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:
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+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, â\nâ to represent a newline character), you must double
+ any â\â that you use in LIKE strings. For example, to search for
+ â\nâ, specify it as â\\nâ. To search for â\â, specify it
as
+ â\\\\â; 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
+
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