[asterisk-commits] bebuild: tag 11.3.0-rc1 r380534 - /tags/11.3.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jan 30 12:13:41 CST 2013
Author: bebuild
Date: Wed Jan 30 12:13:37 2013
New Revision: 380534
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=380534
Log:
Importing files for 11.3.0-rc1 release.
Added:
tags/11.3.0-rc1/.lastclean (with props)
tags/11.3.0-rc1/.version (with props)
tags/11.3.0-rc1/ChangeLog (with props)
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+2013-01-30 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.3.0-rc1 Released.
+
+2013-01-30 17:46 +0000 [r380452-380521] Matthew Jordan <mjordan at digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Support building Asterisk for Raspberry Pi/Raspbian with
+ hard-float support Building Asterisk on Raspbian with hard-float
+ support fails as it uses the string 'linux-gnueabihf' for host
+ os, as opposed to 'linux-gnueabi'. This patch modifies the
+ configure script for Asterisk such that it will match on any
+ string beginning with 'linux-gnueabi', as opposed to requiring an
+ explicit match. (closes issue ASTERISK-21006) Reported by:
+ Christian Hesse Tested by: Christian Hesse patches:
+ linux-gnueabihf.patch uploaded by Christian Hesse (license 6459)
+ linux-gnueabihf-autoconf.patch uploaded by Christian Hesse
+ (license 6459) ........ Merged revisions 380520 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_sip.c: Unregister SIP provider API if module load
+ is declined A user in #asterisk ran into a problem where a
+ configuration error prevented the chan_sip module from being
+ loaded. Upon fixing their configuratione error, they could no
+ longer load the chan_sip module. This was because the
+ configuration checking happened after the SIP provider was
+ registered with the Asterisk core, and subsequent attempts to
+ load the SIP module failed as the provider was already
+ registered. Since we want to detect any failure in registering
+ chan_sip as early as possible (as that could be emblematic of a
+ deeper mismatch between module and Asterisk core), this patch
+ does not change the registration location, but does ensure that
+ if a module load is declined, we unregister the module as the SIP
+ api provider.
+
+ * /, channels/chan_sip.c: Perform case insensitive comparisons for
+ T.38 attributes RFC5347 section 2.5.2 states the following: ...
+ The attribute "T38MaxBitRate" was once incorrectly registered
+ with IANA as "T38maxBitRate" (lower-case "m"). In accordance with
+ T.38 examples and common implementation practice, the form
+ "T38MaxBitRate" SHOULD be generated by implementations conforming
+ to this package. In general, it is RECOMMENDED that
+ implementations of this package accept lowercase, uppercase, and
+ mixed upper/lowercase encodings of all the T.38 attributes. ...
+ Asterisk currently does not perform case insensitive matching on
+ the T.38 attributes. This causes the T38MaxBitRate attribute to
+ be negotiated at 2400 baud instead of 14400 (or whatever value
+ you actually wanted). This patch makes it so that when we compare
+ T.38 attributes, we do so in a case insensitive fashion. Note
+ that while the issue reporter did not directly write the patch,
+ they contributed to it (and would have provided one themselves if
+ the license had gone through a tad faster), and hence get
+ attribution for it. Review:
+ https://reviewboard.asterisk.org/r/2298/ (closes issue
+ ASTERISK-20897) Reported by: Eric Hill Tested by: Eric Hill
+ patches: -- uploaded by Eric Hill ........ Merged revisions
+ 380458 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/res_calendar_icalendar.c, /: Fix memory leak in
+ res_calendar_icalendar The ICalendar module had a systemic memory
+ leak on each fetch of data from the ICalendar source. The
+ previous fetched data was not being properly disposed. This patch
+ makes it so that before each fetch of data, we dispose of the
+ previously fetched data. (closes issue ASTERISK-21012) Reported
+ by: Joel Vandal Tested by: Joel Vandal ........ Merged revisions
+ 380451 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-29 17:54 +0000 [r380384] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_agent.c: chan_agent: Prevent multiple channels
+ from logging in as the same agent. Multiple channels logging in
+ as the same agent can result in dead channels waiting for a
+ condition signal that will never come because another channel
+ thread stole it. A symptom is chan_sip repeatedly generating
+ warning messages about rescheduling autodestruction of dialogs
+ with an agent channel owner. * Made only login_exec() (the app
+ AgentLogin) clear the agent_pvt->chan pointer to prevent multiple
+ channels from logging in as the same agent. agent_read(),
+ agent_call(), and agent_set_base_channel() no longer disconnect
+ the agent channel from the agent_pvt. This also eliminates the
+ need to keep checking for agent_pvt->chan being NULL. * Made
+ agent_hangup() not wake up the AgentLogin agent thread until it
+ is done. * Made agent_request() not able to get the agent until
+ he has logged in and any wrapup time has expired. * Made
+ agent_request() use ast_hangup() instead of agent_hangup() to
+ correctly dispose of a channel. * Removed
+ agent_set_base_channel(). Nobody calls it and it is a bad thing
+ in general. * Made only agent_devicestate() determine the current
+ device state of an agent. Note: Agent group device states have
+ never been supported. Review:
+ https://reviewboard.asterisk.org/r/2260/ ........ Merged
+ revisions 380364 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-29 17:14 +0000 [r380350] David M. Lee <dlee at digium.com>
+
+ * channels/sip/sdp_crypto.c, /: Corrected crypto tag in SDP ANSWER
+ for SRTP. (again) The original fix (r380043) for getting Asterisk
+ to respond with the correct tag overlooked some corner cases, and
+ the fact that the same code is in 1.8. This patch moves the
+ building of the crypto line out of sdp_crypto_process(). Instead,
+ it merely copies the accepted tag. The call to sdp_crypto_offer()
+ will build the crypto line in all cases now, using a tag of "1"
+ in the case of sending offers. (closes issue ASTERISK-20849)
+ Reported by: José Luis Millán Review:
+ https://reviewboard.asterisk.org/r/2295/ ........ Merged
+ revisions 380347 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-29 17:05 +0000 [r380348] Jonathan Rose <jrose at digium.com>
+
+ * main/features.c: call_parking: Make sure fallbacks are used when
+ lacking a flat channel exten A regression was introduced which
+ removed automatic fallback behavior from the PBX. This behavior
+ was used by call parking (or at least documented as how the
+ feature works) in order to select an extension when the flat
+ channel extension wasn't available from the comebackcontext.
+ Parking now handles the fallbacks internally in order to keep
+ behavior matching with how it is documented. (closes issue
+ ASTERISK-20716) Reported by: Chris Gentle Review:
+ https://reviewboard.asterisk.org/r/2296/
+
+2013-01-29 14:45 +0000 [r380298-380331] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: Ensure that a declined media stream is
+ terminated with a '\r\n' In r369028, chan_sip's processing of
+ media streams in an SDP was modified to better handle multiple
+ offered media streams. Part of that change modified how streams
+ were declined. Previously, declined media streams were not
+ handled in an RFC compliant manner; now, we set the port number
+ to 0 in the media stream definition and proceed on with the next
+ media stream. Unfortunately, the formatting of the declined media
+ stream forgot to append a '\r\n' to the end of the media stream.
+ This is normally added to the accepted media streams later on in
+ the processing of the SDP. Since the declined media stream uses a
+ different buffer than the accepted media streams (and is a
+ malloc'd buffer as opposed to a struct ast_str), it's easier to
+ just slap the '\r\n' on the declined media stream buffer rather
+ than attempt to append it later on. So, that's what we do. And
+ now some devices (and probably some providers) will be a bit
+ happier (but probably not terribly happy, since we just rejected
+ something they offered). Review:
+ https://reviewboard.asterisk.org/r/2297/ (closes issue
+ ASTERISK-20908) Reported by: Dennis DeDonatis Tested by: Dennis
+ DeDonatis
+
+ * autoconf/ast_check_pwlib.m4, /, configure: Update configure
+ script to be compatible with ptlib 2.10.9 With ptlib 2.10.9, the
+ configure script fails due to grep returning multiple matches for
+ the pattern it searches for. This patch updates the pattern
+ matching to return only the actual version for the symbol
+ searched for, PTLIB_VERSION. (closes issue ASTERISK-20980)
+ Reported by: Stefan Reuter patches: ASTERISK-20980-1.patch
+ uploaded by Stefan Reuter (license 5339) ........ Merged
+ revisions 380297 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-28 21:08 +0000 [r380255] Sean Bright <sean at malleable.com>
+
+ * /, channels/iax2.h, channels/chan_iax2.c: Correct the number of
+ available call numbers in IAX2. There is currently an edge case
+ where call number 32768 might be allocated for a call, even
+ though the IAX2 protocol requires call numbers be only 15 bits.
+ This resulted in some unpredictable behavior when call number
+ 32678 is chosen. This patch was mostly written by Richard Mudgett
+ via ReviewBoard. I'm just committing it. Review:
+ https://reviewboard.asterisk.org/r/2293/ ........ Merged
+ revisions 380254 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-28 01:57 +0000 [r380211] Russell Bryant <russell at russellbryant.com>
+
+ * /, main/file.c: Change cleanup ordering in filestream destructor.
+ This patch came about due to a problem observed where wav files
+ had an empty header. The header is supposed to be updated in
+ wav_close(). It turns out that this was broken when the
+ cache_record_files option from asterisk.conf was enabled. The
+ cleanup code was moving the file to its final destination
+ *before* running the close() method of the file destructor, so
+ the header didn't get updated. Another problem here is that the
+ move was being done before actually closing the FILE *. Finally,
+ the last bug fixed here is that I noticed that wav_close() checks
+ for stream->filename to be non-NULL. In the previous cleanup
+ order, it's checking a pointer to freed memory. This doesn't
+ actually cause anything to break, but it's treading on dangerous
+ waters. Now the free() of stream->filename is happening after the
+ format module's close() method gets called, so it's safer.
+ Review: https://reviewboard.asterisk.org/r/2286/ ........ Merged
+ revisions 380210 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-27 20:31 +0000 [r380193] Michael L. Young <elgueromexicano at gmail.com>
+
+ * apps/confbridge/conf_config_parser.c: Fix Some Configured
+ Conference Bridge Sounds Not Being Set The "sound_only_one" sound
+ was not being set even though it was configured. In looking into
+ this, I found that the "join" and "leave" prompts were not being
+ set either. (closes issue ASTERISK-20898) Reported by: Stephan
+ Tested by: Stephan Patches:
+ asterisk-20898-custom-sounds-ignored.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2289/
+
+2013-01-24 16:39 +0000 [r380043] David M. Lee <dlee at digium.com>
+
+ * channels/sip/sdp_crypto.c: Corrected crypto tag in SDP ANSWER for
+ SRTP. When Asterisk responds with an SDP ANSWER for SRTP, it had
+ the code to correctly fill in the crypto data, which was
+ overwritten by a call to sdp_crypto_offer. Corrected the
+ situation by changing sdp_crypto_offer to not replacing crypto
+ data if it already exists. (closes issue ASTERISK-20849) Reported
+ by: José Luis Millán Tested by: Iñaki Baz Castillo Patches:
+ fix_sdp_crypto_tags.diff uploaded by Pedro Kiefer (license 6407)
+
+2013-01-24 04:01 +0000 [r380028] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_confbridge.c: Correct documentation for ConfbridgeList
+ AMI action The documentation for ConfbridgeList states that the
+ Conference field is optional. That's not really the case: if you
+ fail to provide a Conference number, the command will kick back
+ an error. (closes issue AST-1090) Reported by: John Bigelow
+
+2013-01-23 00:23 +0000 [r379964] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/astobj2.c: Attempt to be more helpful when using a bad
+ ao2 object pointer. Put the external obj pointer in the message
+ instead of the internal version. ........ Merged revisions 379963
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-22 22:05 +0000 [r379892-379949] Jonathan Rose <jrose at digium.com>
+
+ * res/res_fax_spandsp.c: res_fax_spandsp: fix t38 transmission bug
+ caused by not returning success This patch fixes the problem, but
+ the issue includes a test which is still being considered for the
+ automated test suite. (issue ASTERISK-20919) Reported by: NITESH
+ BANSAL Patches: patch_ast_fax_spandsp.patch uploaded by NITESH
+ BANSAL (license 6418)
+
+ * /, apps/app_meetme.c, sounds/Makefile: app_meetme: Use new
+ prompts for administrator menu The old prompts for the
+ administrator menu were inadequate. They didn't mention that the
+ menu had additional options through the 8 key and pressing the 8
+ key wouldn't reveal what those options were. This patch fixes all
+ of that while also organizing code pertaining to each individual
+ menu type which was previously all stored in one gigantic
+ function along with many of the basic conference functions.
+ (closes issue AST-996) Reported by: John Bigelow Review:
+ http://reviewboard.digium.internal/r/360/ ........ Merged
+ revisions 379885 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-22 14:51 +0000 [r379826] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/app_meetme.c: Fix station ringback; trunk hangup issues
+ in SLA This patch fixes two bugs: * If an outbound call is made
+ from a SLA phone using SLAStation, then there is no ringtone
+ audible to the phone that originates the call. The indication of
+ the ringing was not being passed to the SLA station; this patch
+ fixes that by passing through the progress indications. * If an
+ SLA station hangs up before the called party answers, then the
+ channel to the called party continues to ring until a timeout
+ occurs. If the called party manages to answer, Asterisk attempts
+ to connect the called party to a non-existant MeetMe room. This
+ patch corrects the behavior by abandoning the call attempt if it
+ detects that the SLA station is no longer in use while attempting
+ to call the called party. Review:
+ https://reviewboard.asterisk.org/r/2275/ (closes issue
+ ASTERISK-20462) Reported by: dkerr patches:
+ asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
+ 5558) asterisk-11-bugid20462.patch uploaded by dkerr (license
+ 5558) (closes issue ASTERISK-20440) Reported by: dkerr patches:
+ asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
+ asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
+ 5558) ........ Merged revisions 379825 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-22 00:35 +0000 [r379808] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_bridge.c, apps/app_confbridge.c: confbridge: Minor
+ fixes playing user counts to the conference. * Generate a warning
+ message if sound files do not exist when trying to play the user
+ count to the conference. Use the new helper routine
+ sound_file_exists() for consistency. * Put the new user into
+ autoservice when playing user counts to the conference. * Check
+ the return value of ast_bridge_impart().
+
+2013-01-21 20:40 +0000 [r379790] Matthew Jordan <mjordan at digium.com>
+
+ * contrib/scripts/safe_asterisk, main/asterisk.c,
+ contrib/init.d/rc.suse.asterisk,
+ contrib/init.d/rc.mandriva.asterisk,
+ contrib/init.d/rc.debian.asterisk, /,
+ contrib/init.d/rc.redhat.asterisk, UPGRADE.txt,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk,
+ contrib/init.d/rc.archlinux.asterisk: Update init.d scripts to
+ handle stderr; readd splash screen for remote consoles When
+ r376428 was commited to re-order start up sequences to be more
+ tolerant of forking with thread primitives, a few items were
+ changed that caused changes in behavior on some distros. This
+ includes: * Not displaying the splash screen on a remote console.
+ * Displaying an error message on stderr when a remote console
+ cannot connect to a running instance of Asterisk. In the first
+ case, the splash screen was re-added (thanks to Michael L.
+ Young). In the second case, the various init.d scripts were
+ modified to pipe stderr to /dev/null, as the error message is
+ useful - if you execute a remote console or a remote console
+ command execution and it fail, it should tell you. Note that the
+ error message was always present, it just failed to be printed
+ prior to r376428. Much thanks to the folks who quickly reported
+ this problem, provided solutions, and promptly tested the various
+ init.d scripts on a variety of distros. (closes issue
+ ASTERISK-20945) Reported by: Warren Selby Tested by: Michael L.
+ Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan patches:
+ asterisk-20945-remote-intro-msg.diff uploaded by elguero (license
+ 5026) ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan
+ (license 6283) ........ Merged revisions 379760 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379777 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-01-21 18:33 +0000 [r379719] Kinsey Moore <kmoore at digium.com>
+
+ * /, codecs/codec_ilbc.c: Prevent segfault for interpolated iLBC
+ frames When iLBC is being used with a jitter buffer and the jb
+ has to interpolate frames, it generates frames with a null
+ pointer and a non-zero datalen. This is now handled properly.
+ (closes issue ASTERISK-20914) Reported By: John McEleney Patches:
+ ASTERISK-20914-1.8.diff uploaded by Matt Jordan (license 6283)
+ ........ Merged revisions 379718 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-21 06:27 +0000 [r379677] Damien Wedhorn <voip at facts.com.au>
+
+ * channels/chan_skinny.c: Fix device call logging issues in skinny
+ Skinny device call logging (ie missed, place and received calls)
+ has issues because the incorrect sequence of callstates is/can be
+ sent to the device. This patch removes some extra callstate
+ updates driven by forces external to skinny and ensures the
+ needed intermediary callstate messages are sent. (closes issue
+ ASTERISK-20964) Reported by: wedhorn Tested by: snuffy, myself
+ Patches: ast11-skinny-calllog01.diff uploaded by wedhorn (license
+ 5019)
+
+2013-01-21 04:39 +0000 [r379643] Andrew Latham <lathama at gmail.com>
+
+ * contrib/scripts/install_prereq: Add LDAP libraries to install
+ script Add LDAP dev package to Debian/Ubuntu install list.
+ Existed in Redhat already. (issue ASTERISK-20886)
+
+2013-01-21 04:07 +0000 [r379609] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/app_minivm.c: Fix crash in app_minivm when mime encoding
+ string An incorrect string initializations was left in
+ ast_str_encode_mime from the patch that converted string
+ manipulations to use ast_str strings (r191140). The string
+ initialization causes a crash when ast_str_set is called on the
+ string later on in the function. (closes issue ASTERISK-18697)
+ Reported by: Chris Boot patches:
+ minivm-null-pointer-dereference-fix.patch uploaded by bootc
+ (license 6309) (issue ASTERISK-20854) Reported by: Chris Warr
+ Tested by: Chris Warr ........ Merged revisions 379608 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-20 02:53 +0000 [r379582] Damien Wedhorn <voip at facts.com.au>
+
+ * channels/chan_skinny.c: Fix issues with skinny sessions Fixes a
+ couple of issues with the way skinny handles sessions by ensuring
+ sessions aren't used after being freed. Some other minor changes.
+ Review: https://reviewboard.asterisk.org/r/2272/
+
+2013-01-19 20:49 +0000 [r379548] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac: Add
+ builtin roundf() for systems lacking it. (closes issue
+ ASTERISK-16854) Review: https://reviewboard.asterisk.org/r/2276
+ Reported-by: Ovidiu Sas ........ Merged revisions 379547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-19 00:17 +0000 [r379513] Matthew Jordan <mjordan at digium.com>
+
+ * main/asterisk.c, /: Fix astcanary startup problem due to wrong
+ pid value from before daemon call When Asterisk forks itself into
+ the background via a call to daemon, it must re-set the pid value
+ of the new process. Otherwise, astcanary gets the pid value of
+ the process before the fork, which prevents it from running.
+ Asterisk eventually starts lowering its priority, as it can no
+ longer communicate with the proverbial canary in the coal mine.
+ This patch ensures that the correct process identifier is used by
+ astcanary. Note that this is getting committed to 10 as a
+ regression fix. (closes issue ASTERISK-20947) Reported by: Jakob
+ Hirsch Tested by: mjordan patches:
+ asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch
+ (license 6113) ........ Merged revisions 379509 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 379510 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-01-18 21:46 +0000 [r379478] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_confbridge.c: Fix regression in Confbridge user count
+ When the restructuring work got committed to Confbridge in
+ r375470 to fix many open issues, it caused a regression in the
+ reported count of users when conference information was requested
+ via CLI or manager. This corrects the user count and user
+ information displayed when listing conference information from
+ the CLI and manager. (closes issue ASTERISK-20938) Reported By:
+ Timo Teras Patches: confbridge-list.patch uploaded by Timo Teras
+ (license 5409)
+
+2013-01-18 21:10 +0000 [r379475] David M. Lee <dlee at digium.com>
+
+ * Makefile, configure, include/asterisk/autoconfig.h.in,
+ main/Makefile, configure.ac, UPGRADE.txt, makeopts.in: Specify
+ the -rpath linker flag when prefix != /usr. This allows Asterisk
+ to start without having to specify the LD_LIBRARY_PATH. This can
+ be disabled by passing --disable-rpath to configure. (closes
+ issue ASTERISK-20407) Reported by: David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2132/
+
+2013-01-18 18:13 +0000 [r379460] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_voicemail.c: app_voicemail: Improve msg_id handling
+ app_voicemail will no longer issue error messages when it
+ retrieves an msg_id with a NULL value from realtime and will
+ instead simply populate the msg_id field with a newly generated
+ msg_id. In addition, this patch changes the way msg_ids are
+ generated to eliminate certain causes of duplicate IDs appearing
+ within a single system. In addition, when messages are copied,
+ they will now receive a new msg_id. (closes issue ASTERISK-20717)
+ Reported by: Alec Davis Review:
+ https://reviewboard.asterisk.org/r/2220/
+
+2013-01-18 05:26 +0000 [r379393] David M. Lee <dlee at digium.com>
+
+ * channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c,
+ channels/sip/reqresp_parser.c: Fix Record-Route parsing for large
+ headers. Record-Route parsing copied the header into a char[256]
+ array, which can be a problem if the header is longer than that.
+ This patch parses the header in place, without the copy, avoiding
+ the issue. In addition to the original patch, I added a unit test
+ for the new get_in_brackets_const function. (closes issue
+ ASTERISK-20837) Reported by: Corey Farrell Patches:
+ chan_sip-build_route-optimized-rev1.patch uploaded by Corey
+ Farrell (license 5909) (with minor changes by dlee) ........
+ Merged revisions 379392 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-17 02:30 +0000 [r379343] Matthew Jordan <mjordan at digium.com>
+
+ * /, addons/chan_mobile.c: Fix issue where chan_mobile fails to
+ bind to first available port Per the bluez API, in order to bind
+ to the first available port, the rc_channel field of the socket
+ addressing structure used to bind the socket should be set to 0.
+ Previously, Asterisk had set the rc_channel field set to 1,
+ causing it to connect to whatever happens to be on port 1. We
+ could probably not explicitly set rc_channel to 0 since we memset
+ the struct earlier, but explicitly setting it will hopefully
+ prevent someone from coming in and setting it to some explicit
+ port in the future. (closes issue ASTERISK-16357) Reported by:
+ challado Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin,
+ eliafino, David van Geyn patches: ASTERISK-16357.diff uploaded by
+ Nikolay Ilduganov (license 6253) ........ Merged revisions 379342
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-16 22:49 +0000 [r379311] Mark Michelson <mmichelson at digium.com>
+
+ * main/manager.c, /: Further fix misinformation in the description
+ of manager MailboxStatus command. The description still claimed
+ that it returned the number of messages rather than whether there
+ were messages waiting. ........ Merged revisions 379310 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-16 21:13 +0000 [r379277] Jason Parker <jparker at digium.com>
+
+ * contrib/scripts/install_prereq, /: Reduce number of packages
+ install_prereq installs on Debian systems. 'search' will look for
+ any package containing the name provided, so we need to force a
+ more exact search. ........ Merged revisions 379276 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-16 18:08 +0000 [r379230-379232] Richard Mudgett <rmudgett at digium.com>
+
+ * main/logger.c: Reduce call-id logging resource usage. Since there
+ is no need for the call-id logging ao2 object to have a lock,
+ don't create it with one.
+
+ * channels/chan_misdn.c, /: chan_misdn: Fix compile error. (issue
+ ASTERISK-15456) ........ Merged revisions 379226 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-16 17:45 +0000 [r379146-379228] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_xmpp.c, res/res_jabber.c, doc/appdocsxml.dtd: Let
+ documentation reference links specify which module they're
+ linking to Again, since res_jabber/res_xmpp have duplicate APIs,
+ their documentation ref links have to specify which reference
+ they're referring to. The various documentation parsers can
+ interpret the module attribute however they want in order to
+ construct the appropriate links.
+
+ * doc/appdocsxml.dtd: Update the dtd to actually *support* the
+ module attribute in all elements Mea culpa.
+
+ * res/res_xmpp.c, res/res_jabber.c: Add module tags to
+ documentation for res_jabber/res_xmpp Since res_jabber/res_xmpp
+ provide the same APIs (app/func/manager/etc.), the XML
+ documentation for each needs to call out which module is
+ providing the documentation. The module attribute has been added
+ to the various XML fragments for this purpose.
+
+ * /, addons/chan_mobile.c: Fix parsing SMSSRC for SMS messages The
+ parser for SMS messages would incorrectly parse out the from
+ number. The parsing would incorrectly start scanning for the from
+ number at the same index as the first double quote ("); this
+ would inadvertently cause it to treat the first double quote as
+ the terminating double quote for the from number as well. The
+ SMSSRC should now populate correctly. (closes issue
+ ASTERISK-16822) Reported by: menschentier Tested by: Jonas Falck
+ patches: fixSMSSRC.patch uploaded by jonax (license 6320) (closes
+ issue ASTERISK-19153) Reported by: Panos Gkikakis patches:
+ sms-sender-fix.diff uploaded by roeften (license 5884) ........
+ Merged revisions 379178 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_misdn.c, /: Set the INVALID_EXTEN channel variable
+ when chan_misdn forces the 'i' extension The chan_misdn channel
+ driver will send a channel with an invalid destination to the 'i'
+ extension itself if said extension can be reached. It forgot,
+ however, to set the INVALID_EXTEN channel variable when it
+ bounces the channel to this extension. Dialplan writers
+ everywhere moaned at yet another inconsistency. This is yet
+ another example of why duplicating logic in multiple places
+ results in bugs that stick around in Jira for just under three
+ years. Yes: ASTERISK-15456 was created on January 18th, 2010.
+ Patch committed on January 15th, 2013. Ouch. (closes issue
+ ASTERISK-15456) Reported by: Thomas Omerzu patches:
+ chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license
+ 5927) ........ Merged revisions 379145 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-14 15:27 +0000 [r379020] David M. Lee <dlee at digium.com>
+
+ * /, channels/chan_sip.c: Fix XML encoding of 'identity display' in
+ NOTIFY messages, continued. When r378933 was merged into 1.8, it
+ should have also escaped remote_display, since it will have the
+ same XML encoding problem when the caller/callee roles are
+ reversed. (closes issue ABE-2902) Reported by: Guenther Kelleter
+ ........ Merged revisions 379001 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-13 21:44 +0000 [r378984] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c, /: Reset RTP timestamp; sequence number
+ on SSRC change In r370252 for ASTERISK-18404, Asterisk's handling
+ of RTP was modified to better account for out of order RTP
+ packets. This was accomplished by using the RTP timestamp and
+ sequence number to check for out of order packets. However, when
+ a SSRC change occurs, the timestamp and sequence number will no
+ longer have any relation to the previously received packets. The
+ variables tracking the timestamp and sequence number therefore
+ have to be reset. (closes issue ASTERISK-20906) Reported by:
+ Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco
+ Brolman (license #6442) ........ Merged revisions 378967 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-12 06:36 +0000 [r378934] David M. Lee <dlee at digium.com>
+
+ * include/asterisk/utils.h, /, channels/chan_sip.c,
+ tests/test_xml_escape.c (added), main/utils.c: Fix XML encoding
+ of 'identity display' in NOTIFY messages. XML encoding in
+ chan_sip is accomplished by naively building the XML directly
+ from strings. While this usually works, it fails to take into
+ account escaping the reserved characters in XML. This patch adds
+ an 'ast_xml_escape' function, which works similarly to
+ 'ast_uri_encode'. This is used to properly escape the
+ local_display attribute in XML formatted NOTIFY messages. Several
+ things to note: * The Right Thing(TM) to do would probably be to
+ replace the ast_build_string stuff with building an ast_xml_doc.
+ That's a much bigger change, and out of scope for the original
+ ticket, so I refrained myself. * It is with great sadness that I
+ wrote my own ast_xml_escape function. There's one in libxml2, but
+ it's knee-deep in libxml2-ness, and not easily used to one-off
+ escape a string. * I only escaped the string we know is causing
+ problems (local_display). At least some of the other strings are
+ URI-encoded, which should be XML safe. Rather than figuring out
+ what's safe and escaping what's not, it would be much cleaner to
+ simply build an ast_xml_doc for the messages and let the XML
+ library do the XML escaping. Like I said, that's out of scope.
+ (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by:
+ Guenther Kelleter Review:
+ http://reviewboard.digium.internal/r/365/ ........ Merged
+ revision 378919 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 378933 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-11 23:04 +0000 [r378917] Joshua Colp <jcolp at digium.com>
+
+ * res/res_xmpp.c: Retain XMPP filters across reconnections so
+ external modules continue to function as expected. Previously if
+ an XMPP client reconnected any filters added by an external
+ module were lost. This issue exhibited itself with chan_motif not
+ receiving and reacting to Jingle signaling. (closes issue
+ ASTERISK-20916) Reported by: kuj
+
+2013-01-09 20:29 +0000 [r378734-378780] David M. Lee <dlee at digium.com>
+
+ * main/rtp_engine.c, /: Fix end condition in
+ ast_rtp_lookup_mime_multiple2. The erroneous end condition would
+ never include the AST_RTP_CISCO_DTMF flag in the debug output.
+ (closes issue ASTERISK-20772) Reported by: Xavier Hienne ........
+ Merged revisions 378776 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * include/asterisk/strings.h: Move declaration of
+ ast_regex_string_to_regex_pattern futher down strings.h. The
+ prior location is before the declaration of struct ast_str, which
+ causes compiler warnings. (closes issue ASTERISK-20852) Reported
+ by: Pavel Troller Patches: strings.diff uploaded by Pavel Troller
+ (license 6302)
+
+ * /, include/asterisk/causes.h: Replace errant tabs with spaces in
+ causes.h. (closes issue ASTERISK-20826) Reported by: snuffy
+ Patches: notabs.dif uploaded by snuffy (license 5024) ........
+ Merged revisions 378733 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-09 00:03 +0000 [r378687-378690] Richard Mudgett <rmudgett at digium.com>
+
+ * /, apps/app_queue.c: app_queue: Fix incorrect assertion. (issue
+ ASTERISK-16115) ........ Merged revisions 378689 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, configs/queues.conf.sample, UPGRADE.txt, CHANGES,
+ apps/app_queue.c: app_queue: Fix multiple calls to a queue member
+ that is in only one queue. When ringinuse=no queue members can
+ receive more than one call if these calls happen at nearly the
+ same time. * Fix so a queue member does not receive more than one
+ call from a queue. NOTE: This fix does not prevent multiple calls
+ to a member if the member is in more than one queue. * Did some
+ refactoring to eliminate some code redundancy. (issue
+ ASTERISK-16115) Reported by: nik600 Patches:
+ jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Modified * Revert the -r341580 and -r341599
+ changes adding the queues.conf check_state_unknown option as it
+ was added in an attempt to fix this problem. The fix did not need
+ to be optional. The fix should not have tried to explicitly set
+ the device state. Setting the device state by something other
+ than the device introduces a race condition. I also could not see
+ how the change would be effective other than delaying the
+ app_queue code long enough for the device state to propagate to
+ app_queue. ........ Merged revisions 378663 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 378683 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-01-06 20:40 +0000 [r378622] Damien Wedhorn <voip at facts.com.au>
+
+ * channels/chan_skinny.c: Rewrite skinny dialing to remove threaded
+ simpleswitch This rewrite changes skinny dialing from the
+ threaded simpleswitch to a scheduled timeout approach. There were
+ some underlying issues with the threaded simple switch with
+ occasional corruption and possible segfaults. Review:
+ https://reviewboard.asterisk.org/r/2240/
+
+2013-01-04 23:04 +0000 [r378592] Jonathan Rose <jrose at digium.com>
+
+ * res/res_srtp.c, /: res_srtp: Prevent a crash from occurring due
+ to srtp_create failures in srtp_create Under some circumstances,
+ libsrtp's srtp_create function deallocates memory that it wasn't
+ initially responsible for allocating. Because we weren't
+ initially aware of this behavior, this memory was still used in
+ spite of being unallocated during the course of the
+ srtp_unprotect function. A while back I made a patch which would
+ set this value to NULL, but that exposed a possible condition
+ where we would then try to check a member of the struct which
+ would cause a segfault. In order to address these problems,
+ ast_srtp_unprotect will now set an error value when it ends
+ without a valid SRTP session which will result in the caller of
+ srtp_unprotect observing this error and hanging up the relevant
+ channel instead of trying to keep using the invalid session
+ address. (closes issue ASTERISK-20499) Reported by: Tootai
+ Review:
+ https://reviewboard.asterisk.org/r/2228/diff/#index_header
+ ........ Merged revisions 378591 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-04 22:18 +0000 [r378582] Kinsey Moore <kmoore at digium.com>
+
+ * res/pjproject/aconfigure, res/pjproject/aconfigure.ac,
+ res/pjproject/build/common.mak: Fix pjproject compilation in
+ certain circumstances On a fresh checkout of Asterisk 11, running
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