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.
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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 + [... 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