[asterisk-commits] bebuild: tag 10.2.0-rc1 r353314 - /tags/10.2.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jan 30 08:17:01 CST 2012
Author: bebuild
Date: Mon Jan 30 08:16:57 2012
New Revision: 353314
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=353314
Log:
Importing files for 10.2.0-rc1 release.
Added:
tags/10.2.0-rc1/.lastclean (with props)
tags/10.2.0-rc1/.version (with props)
tags/10.2.0-rc1/ChangeLog (with props)
Added: tags/10.2.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/10.2.0-rc1/.lastclean?view=auto&rev=353314
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--- tags/10.2.0-rc1/ChangeLog (added)
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+2012-01-30 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.2.0-rc1 Released.
+
+2012-01-30 12:48 +0000 [r353261] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, channels/chan_sip.c: Clarify log WARNING message when
+ port-zero SDP 'm' lines received. Previously, if an m-line in an
+ SDP offer or answer had a port number of zero, that line was
+ skipped, and resulted in an 'Unsupported SDP media type...'
+ warning message. This was misleading, as the media type was not
+ unsupported, but was ignored because the m-line indicated that
+ the media stream had been rejected (in an answer) or was not
+ going to be used (in an offer). ........ Merged revisions 353260
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-29 02:44 +0000 [r353176] Russell Bryant <russell at russellbryant.com>
+
+ * main/netsock.c, /: Find even more network interfaces. The
+ previous change made the code look for emN and pciN in addition
+ to what it did originally, which was search for ethN. However, it
+ needed to be looking for pciN#N, so that's what it does now. This
+ also moves the memset() to be before every ioctl(). ........
+ Merged revisions 353175 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-28 14:51 +0000 [r353127] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/rtp_engine.c, /: Add 'L16-256' MIME subtype alias for
+ slin16. Asterisk has supported the 'L16' MIME subtype for 16kHz
+ signed linear (PCM) audio for quite some time, but some endpoints
+ refer to it as 'L16-256'. This commit adds this as an alias for
+ the existing format. ........ Merged revisions 353126 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-28 04:27 +0000 [r353078] Russell Bryant <russell at russellbryant.com>
+
+ * main/netsock.c, /: Update ast_set_default_eid() to find more
+ network interfaces. As of Fedora 15, ethN is not the name of
+ ethernet interfaces. The names are emN or pciN. Update some code
+ that searched for interfaces named ethN to look for the new
+ names, as well. For more information about why this change was
+ made, see this page: http://domsch.com/blog/?p=455 ........
+ Merged revisions 353077 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-27 21:37 +0000 [r352992-353039] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c: Audit of ao2_iterator_init() usage for v10.
+ Missed one.
+
+ * tests/test_format_api.c: Audit of ao2_iterator_init() usage for
+ v10. Fix double format_cap iterator cleanup.
+
+2012-01-27 19:19 +0000 [r352965] Jonathan Rose <jrose at digium.com>
+
+ * /, res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor
+ with no valid channel not close AMI session. I also went ahead
+ and took a little time to make sure that the manager value
+ AMI_SUCCESS was used instead of just return 0 being thrown around
+ everywhere since that's how we handle this stuff these days.
+ (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches:
+ res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey
+ (license 5766) ........ Merged revisions 352959 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-27 18:36 +0000 [r352956] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_srtp.c, main/pbx.c, /, channels/chan_sip.c,
+ include/asterisk/indications.h, res/snmp/agent.c,
+ main/taskprocessor.c, apps/app_queue.c, channels/chan_iax2.c,
+ apps/app_chanspy.c, main/indications.c, res/res_odbc.c: Audit of
+ ao2_iterator_init() usage for v1.8. Fixes numerous reference
+ leaks and missing ao2_iterator_destroy() calls as a result.
+ Review: https://reviewboard.asterisk.org/r/1697/ ........ Merged
+ revisions 352955 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-27 00:08 +0000 [r352863] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
+ revisions 352862 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan
+ 2012) | 12 lines rfc4235 - Section 4.1: Versions MUST be
+ representable using a non-negative 32 bit integer. If a BLF
+ subscription exists for long enough, using %d may print negative
+ version numbers. Unlikely, as 2^32 at 1 update per second is ~137
+ years, or half that before the versions number started going
+ negative. Tested with Asterisk 1.8.8.2 with Grandstream phones.
+ alecdavis (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1694/ ........
+
+2012-01-26 20:22 +0000 [r352817] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: Fix outbound DTMF for inband mode (tell
+ asterisk core to generate DTMF sounds). (Closes issue
+ ASTERISK-19233) Reported by: Matt Behrens Patches:
+ chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
+ ........ Merged revisions 352807 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-26 19:07 +0000 [r352756] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c: Copy amaflags to sip_pvt from peer during
+ create_addr_from_peer For whatever reason, we don't have a single
+ function for copying data like this from SIP peers to the SIP
+ pvt. This patch adds the copying of amaflags to the sip_pvt, but
+ it would probably be worth discussing this function along with
+ the others that essentially just copy some amount of data from a
+ peer to a private. (Closes issue ASTERISK-19029) Reported by:
+ Matt Lehner ........ Merged revisions 352755 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-26 06:33 +0000 [r352705] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * /, channels/chan_sip.c: Merged revisions 352704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan
+ 2012) | 20 lines Cleanup dialog-info+xml Notify dialog Make
+ similar to other Notify messages. sample output: <?xml
+ version="1.0"?> <dialog-info
+ xmlns="urn:ietf:params:xml:ns:dialog-info" version="715"
+ state="full" entity="sip:8523 at 192.168.x.xx"> <dialog id="8523">
+ <state>terminated</state> </dialog> </dialog-info> Tested with
+ Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
+ Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1693/ ........
+
+2012-01-25 22:23 +0000 [r352651] Paul Belanger <pabelanger at digium.com>
+
+ * apps/app_voicemail.c, /: Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2) ........ Merged revisions 352643 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-25 21:18 +0000 [r352616] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, main/test.c: Avoid unnecessary rebuilds of main/test.c.
+ main/test.c includes "asterisk/version.h", when it should include
+ "asterisk/ast_version.h" instead (and it should use the
+ ast_get_version() and ast_get_version_num() functions). This
+ commit modifies it to extract the Asterisk version information
+ using the proper APIs, and as a result means that main/test.c no
+ longer needs to be rebuilt when a Subversion checkout is updated
+ or modified. ........ Merged revisions 352612 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-25 17:30 +0000 [r352556] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Remove some extraneous debugging from
+ registry memleak fix ........ Merged revisions 352551 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-25 17:16 +0000 [r352520] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c, CHANGES, main/message.c,
+ channels/sip/include/sip.h: Fixes for sending SIP MESSAGE outside
+ of calls. * Fix authenticate MESSAGE losing custom headers added
+ by the MESSAGE_DATA function in the authorization attempt. * Pass
+ up better From header contents for SIP to use. Now is in the
+ "display-name" <URI> format expected by MessageSend. (Note that
+ this is a behavior change that could concievably affect some
+ people.) * Block user from adding standard headers that are added
+ automatically. (To, From,...) * Allow the user to override the
+ Content-Type header contents sent by MessageSend. * Decrement
+ Max-Forwards header if the user transferred it from an incoming
+ message. * Expand SIP short header names so the dialplan and
+ other code only has to deal with the full names. * Documents what
+ SIP expects in the MessageSend(from) parameter. (closes issue
+ ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917)
+ Reported by: Shaun Clark Review:
+ https://reviewboard.asterisk.org/r/1683/
+
+2012-01-25 16:54 +0000 [r352516] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/format.c, main/format_cap.c, main/format_pref.c: Eliminate
+ unnecessary rebuilds of main/format*.c. These files have no need
+ to include "asterisk/version.h", and doing so forces them to be
+ rebuilt each time a Subversion checkout moves between 'modified'
+ and 'unmodified' states.
+
+2012-01-25 16:49 +0000 [r352515] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Clean up some SIP registry-related memory
+ leaks 1) Be sure and free at unload the epa_backend we allocate
+ at startup 2) Do the same sip_registry cleanup at unload we do at
+ reload Review: https://reviewboard.asterisk.org/r/1689/ ........
+ Merged revisions 352514 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-25 16:41 +0000 [r352512] Jonathan Rose <jrose at digium.com>
+
+ * /, configs/sip.conf.sample: Redocuments sip types peer, user,
+ friend in sip.conf.sample There was faulty information in the
+ sample config describing user as a synonym for friend so it has
+ been changed to better elaborate on the differences between the
+ three entity types. (closes issue ASTERISK-15537) Reported by:
+ yarique ........ Merged revisions 352511 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-24 22:22 +0000 [r352430] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Don't do a DNS lookup on an outbound
+ REGISTER host if there is an outbound proxy configured. (closes
+ issue ASTERISK-16550) reported by: Olle Johansson ........ Merged
+ revisions 352424 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-24 20:35 +0000 [r352373] Jonathan Rose <jrose at digium.com>
+
+ * /, sounds/Makefile: Set core sounds version to 1.4.22. Now that
+ we have the right license for the Russian 1.4.22 sounds as well
+ as the sounds for the Australian English 1.4.22 sounds, we can
+ finally set the sounds to use 1.4.22! (closes issue
+ ASTERISK-18978) Reported by: Cameron Twomey Patches:
+ confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002
+ uploaded by Cameron Twomey ........ Merged revisions 352367 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-24 17:02 +0000 [r352292] Richard Mudgett <rmudgett at digium.com>
+
+ * /, funcs/func_odbc.c: Fix locking issues with channel datastores
+ in func_odbc.c. * Fixed a potential memory leak when an existing
+ datastore is manually destroyed by inline code instead of calling
+ ast_datastore_free(). (closes issue ASTERISK-17948) Reported by:
+ Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/
+ ........ Merged revisions 352291 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-23 20:30 +0000 [r352228-352231] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/features.c: Fix grammar of comment. ........ Merged
+ revisions 352230 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/features.c: Fix blind transfers from failing if an 'h'
+ extension is present. This prevents the 'h' extension from being
+ run on the transferee channel when it is transferred via a native
+ transfer mechanism such as SIP REFER. (closes ASTERISK-19173)
+ Reported by: Ross Beer Tested by: Kristjan Vrban Patches:
+ ASTERISK-19173 by Mark Michelson (license 5049) Review:
+ https://reviewboard.asterisk.org/r/1685 ........ Merged revisions
+ 352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-23 19:18 +0000 [r352149] Matthew Jordan <mjordan at digium.com>
+
+ * /, res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17,
+ V27, V29) before starting spandsp layer While the FAXOPT function
+ could be used to set the modem capabilities, the input to that
+ function was not being applied correctly to the spandsp layer.
+ This patch applies the current model capabilities before starting
+ the spandsp layer. (closes issue: ASTERISK-16409) Reported by:
+ Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson
+ Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license
+ 5081) spandsp-modems-10.diff uploaded by mnicholson (license
+ 5081) ........ Merged revisions 352144 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-23 17:34 +0000 [r352091] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the
+ defined enum values. The invalid value used when notifycid was
+ enabled was benign. As far as the code was concerned -1 and 1 are
+ equivalent. (closes issue ASTERISK-19232) Reported by: Eike
+ Kuiper ........ Merged revisions 352090 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-21 00:21 +0000 [r352035] Richard Mudgett <rmudgett at digium.com>
+
+ * /, funcs/func_timeout.c, main/app.c: Fix ast_app_dtget() time
+ unit inconsistency. Note: Noone calls ast_app_dtget() with the
+ timeout parameter of zero so the bad code normally will never get
+ executed. * Fix unnecessary floating point division in
+ func_timeout.c timeout_write() when all other values are
+ integers. (closes issue ASTERISK-16817) Reported by: Dmitry
+ Andrianov ........ Merged revisions 352029 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-21 00:08 +0000 [r352015-352017] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Remove XXX comment that is not necessary.
+ ........ Merged revisions 352016 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Fix RTP reference leak. If a blind
+ transfer were initiated using a REFER without a prior reINVITE to
+ place the call on hold, AND if Asterisk were sending RTCP
+ reports, then there was a reference for the RTP instance of the
+ transferer. This fixes the issue by merging two similar but
+ slightly conflicting sections of code into a single area. It also
+ adds a stop_media_flows() call in the case that the transferer's
+ UA never sends a BYE to us like it is supposed to. (issue
+ ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/
+ ........ Merged revisions 352014 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-20 19:35 +0000 [r351816-351861] Kinsey Moore <kmoore at digium.com>
+
+ * /, codecs/ilbc/iLBC_test.c: More corrections for the ilbc code
+ These changes are in a file that is not compiled by default, and
+ so were missed on earlier checks. ........ Merged revisions
+ 351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /: Recorded merge of revisions 351858 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Allow
+ ilbc code to build under dev mode GCC 4.6.3 found some set/unused
+ variables in the ILBC code.
+
+ * codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Restore
+ LSF_check function calls from set/unused variable removal These
+ functions are not noops and modify the array that is passed in.
+ Thanks for the catch Richard.
+
+ * codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Remove more
+ set, but unused variables in the ilbc codec GCC 4.6.3 caught
+ these in dev mode as well.
+
+2012-01-20 15:59 +0000 [r351762] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c: Adds setting of mwi_from field to
+ check_auth_result check_peer_ok (closes ASTERISK-19057) Reported
+ By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license
+ 5242) ........ Merged revisions 351759 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-20 15:54 +0000 [r351761] Matthew Jordan <mjordan at digium.com>
+
+ * codecs/ilbc/helpfun.c, /: Remove unused variable 'tmp' from
+ helpfun in ilbc codec gcc version 4.6.2 caught an unused variable
+ in the ilbc codec library. This would prevent compilation with
+ --enable-dev-mode; variable removed. ........ Merged revisions
+ 351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-20 13:01 +0000 [r351708] Stefan Schmidt <sst at sil.at>
+
+ * /, contrib/asterisk-ng-doxygen: enable doxygen build for files in
+ the channels/sip folder like reqresp_parser.c ........ Merged
+ revisions 351707 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-19 23:25 +0000 [r351646] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor
+ fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in
+ get_calleridname() parsing and ensure that the output buffer is
+ nul terminated. * Make get_calleridname() truncate the name it
+ parses if the given buffer is too small rather than abandoning
+ the parse and not returning anything for the name. Adjusted
+ get_calleridname_test() unit test to handle the truncation
+ change. * Fix get_in_brackets_test() unit test to check the
+ results of get_in_brackets() correctly. * Fix
+ parse_name_andor_addr() to not return the address of a local
+ buffer. This function is currently not used. * Fix potential NULL
+ pointer dereference in sip_sendtext(). * No need to
+ memset(calleridname) in check_user_full() or tmp_name in
+ get_name_and_number() because get_calleridname() ensures that it
+ is nul terminated. * Reply with an accurate response if
+ get_msg_text() fails in receive_message(). This is academic in
+ v1.8 because get_msg_text() can never fail. ........ Merged
+ revisions 351618 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-19 22:43 +0000 [r351612] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_rtp_asterisk.c, /: Correct output of RTCP jitter
+ statistics in SR and RR reports Change the RTCP RR and SR
+ generation code to convert Asterisk's internal jitter statistics
+ to be represented in RTP timestamp units based on the rate of the
+ codec in use instead of in seconds. (closes issue ASTERISK-14530)
+ ........ Merged revisions 351611 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-19 21:47 +0000 [r351560] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c, include/asterisk/netsock2.h: Eliminates
+ doubling the :port part of SIP Notify Message-Account headers.
+ This patch prevents the domain string from getting mangled during
+ the initreqprep step by moving the initialization to before its
+ immediate use. It also documents this pitfall for the
+ ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported
+ by: Yuri Review: https://reviewboard.asterisk.org/r/1678/
+ ........ Merged revisions 351559 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-19 21:12 +0000 [r351505] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Prevent crash when an SDP offer is
+ received with an encrypted video stream when support for video is
+ disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
+ Reported by: Catalin Sanda ........ Merged revisions 351504 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-18 21:05 +0000 [r351451] Matthew Jordan <mjordan at digium.com>
+
+ * codecs/ilbc/helpfun.c (added), codecs/ilbc/LICENSE_ADDENDUM
+ (added), codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c
+ (added), codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c
+ (added), codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h
+ (added), codecs/ilbc/constants.c (added),
+ codecs/ilbc/iLBC_decode.c (added), codecs/ilbc/createCB.h
+ (added), codecs/ilbc/constants.h (added),
+ codecs/ilbc/iLBC_decode.h (added), codecs/ilbc/iCBSearch.c
+ (added), codecs/ilbc/filter.c (added), codecs/ilbc/hpInput.c
+ (added), codecs/ilbc/gainquant.c (added), codecs/ilbc/iCBSearch.h
+ (added), codecs/ilbc/hpOutput.c (added), codecs/ilbc/rfc3951.txt
+ (added), codecs/ilbc/filter.h (added), codecs/ilbc/hpInput.h
+ (added), codecs/ilbc/LPCencode.c (added), codecs/ilbc/gainquant.h
+ (added), codecs/codec_ilbc.c, codecs/ilbc/hpOutput.h (added),
+ codecs/ilbc/StateSearchW.c (added), codecs/ilbc/PATENTS (added),
+ contrib/scripts/get_ilbc_source.sh, codecs/ilbc/LPCencode.h
+ (added), codecs/ilbc/LICENSE (added), codecs/ilbc/StateSearchW.h
+ (added), codecs/ilbc/iCBConstruct.c (added),
+ codecs/ilbc/syntFilter.c (added), /, codecs/ilbc/iCBConstruct.h
+ (added), codecs/ilbc/iLBC_test.c (added),
+ codecs/ilbc/syntFilter.h (added), codecs/ilbc/StateConstructW.c
+ (added), codecs/ilbc/packing.c (added),
+ codecs/ilbc/StateConstructW.h (added), codecs/ilbc/packing.h
+ (added), codecs/ilbc/getCBvec.c (added), codecs/ilbc/LPCdecode.c
+ (added), codecs/ilbc/enhancer.c (added), codecs/ilbc/lsf.c
+ (added), codecs/ilbc/iLBC_encode.c (added),
+ codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added),
+ codecs/ilbc/enhancer.h (added), codecs/ilbc/FrameClassify.c
+ (added), codecs/ilbc/iLBC_define.h (added), codecs/ilbc/lsf.h
+ (added), codecs/ilbc/extract-cfile.awk (added),
+ codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile,
+ codecs/ilbc/FrameClassify.h (added): Include iLBC source code for
+ distribution with Asterisk This patch includes the iLBC source
+ code for distribution with Asterisk. Clarification regarding the
+ iLBC source code was provided by Google, and the appropriate
+ licenses have been included in the codecs/ilbc folder. Review:
+ https://reviewboard.asterisk.org/r/1675 Review:
+ https://reviewboard.asterisk.org/r/1649 (closes issue:
+ ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan
+ ........ Merged revisions 351450 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-18 15:57 +0000 [r351408] Stefan Schmidt <sst at sil.at>
+
+ * /, channels/chan_sip.c: The get_pai function in chan_sip.c didn't
+ recognized a proper callerid name and number from a
+ P-Asserted-Identity cause the header parsing logic was wrong.
+ Changing the parsing functions to the sip header parsing APIs in
+ reqresp_parser.h solves this problem. Review:
+ https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and
+ Mark Michelson ........ Merged revisions 351396 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-17 17:22 +0000 [r351308] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_rtp_asterisk.c, /: Eliminate odd initialization of
+ probation variable. ........ Merged revisions 351306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-17 17:08 +0000 [r351289] Jonathan Rose <jrose at digium.com>
+
+ * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: Adds
+ pjmedia probation concepts to res_rtp_asterisk's learning mode.
+ In order to better handle RTP sources with strictrtp enabled
+ (which is now default in 10) using the learning mode to figure
+ out new sources when they change is handled by checking for a
+ number of consecutive (by sequence number) packets received to an
+ rtp struct based on a new configurable value called 'probation'.
+ Also, during learning mode instead of liberally accepting all
+ packets received, we now reject packets until a clear source has
+ been determined. Review: https://reviewboard.asterisk.org/r/1663/
+ ........ Merged revisions 351287 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-17 16:54 +0000 [r351286] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Use built-in parsing functions for
+ Contact and Record-Route headers. If a Contact or a Record-Route
+ header had a quoted string with an item in angle brackets, then
+ we would mis-parse it. For instance, "Bob <1234>"
+ <1234 at example.org> would be misparsed as having the URI "1234"
+ The fix for this is to use parsing functions from
+ reqresp_parser.h since they are heavily tested and are awesome.
+ (issue ASTERISK-18990) ........ Merged revisions 351284 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-17 16:07 +0000 [r351234] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Fix udptl issue with initial INVITE
+ introduced by r351027 When an inital INVITE occurs that contains
+ image media, a channel is not yet associated with the SIP dialog.
+ The file descriptor associated with the udptl session needs to be
+ set in initialize_udptl or in sip_new to account for this
+ scenario. ........ Merged revisions 351233 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-17 01:43 +0000 [r351183] Russell Bryant <russell at russellbryant.com>
+
+ * /, channels/chan_sip.c: Merged revisions 351182 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012)
+ | 22 lines Add some missing locking in chan_sip. This patch adds
+ some missing locking to the function
+ send_provisional_keepalive_full(). This function is called from
+ the scheduler, which is processed in the SIP monitor thread. The
+ associated channel (or pbx) thread will also be using the same
+ sip_pvt and ast_channel so locking must be used. The
+ sip_pvt_lock_full() function is used to ensure proper locking
+ order in a safe manner. In passing, document a suspected
+ reference counting error in this function. The "fix" is left
+ commented out because when the "fix" is present, crashes occur.
+ My theory is that fixing it is exposing a reference counting
+ error elsewhere, but I don't know where. (Or my analysis of this
+ being a problem could have been completely wrong in the first
+ place). Leave the comment in the code for so that someone may
+ investigate it again in the future. Also add a bit of doxygen to
+ transmit_provisional_response(). (closes issue ASTERISK-18979)
+ Review: https://reviewboard.asterisk.org/r/1648 ........
+
+2012-01-16 21:17 +0000 [r351081-351131] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200
+ response to INVITE When handling a non-2xx final response on an
+ INVITE transaction, we have to keep the transaction around after
+ we send an ACK in case we receive a retransmission of the
+ response so we can re-transmit the ACK, but also tear down the
+ ast_channel as soon as we transmit the ACK. Before this patch, we
+ could fail at both of these things. Calling
+ sip_alreadygone/needdestroy prevented us from keeping the
+ transaction up and retransmitting the ACK, and queueing
+ CONGESTION was not sufficient to cause the channel to be torn
+ down when originating calls via the CLI, for example. This patch
+ queues a hangup with CONGESTION instead of just queueing
+ CONGESTION for these responses and removes the sip_alreadygone
+ and sip_needdestroy calls from handle_response_invite on non-2xx
+ responses. It relies on the hangup calling sip_scheddestroy. For
+ more information, see section 17.1.1.1 of RFC 3261. (closes issue
+ ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/
+ ........ Merged revisions 351130 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Don't prematurely stop SIP session timer
+ When Asterisk is the UAS (incoming call, endpoint is re-inviting)
+ the SIP session timer expires after half the time the sip
+ endpoint indicates in the Session-expires header in
+ proc_session_timer(). The session timer was being stopped totally
+ and being handled as an error case instead of running again until
+ the second expiry. This patch treats the half-time expiry as a
+ non-error case and continues the timer until the true expiry.
+ (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested
+ by: Thomas Arimont Patches: session_timer_fix.diff by Terry
+ Wilson (License #5357) based on session_timer.patch by Thomas
+ Arimont (License #5525) ........ Merged revisions 351080 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-16 19:12 +0000 [r351028] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Create and initialize udptl only when
+ dialog negotiates for image media Prior to this patch, the udptl
+ struct was allocated and initialized when a dialog was associated
+ with a peer that supported T.38, when a new SIP channel was
+ allocated, or what an INVITE request was received. This resulted
+ in any dialog associated with a peer that supported T.38 having
+ udptl support assigned to it, including the UDP ports needed for
+ communication. This occurred even in non-INVITE dialogs that
+ would never send image media. This patch creates and initializes
+ the udptl structure only when the SDP for a dialog specifies that
+ image media is supported, or when Asterisk indicates through the
+ appropriate control frame that a dialog is to support T.38.
+ (closes issue ASTERISK-16698) Reported by: under Tested by:
+ Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan
+ (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar
+ Broad Tested by: Stefan Schmidt review:
+ https://reviewboard.asterisk.org/r/1668/ ........ Merged
+ revisions 351027 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-16 17:11 +0000 [r350978] Sean Bright <sean at malleable.com>
+
+ * main/db.c: Sort the output of 'database showkey' as well. You can
+ pass wildcards (%) to the database CLI commands, so this will
+ sort the returned list of matches.
+
+2012-01-16 17:06 +0000 [r350976] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp_engine.c, /: Add missing code to set direct RTP setup
+ information during dialing. ........ Merged revisions 350975 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-16 14:27 +0000 [r350938] Sean Bright <sean at malleable.com>
+
+ * main/db.c: Sort the output of 'database show' by key. This more
+ closely mimics the behavior of 'database show' before the
+ conversion to sqlite3.
+
+2012-01-15 20:12 +0000 [r350886-350889] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, main/asterisk.c: Allow only one thread at a time to do
+ asterisk cleanup/shutdown. Add locking around the
+ really-really-quit part of the core stop/restart part. Previously
+ more than one thread could be called to do cleanup, causing
+ atexit handlers to be run multiple times, in turn causing
+ segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson
+ Review: https://reviewboard.asterisk.org/r/1662/ Review:
+ https://reviewboard.asterisk.org/r/1658/ ........ Merged
+ revisions 350888 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, utils/extconf.c: Fix -Werror=unused-but-set-variable compile
+ error in utils/extconf.c. Note that I'm not confirming legitimacy
+ of having that file in tree at all. Is anyone using
+ aelparse/conf2ael? (issue ASTERISK-15350) ........ Merged
+ revisions 350885 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-14 16:41 +0000 [r350790-350838] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac,
+ autoconf/libcurl.m4: Ensure that all AC_LANG_PROGRAM calls in the
+ configure script are properly quoted. Recent versions of autoconf
+ (2.68 on my system) won't properly process the configure script
+ unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in
+ the script were, but many were not. This patch corrects the
+ unquoted calls. ........ Merged revisions 350837 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * contrib/scripts/install_prereq, /, channels/chan_h323.c,
+ addons/chan_mobile.c, res/res_pktccops.c: Multiple revisions
+ 350788-350789 ........ r350788 | kpfleming | 2012-01-14 09:22:33
+ -0600 (Sat, 14 Jan 2012) | 8 lines Ensure that two prerequisites
+ are properly installed on Debian-style distributions. * Don't
+ specify a specific version of libgmime; newer versions are
+ available now and acceptable. * Install libsrtp so that res_srtp
+ can be built. ........ r350789 | kpfleming | 2012-01-14 09:23:32
+ -0600 (Sat, 14 Jan 2012) | 3 lines Correct some
+ 'set-but-not-used' variable warnings. ........ Merged revisions
+ 350788-350789 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 22:10 +0000 [r350737] Kinsey Moore <kmoore at digium.com>
+
+ * /, include/asterisk/autoconfig.h.in: Run bootstrap.sh for the for
+ the ASTERISK-18929 fix configure and autoconfig.h.in were not
+ regenerated when the fix was committed. ........ Merged revisions
+ 350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 21:51 +0000 [r350734] Richard Mudgett <rmudgett at digium.com>
+
+ * /, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample:
+ Correct eventtype names in cel_odbc and cel_pgsql sample files
+ ........ Merged revisions 350733 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 21:41 +0000 [r350731] Kinsey Moore <kmoore at digium.com>
+
+ * /, configure.ac, bootstrap.sh, main/asterisk.c: Make sure
+ asterisk builds on OpenBSD OpenBSD defines SO_PEERCRED, but it
+ returns a 'struct sockpeercred', not 'struct ucred', which causes
+ compilation of main/asterisk.c to fail in read_credentials().
+ This allows configure to check for sockpeercred and asterisk to
+ deal with it properly. (closes issue ASTERISK-18929) Reported-by:
+ Barry Miller Patch-by: Barry Miller ........ Merged revisions
+ 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 20:31 +0000 [r350680] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/sip/config_parser.c: Set port to a default sane value
+ if a bogus one is provided when parsing hostnames. ........
+ Merged revisions 350679 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 17:29 +0000 [r350585] Richard Mudgett <rmudgett at digium.com>
+
+ * configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c,
+ configs/cel.conf.sample, /, cel/cel_manager.c,
+ configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
+ main/cel.c, configs/cel_custom.conf.sample: Add missing CEL
+ logging fields to various CEL backends. Multiple revisions
+ 350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51
+ -0600 (Fri, 13 Jan 2012) | 12 lines Add missing CEL logging
+ fields to various CEL backends. * Add missing eventextra to
+ cel_psql.c and cel_odbc.c. * Add missing PeerAccount and
+ EventExtra to cel_manager.c. * Add missing userdeftype support
+ for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample.
+ (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman
+ ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13
+ Jan 2012) | 8 lines Use compatible names for event extra data for
+ various CEL backends. * Change eventextra to extra in cel_psql.c
+ and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c.
+ (issue ASTERISK-17190) ........ Merged revisions 350555,350571
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-13 16:59 +0000 [r350550-350553] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/app_queue.c: Realtime queues failed to load queue
+ information without queue member table Previously, realtime
+ queues could be loaded without defining the queue member table.
+ This allowed for queue members to be dynamic, while the realtime
+ queue definitions could exist in some backing storage. Revision
+ 342223 broke this when it changed the return value for
+ realtime_multientry to return NULL when no results are returned.
+ Previously, an empty ast_config object was expected. (closes
+ issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene
+ Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt
+ Jordan (license 6283) ........ Merged revisions 350552 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * bridges/bridge_builtin_features.c, channels/chan_bridge.c,
+ include/asterisk/bridging.h, apps/app_confbridge.c,
+ main/bridging.c: Fix crash from bridge channel hangup race
+ condition in ConfBridge This patch addresses two issues in
+ ConfBridge and the channel bridge layer: 1. It fixes a race
+ condition wherein the bridge channel could be hung up 2. It
+ removes the deadlock avoidance from the bridging layer and makes
+ the bridge_pvt an ao2 ref counted object Patch by David Vossel
+ (mjordan was merely the commit monkey) (issue ASTERISK-18988)
+ (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested
+ by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by
+ David Vossel (license 5628) (closes issue ASTERISK-19100)
+ Reported by: Matt Jordan Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1654/
+
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