parser calls xmldoc_get_formatted. This function
+	  allocates a string buffer at the beginning of its routine.
+	  Unfortunately, on certain code paths, it also calls
+	  xmldoc_string_cleanup, which assumes that it will create the
+	  string buffer. The previously allocated string buffer is then
+	  leaked by the xmldoc_string_cleanup routine. Now: we don't do
+	  that. (closes issue AST-932) Reported by: Alexander Homig
+	  ........ r371860 | rmudgett | 2012-08-29 13:22:24 -0500 (Wed, 29
+	  Aug 2012) | 12 lines Fix hangup cause passthrough regression. The
+	  v1.8 -r369258 change to fix the F and F(x) action logic
+	  introduced a regression in passing the hangup cause from the
+	  called channel to the caller channel. (closes issue
+	  ASTERISK-20287) Reported by: Konstantin Suvorov Patches:
+	  app_dial_hangupcause.patch (license #6421) patch uploaded by
+	  Konstantin Suvorov (modified) Tested by: rmudgett ........
+	  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 371357,371469,371860,372628
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c: chan_sip: Trigger reinvite if the SDP
+	  answer is included in the SIP ACK Under certain conditions, a SIP
+	  transaction involving directmedia wouldn't trigger a re-invite
+	  because the SDP answer was included in an ACK instead of in a
+	  message that we would have triggered the invite with. This patch
+	  just queues a source change control frame if the dialog is using
+	  directmedia when we find sdp for an ACK. (closes issue AST-913)
+	  Reported by: Thomas Arimont ........ Merged revisions 371337 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/cel.c, main/channel.c, /: Multiple revisions
+	  370205,370273,370360 ........ r370205 | kpfleming | 2012-07-18
+	  14:12:03 -0500 (Wed, 18 Jul 2012) | 18 lines Resolve severe
+	  memory leak in CEL logging modules. A customer reported a
+	  significant memory leak using Asterisk 1.8. They have tracked it
+	  down to ast_cel_fabricate_channel_from_event() in main/cel.c,
+	  which is called by both in-tree CEL logging modules (cel_custom.c
+	  and cel_sqlite3_custom.c) for each and every CEL event that they
+	  log. The cause was an incorrect assumption about how data
+	  attached to an ast_channel would be handled when the channel is
+	  destroyed; the data is now stored in a datastore attached to the
+	  channel, which is destroyed along with the channel at the proper
+	  time. (closes issue AST-916) Reported by: Thomas Arimont Review:
+	  https://reviewboard.asterisk.org/r/2053/ ........ r370273 |
+	  mjordan | 2012-07-19 17:00:14 -0500 (Thu, 19 Jul 2012) | 14 lines
+	  Fix compilation error when MALLOC_DEBUG is enabled To fix a
+	  memory leak in CEL, a channel datastore was introduced whose
+	  destruction function pointer was pointed to the ast_free macro.
+	  Without MALLOC_DEBUG enabled this compiles as fine, as ast_free
+	  is defined as free. With MALLOC_DEBUG enabled, however, ast_free
+	  takes on a definition from a different place then utils.h, and
+	  became undefined. This patch resolves this by using a reference
+	  to ast_free_ptr. When MALLOC_DEBUG is enabled, this calls
+	  ast_free; when MALLOC_DEBUG is not enabled, this is defined to be
+	  ast_free, which is defined to be free. (issue AST-916) Reported
+	  by: Thomas Arimont ........ r370360 | kpfleming | 2012-07-23
+	  09:41:03 -0500 (Mon, 23 Jul 2012) | 10 lines Free any datastores
+	  attached to dummy channels. Revision 370205 added the use of a
+	  datastore attached to a dummy channel to resolve a memory leak,
+	  but ast_dummy_channel_destructor() in this branch did not free
+	  datastores, resulting in a continued (but slightly smaller)
+	  memory leak. This patch backports the change to free said
+	  datastores from the Asterisk trunk. (related to issue AST-916)
+	  ........ Merged revisions 370205,370273,370360 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* configs/sip.conf.sample, channels/sip/include/sip.h, /,
+	  channels/chan_sip.c: Help mitigate potential reinvite glare
+	  scenarios. When Asterisk servers are set up back-to-back, and
+	  direct media is to be used betweeen endpoints, it is fairly
+	  common for the two Asterisk servers to send direct media
+	  reinvites to each other simultaneously. This results in 491s and
+	  ACKs being exchanged between the servers. While the media
+	  eventually gets set up properly, the problem is that there can be
+	  a noticeable delay for the streams to stabilize. This patch adds
+	  a new directmedia option called "outgoing". With this set, an
+	  immediate direct media reinvite will only be sent if the call
+	  direction is outgoing. For incoming dialogs, an immediate direct
+	  media reinvite will not be sent, but further "reactionary" direct
+	  media reinvites may be sent. For those who are having some deja
+	  vu, that's because this patch was originally committed to trunk
+	  since there is a new configuration option added. After seeing a
+	  bug report about audio being slow to set up on SIP calls, it
+	  became apparent that this patch would be the best solution for
+	  resolving the issue. The patch is unintrusive and will have no
+	  effect unless the option is explicitly enabled. (closes issue
+	  AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857)
+	  reported by Matt Jordan ........ Merged revisions 370618 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/ssl.c, main/tcptls.c, /, channels/chan_sip.c: 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
+	  373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-11 15:46 +0000 [r374848]  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 21:16 +0000 [r374807]  Richard Mudgett <rmudgett at digium.com>
+
+	* 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 20:12 +0000 [r374569]  dlee <dlee at localhost>:
+
+	* main/manager.c: Improve AMI long line error handling In AMI's
+	  parser, when it receives a long line (> 1024 characters), it
+	  discards that line, but continues to process the message
+	  normally. Typically, this is not a problem because a) who has
+	  lines that long and b) usually a discarded line results in an
+	  invalid message. But if that line is specifying an optional
+	  field, then the message will be processed, you get a 'Response:
+	  Success', but things don't work the way you expected them to.
+	  This patch changes the behavior when a line-too-long parse error
+	  occurs. * Changes the log message to avoid way-too-long (and
+	  truncated anyways) log messages * Adds a 'parsing' status flag to
+	  Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if,
+	  well, a line is too long * Responds with an appropriate error if
+	  parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John
+	  Bigelow Review: https://reviewboard.asterisk.org/r/2142/
+
+2012-10-05 19:02 +0000 [r374541]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
+	  channels/misdn/isdn_lib.h, channels/chan_misdn.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

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