[asterisk-commits] bebuild: tag 1.8.16.0-rc1 r370605 - /tags/1.8.16.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jul 31 08:25:18 CDT 2012


Author: bebuild
Date: Tue Jul 31 08:25:14 2012
New Revision: 370605

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=370605
Log:
Importing files for 1.8.16.0-rc1 release.

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+2012-07-31  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.16.0-rc1 Released.
+
+2012-07-30 16:47 +0000 [r370563]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_misdn.c: Release B channel allocation on error path
+	  in chan_misdn.
+
+2012-07-25 21:00 +0000 [r370494]  Jonathan Rose <jrose at digium.com>
+
+	* res/res_agi.c: res_agi: Add message indicating need for \n
+	  character in verbose message The while loop responsible for
+	  reading AGI messages from a fastAGI service can end up looping
+	  indefinitely when an AGI script fails to indicate the end of a
+	  message with a \n character. This patch adds an indication that
+	  we are expecting a \n character to end the message to make it
+	  more clear to users that this is necessary if they are receiving
+	  this warning over and over. (issue ASTERISK-20061) Reported by:
+	  Eike Kuiper
+
+2012-07-24 16:53 +0000 [r370429]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/frame.c: Rewrite a comment that didn't adequately explain
+	  the code it was documenting.
+
+2012-07-24 16:49 +0000 [r370428]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* channels/chan_oss.c: chan_oss: fix "sample rate" error message
+
+2012-07-23 21:09 +0000 [r370360-370383]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* funcs/func_shell.c: Improve documentation for the SHELL()
+	  dialplan function.
+
+	* main/channel.c: 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)
+
+2012-07-19 22:07 +0000 [r370275]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has problems
+	  casting away constness.
+
+2012-07-19 22:00 +0000 [r370252-370273]  Matthew Jordan <mjordan at digium.com>
+
+	* main/cel.c: 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
+
+	* res/res_rtp_asterisk.c: Handle extremely out of order RFC 2833
+	  DTMF The current implementation of RFC 2833 DTMF handling in
+	  res_rtp_asterisk will, if a packet arrives out of order, drop the
+	  packet. This is to prevent duplicate ton generation in the
+	  Asterisk core. Since the RTP layer does not buffer data itself,
+	  this is the only option the RTP layer currently has for handling
+	  packets that arrive out of order. For the most part, this doesn't
+	  matter. For a particular digit, so long as a BEGIN packet arrives
+	  before the first END packet, the digit will be produced. If
+	  subsequent BEGIN packets arrive interleaved with the ENDs, they
+	  will be dropped; likewise, if the BEGIN or END packets themselves
+	  are out of order, those packets are dropped but sufficient
+	  information is conveyed to the Asterisk core to produce the
+	  appropriate digit. For certain sequences of DTMF packets - most
+	  notably when, for a particular digit, an END packet arrives
+	  before any BEGIN packet for that digit - this is a real problem.
+	  When an END arrives before any BEGINs, the END packet is dropped
+	  - but at the same time, it causes subsequent BEGIN packets for
+	  that digit to be ignored. When the next in order END packet
+	  arrives, it too is dropped - Asterisk believes that there was no
+	  initial BEGIN. The solution this patch provides is to trust the
+	  END packet to convey the information needed for the Asterisk core
+	  to produce the DTMF digit. If we receive an END packet, and it: *
+	  Has a timestamp greater then the last timestamp received from an
+	  END packet * Does not have the same sequence number as the last
+	  received sequence number (and is thus not an END packet
+	  retransmission) Then we send the END frame up to the Asterisk
+	  core. It contains enough DTMF information for Asterisk to produce
+	  the digit. On the other hand, if we receive a BEGIN or
+	  continuation packet that occurs with a timestamp equal to or less
+	  then the last END timestamp, then we've received something out of
+	  order - but we already have received enough information to
+	  produce the digit. These packets are dropped. Much thanks goes to
+	  Olle Johansson (oej) for providing the idea for this solution.
+	  Review: https://reviewboard.asterisk.org/r/2033/ (issue
+	  ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt
+	  Jordan
+
+2012-07-18 19:12 +0000 [r370183-370205]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/cel.c: 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) Review:
+	  https://reviewboard.asterisk.org/r/2053/
+
+	* apps/app_macro.c, channels/chan_iax2.c, apps/app_mixmonitor.c,
+	  apps/app_stack.c, funcs/func_global.c, res/res_odbc.c,
+	  main/channel.c, addons/app_mysql.c, main/pbx.c,
+	  funcs/func_curl.c, main/ccss.c, funcs/func_odbc.c,
+	  funcs/func_lock.c: Ensure that all ast_datastore_info structures
+	  are 'const'. While addressing a bug, I came across a instance of
+	  'struct ast_datastore_info' that was not declared 'const'. Since
+	  the API already expects them to be 'const', this patch changes
+	  the declarations of all existing instances that were not already
+	  declared that way.
+
+2012-07-16 19:50 +0000 [r370131]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* channels/chan_sip.c: Code cleanup and bugfix in chan_sip
+	  outboundproxy parsing. The bug was clearing the global
+	  outboundproxy when a peer-specific outboundproxy was bad. The
+	  cleanup reduces duplicate code. Review:
+	  https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark
+	  Michelson
+
+2012-07-16 13:44 +0000 [r370081]  Kinsey Moore <kmoore at digium.com>
+
+	* UPGRADE.txt, CHANGES: Add comments about the BUILD_NATIVE change
+	  This is a significant change and mention of it should have gone
+	  into UPGRADE.txt and CHANGES.
+
+2012-07-12 20:15 +0000 [r370017]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c: Add missing
+	  ast_hangup() calls on some analog exception paths. Make starting
+	  analog_ss_thread() or __analog_ss_thread() failure paths hangup
+	  the channel.
+
+2012-07-12 20:05 +0000 [r369993-370014]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_sip.c: Include Expires header for SIP PUBLISH
+	  requests RFC3903 requres SIP PUBLISH requests to have Expires
+	  headers, so add them. Review:
+	  https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth
+
+	* channels/chan_sip.c: Prevent double uri_escaping in chan_sip when
+	  pedantic is enabled If pedantic mode is enabled, outbound invites
+	  will have double-escaped contacts. This avoids setting an
+	  already-escaped string into a field where it is expected to be
+	  unescaped. (closes issue ASTERISK-20023) Reported-by: Walter
+	  Doekes
+
+2012-07-12 14:23 +0000 [r369970]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* funcs/func_math.c: Correct Documentation For DEC Function The
+	  documentation for DEC in func_math.c was incorrect. Looks like a
+	  copy and paste error. (Closes issue ASTERISK-20095) Reported by:
+	  Billy Chia Tested by: Michael L. Young Patches: func_math.patch
+	  uploaded by Billy Chia (license 6381)
+
+2012-07-11 17:08 +0000 [r369937]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* funcs/func_realtime.c: Allow the REALTIME() function to report
+	  errors back to the caller. Also, do more error checking on the
+	  arguments specified to the REALTIME() function and clarify the
+	  documentation. While I was editing the file, a few coding
+	  guidelines fixups, as well. Review:
+	  https://reviewboard.asterisk.org/r/2031/
+
+2012-07-30  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.15.0 Released.
+
+2012-07-11  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.15.0-rc1 Released.
+
+2012-07-10 13:33 +0000 [r369869]  Kinsey Moore <kmoore at digium.com>
+
+	* apps/app_stack.c, main/pbx.c: Improve Goto and GotoIf related
+	  documentation Correct documentation on labeliftrue and
+	  labeliffalse parameters of GotoIf() and update several other
+	  locations that use the same syntax. (closes issue ASTERISK-20007)
+	  Patch-by: Leif Madsen Reported-by: WIMPy
+
+2012-07-09 17:05 +0000 [r369818]  Jason Parker <jparker at digium.com>
+
+	* configs/sip_notify.conf.sample: Add Digium phones context to
+	  sip_notify sample config. This makes it so that they can be
+	  reconfigured remotely. (closes issue ASTERISK-19910)
+
+2012-07-09 14:38 +0000 [r369792]  Jonathan Rose <jrose at digium.com>
+
+	* channels/chan_sip.c: chan_sip: Fix small behavioral change
+	  accidentally introduced in r369750 When removing the warning for
+	  AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed
+	  the return value, which would likely make the indication not be
+	  sent in audio. This fixes that while still removing the warning
+	  message.
+
+2012-07-06 20:54 +0000 [r369750]  Jonathan Rose <jrose at digium.com>
+
+	* channels/chan_sip.c: chan_sip: Add case for FLASH control frames
+	  so that we don't display a warning. chan_sip channels can receive
+	  flash control frames when connected to analog phones and possibly
+	  for other reasons. There really isn't a reason to warn when these
+	  frames are received, we can safely ignore them. Patches:
+	  dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182)
+
+2012-07-06 18:40 +0000 [r369708-369731]  Mark Michelson <mmichelson at digium.com>
+
+	* main/tcptls.c: 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
+
+	* main/bridging.c: Fix bridging thread leak. The bridge thread was
+	  exiting but was never being reaped using pthread_join(). This has
+	  been fixed now by calling pthread_join() in ast_bridge_destroy().
+	  (closes issue ASTERISK-19834) Reported by Marcus Hunger Review:
+	  https://reviewboard.asterisk.org/r/2012
+
+2012-07-05 19:01 +0000 [r369652]  Kinsey Moore <kmoore at digium.com>
+
+	* apps/app_voicemail.c: 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)
+
+2012-07-05 17:01 +0000 [r369626]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_sip.c: 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)
+
+2012-07-03 16:58 +0000 [r369557-369579]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: 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)
+
+	* channels/sip/include/sip.h, channels/chan_sip.c: 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/
+
+2012-06-29 16:52 +0000 [r369471-369490]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: With some configurations a transport is not
+	  actually specified so assume UDP in these cases.
+
+	* channels/chan_sip.c: Make the address family filter specific to
+	  the transport. (closes issue ASTERISK-16618) Reported by: Leif
+	  Madsen Review: https://reviewboard.asterisk.org/r/1667/
+
+2012-06-27 20:58 +0000 [r369436]  Terry Wilson <twilson at digium.com>
+
+	* channels/sip/include/sip.h, channels/chan_sip.c: 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
+
+2012-06-26 13:21 +0000 [r369366-369390]  Matthew Jordan <mjordan at digium.com>
+
+	* main/adsi.c: Fix crash in unloading of res_adsi module When
+	  res_adsi is unloaded, it removes the ADSI functions that it
+	  previously installed by passing a NULL adsi_funcs pointer to
+	  ast_adsi_install_funcs. This function was not checking whether or
+	  not the adsi_funcs pointer passed in was NULL before
+	  dereferencing it to check whether or not the version of the
+	  functions matches what the core was expecting it. This patch
+	  makes it so that the version is only checked if a potentially
+	  valid adsi_funcs pointer was passed in. Passing in NULL removes
+	  the installed functions, bypassing the version check.
+
+	* main/cdr.c: Tweak CDR change in r369351 As Tilghman pointed out
+	  on review 1996, the check to see if a CDR end time has been set
+	  is sufficient to know whether or not the duration value can be
+	  used. The check-in done for r369351 forgot to include this
+	  change.
+
+2012-06-25 19:13 +0000 [r369352]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/sip/include/sip.h, channels/chan_sip.c: Re-fix how local
+	  tag is generated when sending a 481 to an INVITE. Match our local
+	  tag to whatever to-tag was sent in the initial INVITE. Because
+	  the size of the to-tag may not fit in the buffer in the sip_pvt,
+	  it has been changed to a string field. (closes issue
+	  ASTERISK-19892) reported by Walter Doekes Review:
+	  https://reviewboard.asterisk.org/r/1977
+
+2012-06-25 19:12 +0000 [r369351]  Matthew Jordan <mjordan at digium.com>
+
+	* main/cdr.c: Fix incorrect duration reporting in CDRs created in
+	  batch mode Certain places in core/cdr.c would, if the duration
+	  value were 0, calculate the duration as being the delta between
+	  the current time and the time at which the CDR record was
+	  started. While this does not typically cause a problem in
+	  non-batch mode, this can cause an issue in batch mode where CDR
+	  records are gathered and written long after those calls have
+	  ended. In particular, this affects calls that were never
+	  answered, as those are expected to have a duration of 0. Often,
+	  this would result in CDR logs with a significant number of calls
+	  with lengthy durations, but dispositions of "BUSY". Note that
+	  this does not affect cdr_csv, as that backend does not use
+	  ast_cdr_getvar and instead directly reports the duration value.
+	  The affected core backends include cdr_apative_odbc and
+	  cdr_custom; other extended or deprecated CDR backends may
+	  potentially still directly manipulate the duration values. (issue
+	  ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883)
+	  Reported by: Thomas Arimont Tested by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/1996/
+
+2012-06-25 15:57 +0000 [r369327]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/features.c: Fix Bridge application occasionally returning to
+	  the wrong location. * Fix do_bridge_masquerade() getting the
+	  resume location from the zombie channel. The code must not touch
+	  a clone channel after it has masqueraded it. The clone channel
+	  has become a zombie and is starting to hangup. (closes issue
+	  ASTERISK-19985) Reported by: jamicque Patches:
+	  jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
+	  rmudgett Tested by: jamicque
+
+2012-06-25 15:50 +0000 [r369302-369324]  Mark Michelson <mmichelson at digium.com>
+
+	* main/adsi.c (added): Forgot to svn add this file in my last
+	  commit.
+
+	* res/res_adsi.exports.in (removed), include/asterisk/adsi.h,
+	  main/Makefile, res/res_adsi.c: Eliminate embedding of res_adsi.so
+	  module. The way this is done is to stop using the optional API.
+	  Instead, res_adsi.so, when loaded fills in a table of function
+	  pointers. Review: https://reviewboard.asterisk.org/r/1991
+
+	* channels/chan_sip.c: Be more consistent with the return code for
+	  requests received from invalid domain. When Asterisk receives an
+	  INVITE from an external domain when allowexternaldomains=no send
+	  a 403 instead of a 404. This is consistent with Asterisk's
+	  behavior when receiving a REGISTER in this situation. (Closes
+	  issue ASTERISK-19601) Reported by Matthew Jordan Patches:
+	  ASTERISK-19601-no401.patch uploaded by Mark Michelson (License
+	  #5049)
+
+2012-06-23 00:04 +0000 [r369235-369282]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/features.c: Fix Bridge application and AMI Bridge action
+	  error handling. * Fix AMI Bridge action disconnecting the AMI
+	  link on error. * Fix AMI Bridge action and Bridge application not
+	  checking if their masquerades were successful. * Fix Bridge
+	  application running the h-exten when it should not. * Made
+	  do_bridge_masquerade() return if the masquerade was successful so
+	  the Bridge application and AMI Bridge action could deal with it
+	  correctly. * Made bridge_call_thread_launch() hangup the passed
+	  in channels if the bridge_call_thread fails to start. Those
+	  channels would have been orphaned. * Made builtin_atxfer() check
+	  the success of the transfer masquerade setup.
+
+	* apps/app_queue.c: Explicitly check caller hangup in app Queue
+	  rather than a polluted res2 value.
+
+	* apps/app_dial.c: Check if PBX was started and fix F and F(x)
+	  action logic in Dial application.
+
+	* main/ccss.c: Check if PBX was started for generic CCSS recall.
+
+	* channels/chan_sip.c: Change incorrect chan_sip zombie hangup
+	  debug message. They are all zombies now.
+
+2012-06-22 19:28 +0000 [r369214]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Don't crash on a guest directmedia call A
+	  sip_pvt may not have relatedpeer set if a call doesn't match up
+	  with a peer. If there is no relatedpeer, there is no direct media
+	  ACL to apply, so just return that it is allowed. (closes issue
+	  ASTERISK-20040) Reported by: Terry Wilson
+
+2012-06-22 17:14 +0000 [r369195]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_sip.c: Don't parse media stream state for SIP video
+	  streams The sendonly/recvonly/sendrecv/inactive media stream
+	  attributes were parsed for video, but nothing was ever done with
+	  them. With this code removed, an UNSUPPORTED message is produced
+	  when these attributes are used in conjunction with a video stream
+	  which is the better behavior since they were never really
+	  supported in the first place.
+
+2012-06-20 17:33 +0000 [r369130-369146]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: fix
+	  locking issue on empty callList (issue ASTERISK-19298) Reported
+	  by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch
+
+	* addons/chan_ooh323.c: fix compile error (1.8 don't have
+	  ast_channel_name macro)
+
+2012-06-20 02:03 +0000 [r369108]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* include/asterisk/netsock2.h, main/netsock2.c: Fix NULL pointer
+	  segfault in ast_sockaddr_parse() While working with
+	  ast_parse_arg() to perform a validity check, a segfault occurred.
+	  The segfault occurred due to passing a NULL pointer to
+	  ast_sockaddr_parse() from ast_parse_arg(). According to the
+	  documentation in config.h, "result pointer to the result. NULL is
+	  valid here, and can be used to perform only the validity checks."
+	  This patch fixes the segfault by checking for a NULL pointer.
+	  This patch also adds documentation to netsock2.h about why it is
+	  necessary to check for a NULL pointer. (Closes issue
+	  ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael
+	  L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded
+	  by Michael L. Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/1990/
+
+2012-06-19 23:28 +0000 [r369090]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c: check rtptimeouts in ooh323 channels as per
+	  config file (rtp voice, video, udptl except rtcp) (closes issue
+	  ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
+	  19179-ooh323-2.patch
+
+2012-06-19 15:30 +0000 [r369066]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Fix request routing issue when outboundproxy
+	  is used. Asterisk was incorrectly setting the destination of
+	  CANCELs and ACKs for error responses to the URI of the initial
+	  INVITE. This resulted in further requests, such as INVITEs with
+	  authentication credentials, to be routed incorrectly. Instead,
+	  when these CANCEL or ACKs are to be sent, we should simply keep
+	  the destination the same as what it previously was. There is no
+	  need to alter it any. (closes issue ASTERISK-20008) Reported by
+	  Marcus Hunger Patches: ASTERISK-20008.patch uploaded by Mark
+	  Michelson (license #5049)
+
+2012-06-18 18:07 +0000 [r369043]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/features.c: Fix monitoring calls put in a parking lot. * Fix
+	  a regression that was introduced by -r366167 which effectively
+	  disabled monitoring parked calls. (closes issue ASTERISK-20012)
+	  Reported by: sdolloff Tested by: rmudgett
+
+2012-06-15 15:57 +0000 [r369001-369002]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* build_tools/find_missing_support_level (added): Add a script to
+	  enable finding source files without support-levels defined.
+
+	* main/devicestate.c, main/astfd.c, main/ssl.c,
+	  main/taskprocessor.c, main/astobj2.c, main/indications.c,
+	  main/config.c, main/loader.c, main/term.c, main/cli.c,
+	  channels/sig_analog.c, main/framehook.c, main/strcompat.c,
+	  main/plc.c, res/ais/evt.c, main/fskmodem_int.c, main/syslog.c,
+	  main/stdtime/localtime.c, main/db.c, main/bridging.c,
+	  channels/sig_ss7.c, main/datastore.c, main/sched.c,
+	  channels/sip/sdp_crypto.c, main/pbx.c, main/strings.c,
+	  channels/vcodecs.c, channels/iax2-provision.c, main/aoc.c,
+	  pbx/dundi-parser.c, main/cel.c, channels/iax2-parser.c,
+	  main/chanvars.c, main/netsock.c, main/data.c, main/srv.c,
+	  channels/chan_misdn.c, main/privacy.c,
+	  channels/sip/dialplan_functions.c, main/fixedjitterbuf.c,
+	  main/test.c, main/audiohook.c, main/alaw.c, main/asterisk.c,
+	  main/timing.c, main/global_datastores.c, main/fskmodem_float.c,
+	  main/ccss.c, channels/sip/reqresp_parser.c,
+	  channels/misdn/isdn_msg_parser.c, main/utils.c, main/xml.c,
+	  main/autochan.c, main/enum.c, channels/misdn/isdn_lib.c,
+	  main/fskmodem.c, channels/misdn_config.c, main/io.c,
+	  res/ael/pval.c, main/channel.c, main/cdr.c, main/ulaw.c,
+	  main/dial.c, main/tdd.c, main/heap.c, channels/console_gui.c,
+	  channels/misdn/ie.c, main/logger.c, channels/console_board.c,
+	  main/app.c, main/image.c, main/dns.c, main/lock.c, main/stun.c,
+	  main/dnsmgr.c, channels/sip/srtp.c, main/translate.c,
+	  main/slinfactory.c, main/jitterbuf.c, main/acl.c,
+	  channels/sig_pri.c, main/tcptls.c, main/hashtab.c,
+	  main/abstract_jb.c, main/callerid.c, main/file.c,
+	  res/snmp/agent.c, main/astmm.c, channels/misdn/portinfo.c,
+	  main/event.c, channels/sip/config_parser.c, channels/vgrabbers.c,
+	  main/xmldoc.c, main/dsp.c, main/udptl.c, main/netsock2.c,
+	  main/autoservice.c, main/rtp_engine.c, main/frame.c,
+	  main/security_events.c, res/ais/clm.c, main/threadstorage.c,
+	  main/say.c, channels/console_video.c: Add support-level
+	  indications to many more source files. Since we now have tools
+	  that scan through the source tree looking for files with specific
+	  support levels, we need to ensure that every file that is a
+	  component of a 'core' or 'extended' module (or the main Asterisk
+	  binary) is explicitly marked with its support level. This patch
+	  adds support-level indications to many more source files in tree,
+	  but avoids adding them to third-party libraries that are included
+	  in the tree and to source files that don't end up involved in
+	  Asterisk itself.
+
+2012-06-14 15:23 +0000 [r368898-368927]  Mark Michelson <mmichelson at digium.com>
+
+	* main/Makefile: Revert Makefile change to remove embedding
+	  res_adsi.so The change has resulted in a linking error for
+	  certain versions of GCC. This is much worse than the original
+	  issue, so for now, temporarily revert the change. A more thorough
+	  change will be sought out.
+
+	* funcs/func_volume.c: Fix a deadlock that occurs when func_volume
+	  is used on a local channel. This was discovered by trying to
+	  perform a call forward to an extension that makes use of
+	  func_volume. When the local channel is optimized away, the
+	  datastore on the local;2 channel would have its audiohook
+	  destroyed rather than detaching the audiohook from the channel
+	  and then destroying it. With this patch, func_volume's datastore
+	  destructor takes the proper route of detaching the audiohook and
+	  then destroying it. (closes issue ASTERISK-19611) reported by
+	  Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark
+	  Michelson (license #5049)
+
+2012-06-13 20:26 +0000 [r368894]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_smdi.c, res/res_adsi.c: Mark res_smdi/res_adsi as 'core'
+	  supported modules Recently, various issues surrounding weak
+	  symbols have caused problems with modules that rely on that
+	  feature to be enabled in menuselect. This includes app_voicemail
+	  and chan_dahdi, as they both rely upon res_smdi and res_adsi,
+	  which, in certain circumstances, may not be enabled by default in
+	  menuselect. Because res_smdi/res_adsi are dependencies for
+	  chan_dahdi/app_voicemail, this patch marks both as 'core'
+	  supported modules. This will allow both app_voicemail and
+	  chan_dahdi to be enabled as well, regardless of whether or not
+	  that system supports weak symbols. (issue AST-900) Reported by:
+	  Thomas Arimont (issue AST-885) Reported by: Denis Alberto
+	  Martinez
+
+2012-06-13 19:00 +0000 [r368873]  Mark Michelson <mmichelson at digium.com>
+
+	* main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+
+	  the result is that Asterisk has a phantom module loaded at
+	  startup, claiming to be res_adsi. (closes issue ASTERISK-19920)
+	  reported by Leif Madsen
+
+2012-06-13 14:27 +0000 [r368830-368852]  Matthew Jordan <mjordan at digium.com>
+
+	* Makefile: Do not install empty directories; add ASTLIBDIR r368830
+	  modified the installation script to only create a directory if
+	  that directory does not exist. If some directory variable was
+	  empty, it would attempt to create the empty location. It also
+	  failed to create the ASTLIBDIR directory. This patch fixes it
+	  such that the correct directories are made and only created if a
+	  value specifying them actually exists.
+
+	* Makefile: Do not perform install on existing directories If a
+	  directory already exists, performing a 'make install' will remove
+	  the permissions associated with the current directory and replace
+	  them with the permissions of the user executing the install. This
+	  patch changes this behavior to only perform an install on the
+	  directory if the directory does not exist. Thus, if a user later
+	  changes the permissions on that directory, those permissions will
+	  be preserved in subsequent installs. Review:
+	  https://reviewboard.asterisk.org/r/1986 Review:
+	  https://reviewboard.asterisk.org/r/1864 (closes issue
+	  ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger,
+	  Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded
+	  by mjordan)
+
+2012-06-12 15:36 +0000 [r368807]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Set the Caller ID "tag" on peers even if
+	  remote party information is present. On incoming calls, we were
+	  setting the cid_tag on the dialog only if there was no remote
+	  party information (Remote-Party-ID or P-Asserted-Identity)
+	  present. The Caller ID tag is an invented parameter, though, and
+	  should be set no matter the circumstance. (closes issue
+	  ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884)
+	  Reported by Trey Blancher
+
+2012-06-11 17:03 +0000 [r368759]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/channel.h, channels/chan_iax2.c, main/channel.c,
+	  channels/chan_dahdi.c, channels/sig_analog.c,
+	  channels/chan_sip.c: Fix deadlock potential with
+	  ast_set_hangupsource() calls. Calling ast_set_hangupsource() with
+	  the channel lock held can result in a deadlock because the
+	  function also locks the bridged channel. (issue ASTERISK-19537)
+	  (closes issue AST-891) Reported by: Guenther Kelleter Tested by:
+	  Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec
+	  Davis
+
+2012-06-11 15:13 +0000 [r368719-368738]  Kinsey Moore <kmoore at digium.com>
+
+	* apps/app_queue.c, main/loader.c, channels/chan_dahdi.c,
+	  res/res_config_odbc.c, channels/sip/dialplan_functions.c,
+	  pbx/pbx_config.c, apps/app_directory.c, res/res_odbc.c,
+	  res/res_speech.c, apps/app_voicemail.c, main/udptl.c,
+	  channels/sip/sdp_crypto.c, channels/chan_sip.c, res/res_fax.c,
+	  main/say.c, funcs/func_strings.c, channels/sip/reqresp_parser.c:
+	  Fix coverity UNUSED_VALUE findings in core support level files
+	  Most of these were just saving returned values without using them
+	  and in some cases the variable being saved to could be removed as
+	  well. (issue ASTERISK-19672)
+
+	* main/md5.c: Fix compilation in dev-mode Backport a compilation
+	  fix in md5.c from trunk that only showed up in dev-mode under
+	  certain compiler versions.
+
+2012-07-10  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.14.0 Released.
+
+2012-07-06  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.14.0-rc2 Released.
+
+	* AST-2012-010: Possible Resource Leak on Uncompleted Re-INVITE
+  	  transactions
+
+	* AST-2012-011: Remote Crash Vulnerability in VoiceMail Application
+
+	* Fix crash on a guest directmedia call
+
+	  A sip_pvt may not have relatedpeer set if a call doesn't match up
+	  with a peer.  If there is no relatedpeer, there is no direct media
+	  ACL to apply, so just return that is is allowed.
+
+	  (closes issue ASTERISK-20040)
+	
+	* Fix request routing issue when outboundproxy is used
+
+	  Asterisk was incorrectly setting the destination of CANCELs and ACKs
+	  for error responses to the URI of the initial INVITE.  This resulted
+	  in further requests, such as INVITEs with authentication
+	  credentials, to be routed incorrectly.  Instead when these CANCEL or
+	  ACKs are to be esnt, we should simply keep the destination the same
+	  as what it previously was.  There is no need to alter it any.
+
+	  (closes issue ASTERISK-20008)
+	
+	* Fix monitoring calls put in a parking lot
+
+	  Fix a regression that was introduced by r366167 which effectively
+	  disabled monitoring parked calls.
+
+	  (closes issue ASTERISK-20012)
+
+2012-06-08  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.14.0-rc1 Released.
+
+2012-06-06 21:27 +0000 [r368644]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, channels/sig_analog.c: Fix POTS flash hook
+	  to orignate a second call deadlock. A deadlock can occur when a
+	  POTS phone tries to flash hook to originate a second call for
+	  3-way or transfer. If another process is scanning the channels
+	  container when the POTS line flash hooks then a deadlock will
+	  occur. * Release the channel and private locks when creating a
+	  new channel as a result of a flash hook. (closes issue
+	  ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett
+
+2012-06-06 19:13 +0000 [r368625]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Fix a specific scenario where ACKs are not
+	  matched. If a dialog-starting INVITE contains a to-tag, then
+	  Asterisk will respond with a 481. In this case, the resulting
+	  incoming ACK would not be matched, so Asterisk would continue
+	  retransmitting the 481 until the transaction times out. There
+	  were two issues. Asterisk, upon creating a sip_pvt would generate
+	  a local tag. However, when the time came to transmit the 481,
+	  since there was a to-tag in the INVITE, Asterisk would place this
+	  original to-tag in the 481 response. When the ACK came in,
+	  Asterisk would attempt to match the to-tag in the ACK to the
+	  generated local tag. Unfortunately, Asterisk never actually
+	  transmitted a response with the generated local tag, so the
+	  to-tag in the ACK would not match. The other problem was that
+	  when the 481 was sent, nothing was set on the sip_pvt to indicate
+	  what CSeq is expected in the ACK. To fix the first problem, we
+	  zero out the to-tag seen in the incoming INVITE. This way,
+	  Asterisk, when time to send a response, will send its generated
+	  local tag instead. To fix the second problem, we set the
+	  sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
+	  481. (closes issue ASTERISK-19892) Reported by Mark Michelson
+
+2012-06-06 17:20 +0000 [r368604]  Matthew Jordan <mjordan at digium.com>
+
+	* build_tools/make_version: Add feature modifier to versions
+	  produced from branches Certain branches, such as Certified
+	  Asterisk, may have a modifier added to them that specifies the
+	  features available in that branch. For branches, this modifier is
+	  expected to be reflected in the location of the branch in
+	  subversion. For example, a subversion of URL of
+	  /certified/branches/1.8.11 would have a feature modifier of
+	  'certified'. This is slightly different then how features are
+	  determined for tags, where the feature is part of the actual tag
+	  name, e.g., "10.5.0-digiumphones". In keeping with the
+	  nomenclature used for tags, the feature specifier for branches is
+	  translated and placed after the revision numbers. For the example
+	  given previously, this would result in a branch version of
+	  "Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
+
+2012-06-06 16:07 +0000 [r368586]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_sip.c: Ensure overlapping hold flags do not

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