Author: bebuild
Date: Mon Mar 23 11:43:24 2015
New Revision: 433308

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=433308
Log:
Importing files for 11.17.0-rc1 release.

Added:
    tags/11.17.0-rc1/.lastclean   (with props)
    tags/11.17.0-rc1/.version   (with props)
    tags/11.17.0-rc1/ChangeLog   (with props)

Added: tags/11.17.0-rc1/.lastclean
URL: 
http://svnview.digium.com/svn/asterisk/tags/11.17.0-rc1/.lastclean?view=auto&rev=433308
==============================================================================
--- tags/11.17.0-rc1/.lastclean (added)
+++ tags/11.17.0-rc1/.lastclean Mon Mar 23 11:43:24 2015
@@ -1,0 +1,1 @@
+40

Propchange: tags/11.17.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/11.17.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/11.17.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/11.17.0-rc1/.version
URL: 
http://svnview.digium.com/svn/asterisk/tags/11.17.0-rc1/.version?view=auto&rev=433308
==============================================================================
--- tags/11.17.0-rc1/.version (added)
+++ tags/11.17.0-rc1/.version Mon Mar 23 11:43:24 2015
@@ -1,0 +1,1 @@
+11.17.0-rc1

Propchange: tags/11.17.0-rc1/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/11.17.0-rc1/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/11.17.0-rc1/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/11.17.0-rc1/ChangeLog
URL: 
http://svnview.digium.com/svn/asterisk/tags/11.17.0-rc1/ChangeLog?view=auto&rev=433308
==============================================================================
--- tags/11.17.0-rc1/ChangeLog (added)
+++ tags/11.17.0-rc1/ChangeLog Mon Mar 23 11:43:24 2015
@@ -1,0 +1,33413 @@
+2015-03-23  Asterisk Development Team <asteriskt...@digium.com>
+
+       * Asterisk 11.17.0-rc1 Released.
+
+2015-03-22 23:55 +0000 [r433245-433268]  Matthew Jordan <mjor...@digium.com>
+
+       * apps/app_queue.c: Fix compilations errors on 64-bit OpenBSD
+         systems In versiong 5.5, OpenBSD went to 64-bit time values. This
+         requires a cast to (long) when printing members of certain time
+         structs. Review: https://reviewboard.asterisk.org/r/4507
+         ASTERISK-24879 #close Reported by: snuffy Tested by: snuffy
+         patches: openbsd-time64.diff uploaded by snuffy (License 5024)
+
+       * main/asterisk.c, main/xmldoc.c: Fix compilation issues for
+         OpenBSD This patch addresses compilation issues for OpenBSD.
+         Specifically, it addresses: * It allows including <sys/vmmeter.h>
+         in asterisk.c * Provides a needed (size_t) cast in xmldoc.c In
+         13+, it also addresses a conditional inclusion in loader.c.
+         Review: https://reviewboard.asterisk.org/r/4506 ASTERISK-24880
+         #close Reported by: snuffy Tested by: snuffy patches:
+         misc-openbsd.diff uploaded by snuffy (License 5024)
+
+2015-03-19 19:19 +0000 [r433173]  Matthew Jordan <mjor...@digium.com>
+
+       * tests/test_func_file.c, funcs/func_env.c: funcs/func_env: Fix
+         regression caused in FILE read operation When r432935 was merged,
+         it did correctly fix a situation where a FILE read operation on
+         the middle of a file buffer would not read the requested length
+         in the parameters passed to the FILE function. Unfortunately, it
+         would also allow the FILE function to append more bytes than what
+         was available in the buffer if the length exceeded the end of the
+         buffer length. This patch takes the minimum of the remaining
+         bytes in the buffer along with the calculated length to append
+         provided by the original patch, and uses that as the length to
+         append in the return result. This patch also updates the unit
+         tests with the scenarios that were originally pointed out in
+         ASTERISK-21765 that the original implementation treated
+         incorrectly. ASTERISK-21765
+
+2015-03-19 10:19 +0000 [r433112-433122]  Corey Farrell <g...@cfware.com>
+
+       * main/logger.c: logger: Apply default console logging when
+         configuration cannot be loaded. When logger.conf is missing or
+         invalid enable console logging and display an error message.
+         ASTERISK-24817 #close Reported by: Corey Farrell Review:
+         https://reviewboard.asterisk.org/r/4497/
+
+       * channels/chan_sip.c: chan_sip: Fix dialog reference leaked to
+         scheduler for reinvite_timeout. Release the scheduler reference
+         to the dialog for reinvite timeout during dialog_unlink_all.
+         ASTERISK-24876 #close Reported by: Corey Farrell Review:
+         https://reviewboard.asterisk.org/r/4491/
+
+2015-03-17 22:28 +0000 [r433086]  Scott Griepentrog <sgriepent...@digium.com>
+
+       * main/utils.c, main/asterisk.c, main/xmldoc.c: Various: backport
+         of bugfixes found via chaos Using DEBUG_CHAOS several instances
+         of a null pointer crash, and one uninitialized variable were
+         uncovered and fixed. Also added details on why Asterisk failed to
+         initialize. This is a backport of the fixes from Asterisk 13.
+         Review: https://reviewboard.asterisk.org/r/4468/
+
+2015-03-17 21:43 +0000 [r433056]  Richard Mudgett <rmudg...@digium.com>
+
+       * apps/app_externalivr.c, main/netsock2.c: Audit
+         ast_sockaddr_resolve() usage for memory leaks. Valgrind found
+         some memory leaks associated with ast_sockaddr_resolve(). Most of
+         the leaks had already been fixed by earlier memory leak hunt
+         patches. This patch performs an audit of ast_sockaddr_resolve()
+         and found one more. * Fix ast_sockaddr_resolve() memory leak in
+         apps/app_externalivr.c:app_exec(). * Made
+         main/netsock2.c:ast_sockaddr_resolve() always set the addrs
+         parameter for safety so the pointer will never be uninitialized
+         on return. The same goes for
+         res/res_pjsip_acl.c:extract_contact_addr(). Review:
+         https://reviewboard.asterisk.org/r/4509/
+
+2015-03-14 02:27 +0000 [r432970]  Matthew Jordan <mjor...@digium.com>
+
+       * main/frame.c: main/frame: Don't report empty disallow values as
+         an error In realtime, it is normal to have a database with both
+         'allow' and 'disallow' columns in the schema. It is perfectly
+         valid to have an 'allow' value of '!all,g722,ulaw,alaw' and no
+         'disallow' value. Unlike in static conf files, you can't *not*
+         provide the disallow value. Thus, the empty disallow value causes
+         a spurious WARNING message, which is kind of annoying. This patch
+         makes it so that a 'disallow' value with no ... value ... is
+         ignored. Granted, you can still screw this up as well, as
+         technically specifying 'disallow=all,!ulaw' allows only ulaw, and
+         then you would have no 'allow' value in your database. But
+         really, why would you do that? WHY? ASTERISK-16779 #close
+         Reported by: Atis Lezdins
+
+2015-03-14 01:59 +0000 [r432944-432948]  Joshua Colp <jc...@digium.com>
+
+       * funcs/func_curl.c: func_curl: Don't hold exclusive lock when
+         performing HTTP request. This code originally kept a lock held
+         when performing the HTTP request to ensure that the options
+         provided to curl remain valid. This doesn't seem to be necessary
+         these days and holding the lock caused requests to happen
+         sequentially instead of in parallel. ASTERISK-18708 #close
+         Reported by: Dave Cabot
+
+       * main/cli.c: core: Fix tab completion of "core set debug channel"
+         CLI command. The "core set debug channel" CLI command mistakenly
+         had source filenames added to its tab completion. This occurred
+         because the CLI generator fell back to the "core set debug"
+         command which permits setting debug at a source filename level.
+         ASTERISK-21038 #close Reported by: Richard Kenner
+
+2015-03-14 01:21 +0000 [r432918-432935]  Matthew Jordan <mjor...@digium.com>
+
+       * funcs/func_env.c: FILE: fix retrieval of file contents when
+         offset is specified The loop that reads in a file was not
+         correctly using the offset when determining what bytes to append
+         to the output. This patch corrects the logic such that the
+         correct portion of the file is extracted when an offset is
+         specified. ASTERISK-21765 Reported by: John Zhong Tested by: Matt
+         Jordan, Di-Shi Sun patches: file_read_390821.patch uploaded by
+         Di-Shi Sun (License 5076)
+
+       * apps/app_amd.c, configs/amd.conf.sample: apps/app_amd: Document
+         maximum_word_length option; fix AMDCAUSE documentation This patch
+         corrects the documentation for the AMD application. Specifically:
+         * It documents the maximum_word_length option, which limits the
+         maximum allowed length of a single utterance. * It clarifies the
+         AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH was
+         documented as MAXWORDS, while MAXWORDS was undocumented. Thanks
+         to the issue reporter, Frank DiGennaro, for pointing out the
+         issues. ASTERISK-19470 #close Reported by: Frank DiGennaro
+
+2015-03-12 12:57 +0000 [r432807-432810]  Matthew Jordan <mjor...@digium.com>
+
+       * main/audiohook.c: main/audiohook: Update internal sample rate on
+         reads When an audiohook is created (which is used by the various
+         Spy applications and Snoop channel in Asterisk 13+), it initially
+         is given a sample rate of 8kHz. It is expected, however, that
+         this rate may change based on the media that passes through the
+         audiohook. However, the read/write operations on the audiohook
+         behave very differently. When a frame is written to the
+         audiohook, the format of the frame is checked against the
+         internal sample rate. If the rate of the format does not match
+         the internal sample rate, the internal sample rate is updated and
+         a new SLIN format is chosen based on that sample rate. This works
+         just fine. When a frame is read, however, we do something quite
+         different. If the format rate matches the internal sample rate,
+         all is fine. However, if the rates don't match, the audiohook
+         attempts to "fix up" the number of samples that were requested.
+         This can result in some seriously large number of samples being
+         requested from the read/write factories. Consider the worst case
+         - 192kHz SLIN. If we attempt to read 20ms worth of audio produced
+         at that rate, we'd request 3840 samples (192000 / (1000 / 20)).
+         However, if the audiohook is still expecting an internal sample
+         rate of 8000, we'll attempt to "fix up" the requested samples to:
+         samples_converted = samples * (ast_format_get_sample_rate(format)
+         / (float) audiohook->hook_internal_samp_rate); which is: 92160 =
+         3840 * (192000 / 8000) This results in us attempting to read
+         92160 samples from our factories, as opposed to the 3840 that we
+         actually wanted. On a 64-bit machine, this miraculously survives
+         - despite allocating up to two buffers of length 92160 on the
+         stack. The 32-bit machines aren't quite so lucky. Even in the
+         case where this works, we will either (a) get way more samples
+         than we wanted; or (b) get about 3840 samples, assuming the
+         timing is pretty good on the machine. Either way, the calculation
+         being performed is wrong, based on the API users expectations. My
+         first inclination was to allocate the buffers on the heap. As it
+         is, however, there's at least two drawbacks with doing this: (1)
+         It's a bit complicated, as the size of the buffers may change
+         during the lifetime of the audiohook (ew). (2) The stack is
+         faster (yay); the heap is slower (boo). Since our calculation is
+         flat out wrong in the first place, this patch fixes this issue by
+         instead updating the internal sample rate based on the format
+         passed into the read operation. This causes us to read the
+         correct number of samples, and has the added benefit of setting
+         the audihook with the right SLIN format. Note that this issue was
+         caught by the Asterisk Test Suite as a result of r432195 in the
+         13 branch. Because this issue is also theoretically possible in
+         Asterisk 11, the change is being made here as well. Review:
+         https://reviewboard.asterisk.org/r/4475/
+
+       * makeopts.in, Makefile, include/asterisk/utils.h, configure,
+         main/Makefile, configure.ac, include/asterisk/inline_api.h: Add
+         support for the clang compiler; update RAII_VAR to use
+         BlocksRuntime RAII_VAR, which is used extensively in Asterisk to
+         manage reference counted resources, uses a GCC extension to
+         automatically invoke a cleanup function when a variable loses
+         scope. While this functionality is incredibly useful and has
+         prevented a large number of memory leaks, it also prevents
+         Asterisk from being compiled with clang. This patch updates the
+         RAII_VAR macro such that it can be compiled with clang. It makes
+         use of the BlocksRuntime, which allows for a closure to be
+         created that performs the actual cleanup. Note that this does not
+         attempt to address the numerous warnings that the clang compiler
+         catches in Asterisk. Much thanks for this patch goes to: * The
+         folks on StackOverflow who asked this question and Leushenko for
+         providing the answer that formed the basis of this code:
+         
http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang
+         * Diederik de Groot, who has been extremely patient in working on
+         getting this patch into Asterisk. Review:
+         https://reviewboard.asterisk.org/r/4370/ ASTERISK-24133
+         ASTERISK-23666 ASTERISK-20399 ASTERISK-20850 #close Reported by:
+         Diederik de Groot patches: RAII_CLANG.patch uploaded by Diederik
+         de Groot (License 6600)
+
+2015-03-10 21:32 +0000 [r432691-432720]  Matthew Jordan <mjor...@digium.com>
+
+       * res/res_config_odbc.c: res/res_config_odbc: Fix improper escaping
+         of backslashes with MySQL When escaping backslashes with MySQL,
+         the proper way to escape the characters in a LIKE clause is to
+         escape the '\' four times, i.e., '\\\\'. To quote the MySQL
+         manual: "Because MySQL uses C escape syntax in strings (for
+         example, “\n” to represent a newline character), you must double
+         any “\” that you use in LIKE strings. For example, to search for
+         “\n”, specify it as “\\n”. To search for “\”, specify it 
as
+         “\\\\”; this is because the backslashes are stripped once by the
+         parser and again when the pattern match is made, leaving a single
+         backslash to be matched against." ASTERISK-24808 #close Reported
+         by: Javier Acosta patches: res_config_odbc.diff uploaded by
+         Javier Acosta (License 6690)
+
+       * apps/app_voicemail.c: app_voicemail: Fix crash with IMAP backends
+         when greetings aren't present When an IMAP backend is in use and
+         greetings are set to be used, but aren't present for a user in
+         their IMAP folder, Asterisk will crash. This occurs due to the
+         mailstream being set to the 'greetings' folder and being left in
+         that particular state, regardless of the success/failure of the
+         attempt to access the folder the mailstream points to. Later
+         access of the mailstream assumes that it points to the 'INBOX'
+         (or some other folder), resulting in either a crash (if the
+         greetings folder didn't exist and the mailstream is invalid) or
+         an inability to read messages from the 'INBOX' folder. This patch
+         restores the mailstream to its correct state after accessing the
+         greetings. This fixes the crash, and sets the mailstream to the
+         state that VoiceMailMain expects. Note that while ASTERISK-23390
+         also contained a patch for this issue, the patch on
+         ASTERISK-24786 is the one being merged here. Review:
+         https://reviewboard.asterisk.org/r/4459/ ASTERISK-23390 #close
+         Reported by: Ben Smithurst ASTERISK-24786 #close Reported by:
+         Graham Barnett Tested by: Graham Barnett patches:
+         app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett
+         (License 6685)
+
+       * main/stdtime/localtime.c: localtime: Fix file descriptor leak on
+         kqueue(2) systems The localtime management in the Asterisk core
+         contains a thread that watches for changes in the local timezone.
+         On systems where the directory containing /etc/localtime is
+         modified frequently, the thread monitoring the changes will be
+         woken up to determine if any changes in timezone have occurred.
+         When using kqueue(2), this can cause a leak of file descriptors
+         due to some improper management of resources. This patch updates
+         the kqueue(2) handling in localtime, such that is no longer leaks
+         resources. Review: https://reviewboard.asterisk.org/r/4450/
+         ASTERISK-24739 #close Reported by: Ed Hynan patches:
+         11.15.0-u.diff uploaded by Ed Hynan (Licnese 6680) 11.7.0-u.diff
+         uploaded by Ed Hynan (License 6680) svn-trunk-Jan-26-2015-u.diff
+         uploaded by Ed Hynan (License 6680)
+
+2015-03-06 19:52 +0000 [r432526-432530]  Richard Mudgett <rmudg...@digium.com>
+
+       * channels/chan_dahdi.c, channels/sig_analog.c,
+         channels/sig_analog.h, UPGRADE.txt: chan_dahdi/sig_analog: Fix
+         distinctive ring detection to suck less. The distinctive ring
+         feature interferes with detecting Caller ID and appears to have
+         been broken for years. What happens is if you have a ring-ring
+         cadence as used in the UK you get too many DAHDI events for the
+         distinctive ring pattern array and Caller ID detection is
+         aborted. I think when Zapata/DAHDI added the ring begin event it
+         broke distinctive ring. More events happen than before and the
+         code does no filtering of which event times are recorded in the
+         pattern array. * Made distinctive ring only record the ringt
+         count when the ring ends instead of on just any DAHDI event.
+         Distinctive ring can be ring, ring-ring, ring-ring-ring, or
+         different ring durations for the up to three rings. * Fixed the
+         distinctive ring detection enable (chan_dahdi.conf option
+         usedistinctiveringdetection) to be per port instead of somewhat
+         per port and somewhat global. This has been broken since v1.8. *
+         Fixed using the default distinctive ring context when the
+         detected pattern does not match any configured dringX patterns.
+         The default context did not get set when the previous call was a
+         matched distinctive ring pattern and the current call is not
+         matched. This has been broken since v1.8. * Made distinctive ring
+         have no effect on Caller ID detection when it is disabled. Caller
+         ID detection just monitors for 10 seconds before giving up. *
+         Fixed leak of struct callerid_state memory when a polarity
+         reversal during Caller ID detection causes the incoming call to
+         be aborted. DAHDI-1143 AST-1545 ASTERISK-24825 #close Reported
+         by: Richard Mudgett ASTERISK-17588 Reported by: Daniel Flounders
+         Review: https://reviewboard.asterisk.org/r/4444/
+
+       * channels/chan_sip.c: chan_sip: Fix realtime locking inversion
+         when poking a just built peer. When a realtime peer is built it
+         can cause a locking inversion when the just built peer is poked.
+         If the CLI command "sip show channels" is periodically executed
+         then a deadlock can happen because of the locking inversion. *
+         Push the peer poke off onto the scheduler thread to avoid the
+         locking inversion of the just built realtime peer. AST-1540
+         ASTERISK-24838 #close Reported by: Richard Mudgett Review:
+         https://reviewboard.asterisk.org/r/4454/
+
+2015-03-05 16:35 +0000 [r432484]  George Joseph <george.jos...@fairview5.com>
+
+       * apps/app_voicemail.c: app_voicemail: Fix compile breaking in
+         app_voicemail with IMAP_STORAGE. There is a leftover "assert" in
+         app_voicemail/__messagecount that references variables that don't
+         exist. This causes the compile to fail when --enable-dev-mode and
+         IMAP_STORAGE are selected. This patch removes the assert.
+         Tested-by: George Joseph Review:
+         https://reviewboard.asterisk.org/r/4461/
+
+2015-02-26 17:06 +0000 [r432362]  Kevin Harwell <kharw...@digium.com>
+
+       * apps/app_chanspy.c, main/channel.c: app_chanspy, channel: fix
+         frame leaks Fixed a couple of frame leaks that were found during
+         testing. ASTERISK-24828 #close Reported by: John Hardin Review:
+         https://reviewboard.asterisk.org/r/4445/
+
+2015-02-26 04:56 +0000 [r432239-432341]  Matthew Jordan <mjor...@digium.com>
+
+       * channels/Makefile, apps/Makefile: make: Remove 'res_features'
+         from libraries to link against with cygwin/mingw32 Both the apps
+         and channels Makefiles still listed 'res_features' as modules to
+         link against when compiling for cygwin or mingw32. This module
+         hasn't existed for quite some time. ASTERISK-18105 #close
+         Reported by: feyfre
+
+       * channels/chan_sip.c: channels/chan_sip: Don't send a BYE after
+         final response when PBX thread fails When Asterisk fails to start
+         a PBX thread for a new channel - for example, when the maxcalls
+         setting in asterisk.conf is exceeded - we currently send a final
+         response, and then attempt to send a BYE request to the UA. Since
+         that's all sorts of wrong, this patch fixes that by setting
+         sipalreadygone on the sip_pvt such that we don't get stuck
+         sending BYE requests to something that does not want it. Note
+         that this patch is a slight modification of the one on
+         ASTERISK-15434. For clarity, it explicitly calls sipalreadygone
+         with the calls to transmit a final response. ASTERISK-21845
+         ASTERISK-15434 #close Reported by: Makoto Dei Tested by: Matt
+         Jordan patches: sip-pbxstart-failed.patch uploaded by Makoto Dei
+         (License 5027)
+
+       * configure, configure.ac: configure: Promote SQLite3 "not
+         installed" warning to error Since Asterisk won't build without
+         the library, not having it is definitely an error. Thanks to Kyle
+         Kurz for pointing this out.
+
+       * channels/chan_sip.c: channels/chan_sip: Clarify WARNING message
+         in mismatched SRTP scenario When we receive an SDP as part of an
+         offer/answer for a peer/friend has been configured to require
+         encryption, and that SDP offer/answer failed to provide
+         acceptable crypto attributes, we currently issue a WARNING that
+         uses the phrase "we" and "requested". In this case, both of those
+         terms are ambiguous - the user will probably think "we" is
+         Asterisk (it most likely isn't) and it may not be a "request", so
+         much as an SDP that was received in some fashion. This patch
+         makes the WARNING messages slightly less bad and a bit more
+         accurate as well. ASTERISK-23214 #close Reported by: Rusty Newton
+
+       * channels/sip/sdp_crypto.c: channels/sip/sdp_crypto: Handle SRTP
+         keys negotiated with key lifetime/MKI Prior to this patch, SDP
+         offers negotiating SDES-SRTP crypto attributes would be rejected
+         if those crypto attributes contained either a key lifetime or a
+         MKI parameter. While from a theoretical point of view this was
+         defensible - Asterisk does not support key lifetimes or multiple
+         crypto keys - from a practical point of view, this is quite a
+         problem. A large number of endpoints offer lifetimes/MKI, which
+         Asterisk can tolerate so long as it doesn't actually have to
+         support anything more than a single key or refresh the key. In
+         reality, this is (so far as we've seen) always the case. This
+         patch is a forward port of Olle's work in the
+         lingon-srtp-key-lifetime-1.8 branch. To quote Olle from
+         ASTERISK-17721, it handles lifetime/MKI parameters in the
+         following fashion: > The Lingon branch now handle lifetime and
+         MKI parameters. > > We only accept lifetimes up to max for the
+         crypto and higher than 10 hours > for packetization of 20 ms (50
+         pps). > > We only handle MKI with index 1. > > We do not really
+         bother with counting packets and reinviting at end of > lifetime,
+         so the min of 10 hours kind of takes care of most calls. If there
+         > are longer ones, we rely on the other side for re-invites. > >
+         It's still not perfect, but I personally think this is an
+         improvement. A > configuration option for minimum lifetime
+         accepted could be added. When the patch was ported forward, I
+         decided against adding a configuration option as Olle's handling
+         was more than sufficient for every case I've seen come through
+         the issue tracker or through interoperability testing. We can
+         revisit that decision if it proves to be false. A few small other
+         tweaks were made to the surrounding code to reduce indentation
+         and provide better type safety for the 'tag' parameter. Review:
+         https://reviewboard.asterisk.org/r/4419/ Review:
+         https://reviewboard.asterisk.org/r/4418/ ASTERISK-17721 #close
+         Reported by: Terry Wilson ASTERISK-17899 #close Reported by:
+         Dwayne Hubbard patches: lingon-srtp-key-lifetime-1.8.diff
+         uploaded by oej (License 5267) ASTERISK-20233 Reported by: tootai
+         ASTERISK-22748 Reported by: Alejandro Mejia
+
+2015-02-25 20:43 +0000 [r432236]  David M. Lee <d...@digium.com>
+
+       * res/res_http_websocket.c: Increase WebSocket frame size and
+         improve large read handling Some WebSocket applications, like
+         [chan_respoke][], require a larger frame size than the default
+         8k; this patch bumps the default to 16k. This patch also fixes
+         some problems exacerbated by large frames. The sanity counter was
+         decremented on every fread attempt in ws_safe_read(), regardless
+         of whether data was read from the socket or not. For large
+         frames, this could result in loss of sanity prior to reading the
+         entire frame. (16k frame / 1448 bytes per segment = 12 segments).
+         This patch changes the sanity counter so that it only decrements
+         when fread() doesn't read any bytes. This more closely matches
+         the original intention of ws_safe_read(), given that the error
+         message is "Websocket seems unresponsive". This patch also
+         properly logs EOF conditions, so disconnects are no longer
+         confused with unresponsive connections. [chan_respoke]:
+         https://github.com/respoke/chan_respoke Review:
+         https://reviewboard.asterisk.org/r/4431/
+
+2015-02-24 22:14 +0000 [r432198]  Matthew Jordan <mjor...@digium.com>
+
+       * channels/chan_sip.c: channels/chan_sip: Fix crash when
+         transmitting packet after thread shutdown When the monitor thread
+         is stopped, its pthread ID is set to a specific value
+         (AST_PTHREADT_STOP) so that later portions of the code can
+         determine whether or not it is safe to manipulate the thread.
+         Unfortunately, __sip_reliable_xmit failed to check for that
+         value, checking instead only for AST_PTHREAD_STOP. Passing the
+         invalid yet very specific value to pthread_kill causes a crash.
+         This patch adds a check for AST_PTHREADT_STOP in
+         __sip_reliable_xmit such that it doesn't attempt to poke the
+         thread if the thread has already been stopped. ASTERISK-24800
+         #close Reported by: JoshE
+
+2015-02-24 18:22 +0000 [r432174]  Kevin Harwell <kharw...@digium.com>
+
+       * bridges/bridge_softmix.c: bridge_softmix: G.729 codec license
+         held When more than one call using the same codec type enters
+         into a softmix bridge and no audio is present for a channel the
+         bridge optimizes the out frame by using the same one for all
+         channels with the same codec type. Unfortunately, when that
+         number (channels with same codec type) dropped to <= 1 the codec
+         was not dereferenced. At least not until all parties left the
+         bridge. Thus in the case of G.729 the license was not released.
+         This patch ensures that the codec is dereferenced immediately
+         when the optimization no longer applies. ASTERISK-24797 #close
+         Reported by: Luke Hulsey Review:
+         https://reviewboard.asterisk.org/r/4429/
+
+2015-02-21 17:34 +0000 [r432098]  Matthew Jordan <mjor...@digium.com>
+
+       * apps/app_voicemail.c: apps/app_voicemail: Demote an ERROR message
+         to a WARNING message When using IMAP voicemail with FreePBX, you
+         will often get ERROR messages complaining about not being able to
+         find a mailbox. This is due to how FreePBX handles voicemail
+         mailboxes. Unfortunately, app_voicemail has to consider this a
+         configuration error, as in any other system it would be
+         indicative of someone misconfiguring their system. Regardless, a
+         misconfiguration is a WARNING, and not an ERROR. This patch
+         demotes the message so that system administrators can hopefully
+         reduce some of the noise in their log files. Note that in the
+         original patch this was made into a NOTICE, but that's a too
+         forgiving. ASTERISK-24790 #close Reported by: Graham Barnett
+         patches: app_voicemail.c.patch_noise uploaded by Graham Barnett
+         (License 6685)
+
+2015-02-21 14:04 +0000 [r432078]  Joshua Colp <jc...@digium.com>
+
+       * main/http.c: http: Add missing html tag to 'httpstatus'
+         functionality. ASTERISK-24724 #close Reported by: Ashley Sanders
+
+2015-02-21 02:55 +0000 [r432054-432058]  Corey Farrell <g...@cfware.com>
+
+       * main/loader.c: Allow shutdown to unload modules that register
+         bucket scheme's or codec's. * Change __ast_module_shutdown_ref to
+         be NULL safe (11+). * Allow modules that call
+         ast_bucket_scheme_register or ast_codec_register to be unloaded
+         during graceful shutdown only (13+ only). ASTERISK-24796 #close
+         Reported by: Corey Farrell Review:
+         https://reviewboard.asterisk.org/r/4428/
+
+       * include/asterisk/lock.h: asterisk/lock.h: Fix syntax errors for
+         non-gcc OSX with 64-bit integers. Add a couple of missing closing
+         brackets / parenthesis. ASTERISK-24814 #close Reported by: Corey
+         Farrell Review: https://reviewboard.asterisk.org/r/4436/
+
+2015-02-20 17:43 +0000 [r432032]  Richard Mudgett <rmudg...@digium.com>
+
+       * channels/sig_analog.c: chan_dahdi/sig_analog: Put log message
+         strings on one line. With the log messages on one line, you can
+         search for the log message seen in the log and expect to find it.
+
+2015-02-20 15:45 +0000 [r432012]  Matthew Jordan <mjor...@digium.com>
+
+       * apps/app_voicemail.c: apps/app_voicemail: Fix IMAP header
+         compatibility issue with Microsoft Exchange When interfacing with
+         Microsoft Exchange, custom headers will be returned as all lower
+         case. Currently, the IMAP header code will fail to parse the
+         returned custom headers, as it will be performing a case
+         sensitive comparison. This can cause playback of messages to
+         fail, as needed information - such as origtime - will not be
+         present. This patch updates app_voicemail's header parsing code
+         to perform a case insensitive lookup for the requested custom
+         headers. Since the headers are specific to Asterisk, e.g.,
+         'x-asterisk-vm-orig-time', and headers should be unique in an
+         IMAP message, this should cause no issues with other systems.
+         ASTERISK-24787 #close Reported by: Graham Barnett patches:
+         app_voicemail.c.patch_MSExchange uploaded by Graham Barnett
+         (License 6685)
+
+2015-02-19 21:23 +0000 [r431992]  Richard Mudgett <rmudg...@digium.com>
+
+       * channels/chan_dahdi.c, channels/sig_analog.c: chan_dahdi: Remove
+         some dead code.
+
+2015-02-19 15:21 +0000 [r431936]  Matthew Jordan <mjor...@digium.com>
+
+       * main/tcptls.c: tcptls: Handle new OpenSSL compile time option to
+         disable SSLv3 Some distributions are going to disable SSLv3 at
+         compile time. This option can be checked using the directive
+         OPENSSL_NO_SSL3_METHOD. This patch updates the TCP/TLS handling
+         in Asterisk to look for that directive before attempting to use
+         the SSLv3 specific methods. ASTERISK-24799 #close Reported by:
+         Alexander Traud patches: no-ssl3-method.patch uploaded by
+         Alexander Traud (License 6520)
+
+2015-02-19 01:59 +0000 [r431916]  Corey Farrell <g...@cfware.com>
+
+       * channels/chan_iax2.c, main/sched.c, include/asterisk/sched.h:
+         Create work around for scheduler leaks during shutdown. * Added
+         ast_sched_clean_by_callback for cleanup of scheduled events that
+         have not yet fired. * Run all pending peercnt_remove_cb and
+         replace_callno events in chan_iax2. Cleanup of replace_callno
+         events is only run 11, since it no longer releases any references
+         or allocations in 13+. ASTERISK-24451 #close Reported by: Corey
+         Farrell Review: https://reviewboard.asterisk.org/r/4425/
+
+2015-02-15 00:31 +0000 [r431788]  Matthew Jordan <mjor...@digium.com>
+
+       * apps/app_mixmonitor.c: apps/app_mixmonitor: Move Test Event for
+         MIXMONITOR_END to after it finishes The Test Event for
+         MIXMONITOR_END - which signals that a MixMonitor has completed -
+         technically fired before the filestream was closed. If a test
+         used this to trigger a condition to verify that the file was
+         written, it could result in a race condition where the file size
+         would not be what the test expected. Luckily, no tests were using
+         this (although they should have been). Since the test event
+         needed to be moved after the point where the MixMonitor autochan
+         has been destroyed, the test event no longer emits the channel
+         name. Luckily, nothing needs it.
+
+2015-02-11 17:11 +0000 [r431673]  Matthew Jordan <mjor...@digium.com>
+
+       * channels/chan_sip.c: channels/chan_sip: Fix RealTime error during
+         SIP unregistration with MariaDB When a SIP device that has its
+         registration stored in RealTime unregisters, the entry for that
+         device is updated with blank values, i.e., "", indicating that it
+         is no longer registered. Unfortunately, one of those values that
+         is 'blanked' is the device's port. If the column type for the
+         port is not a string datatype (the recommended type is integer),
+         an ODBC or database error will be thrown. MariaDB does not coerce
+         empty strings to a valid integer value. This patch updates the
+         query run from chan_sip such that it replaces the port value with
+         a value of '0', as opposed to a blank value. This is the value
+         that other database backends coerce the empty string ("") to
+         already, and the handling of reading a RealTime registration
+         value from a backend already anticipates receiving a port of '0'
+         from the backends. ASTERISK-24772 #close Reported by: Richard
+         Miller patches: chan_sip.diff uploaded by Richard Miller (License
+         5685)
+
+2015-02-11 16:46 +0000 [r431669]  Kevin Harwell <kharw...@digium.com>
+
+       * res/res_http_websocket.c: res_http_websocket: websocket write
+         timeout fails to fully disconnect When writing to a websocket if
+         a timeout occurred the underlying socket did not get
+         closed/disconnected. This patch makes sure the websocket gets
+         disconnected on a write timeout. ASTERISK-24701 #close Reported
+         by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4412/
+
+2015-02-11 15:38 +0000 [r431662]  Corey Farrell <g...@cfware.com>
+
+       * bridges/bridge_builtin_features.c, include/asterisk/module.h,
+         main/loader.c: Enable REF_DEBUG for ast_module_ref /
+         ast_module_unref. Add ast_module_shutdown_ref for use by modules
+         that can only be unloaded during graceful shutdown. When
+         REF_DEBUG is enabled: * Add an empty ao2 object to struct
+         ast_module. * Allocate ao2 object when the module is loaded. *
+         Perform an ao2_ref in each place where mod->usecount is
+         manipulated. * ao2_cleanup on module unload. ASTERISK-24479
+         #close Reported by: Corey Farrell Review:
+         https://reviewboard.asterisk.org/r/4141/
+
+2015-02-09 02:44 +0000 [r431617-431620]  Matthew Jordan <mjor...@digium.com>
+
+       * channels/sip/include/sip.h, channels/chan_sip.c:
+         channels/chan_sip: Ensure that a BYE is sent during INVITE
+         w/Replaces transfer Consider a scenario where Alice and Bob have
+         an established dialog with each other external to Asterisk. Bob
+         decides to perform an attended transfer of Alice to Asterisk. In
+         this case, Alice will send an INVITE with Replaces to Asterisk,
+         where the Replaces specifies Bob's dialog with Asterisk. In this
+         particular scenario, Asterisk will complete the transfer, but -
+         since Bob's channel has had Alice masqueraded into it and is now
+         a Zombie - a BYE request will not be sent. This patch fixes that
+         issue by adding a new flag to chan_sip that tracks whether or not
+         we have an INVITE with Replaces. If we do, the flag is used on
+         the sip_pvt to ensure that a BYE request is sent, even if the
+         channel has been masqueraded away. Review:
+         https://reviewboard.asterisk.org/r/4362/ ASTERISK-22436 #close
+         Reported by: Eelco Brolman Tested by: Jeremiah Gowdy, Kristian
+         Høgh patches: asterisk-11-hangup-replaced-3.diff uploaded by
+         Jeremiah Gowdy (License 6358)
+
+       * res/res_odbc.c: res/res_odbc: Remove unneeded queries when
+         determining if a table exists This patch modifies the
+         ast_odbc_find_table function such that it only performs a lookup
+         of the requested table if the table is not already known. Prior
+         to this patch, a queries would be executed against the database
+         even if the table was already known and cached. Review:
+         https://reviewboard.asterisk.org/r/4405/ ASTERISK-24742 #close
+         Reported by: ibercom patches: patch.diff uploaded by ibercom
+         (License 6599)
+
+2015-02-06 21:26 +0000 [r431582]  Scott Griepentrog <sgriepent...@digium.com>
+
+       * main/config.c: config hooks: correct ref leaks This small patch
+         fixes a ref leak when adding a config hook and cleans up the
+         container on shutdown. Review:
+         https://reviewboard.asterisk.org/r/4407
+
+2015-02-06  Asterisk Development Team <asteriskt...@digium.com>
+
+       * Asterisk 11.16.0 Released.
+
+2015-01-30  Asterisk Development Team <asteriskt...@digium.com>
+
+       * Asterisk 11.16.0-rc1 Released.
+
+2015-01-30 16:55 +0000 [r431423-431472]  Mark Michelson <mmichel...@digium.com>
+
+       * main/pbx.c: Backport memory leak fix in pbx.c from branch 13
+         revision 431468
+
+       * channels/chan_sip.c: Use SIPS URIs in Contact headers when
+         appropriate. RFC 3261 sections 8.1.1.8 and 12.1.1 dictate
+         specific scenarios when we are required to use SIPS URIs in
+         Contact headers. Asterisk's non-compliance with this could
+         actually cause calls to get dropped when communicating with
+         clients that are strict about checking the Contact header. Both
+         of the SIP stacks in Asterisk suffered from this issue. This
+         changeset corrects the behavior in chan_sip. ASTERISK-24646
+         #close Reported by Stephan Eisvogel Review:
+         https://reviewboard.asterisk.org/r/4346
+
+2015-01-29 12:08 +0000 [r431384]  Joshua Colp <jc...@digium.com>
+
+       * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix DTLS when used with
+         OpenSSL 1.0.1k A recent security fix for OpenSSL broke DTLS
+         negotiation for many applications. This was caused by read ahead
+         not being enabled when it should be. While a commit has gone into
+         OpenSSL to force read ahead on for DTLS it may take some time for
+         a release to be made and the change to be present in
+         distributions (if at all). As enabling read ahead is a simple one
+         line change this commit does that and fixes the issue.
+         ASTERISK-24711 #close Reported by: Jared Biel
+
+2015-01-28 17:12 +0000 [r431297-431298]  Mark Michelson <mmichel...@digium.com>
+
+       * funcs/func_curl.c: Fix compilation error from previous patch.
+
+       * funcs/func_curl.c: Mitigate possible HTTP injection attacks using
+         CURL() function in Asterisk. CVE-2014-8150 disclosed a
+         vulnerability in libcURL where HTTP request injection can be
+         performed given properly-crafted URLs. Since Asterisk makes use
+         of libcURL, and it is possible that users of Asterisk may get
+         cURL URLs from user input or remote sources, we have made a patch
+         to Asterisk to prevent such HTTP injection attacks from
+         originating from Asterisk. ASTERISK-24676 #close Reported by Matt
+         Jordan Review: https://reviewboard.asterisk.org/r/4364
+         AST-2015-002
+

[... 32749 lines stripped ...]

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

svn-commits mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/svn-commits

Reply via email to