[asterisk-commits] bebuild: tag 1.8.10.0-rc1 r353310 - /tags/1.8.10.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jan 30 08:13:29 CST 2012
Author: bebuild
Date: Mon Jan 30 08:13:25 2012
New Revision: 353310
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=353310
Log:
Importing files for 1.8.10.0-rc1 release.
Added:
tags/1.8.10.0-rc1/.lastclean (with props)
tags/1.8.10.0-rc1/.version (with props)
tags/1.8.10.0-rc1/ChangeLog (with props)
Added: tags/1.8.10.0-rc1/.lastclean
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--- tags/1.8.10.0-rc1/ChangeLog (added)
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+2012-01-30 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.10.0-rc1 Released.
+
+2012-01-30 12:42 +0000 [r353260] 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).
+
+2012-01-29 02:42 +0000 [r353175] 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().
+
+2012-01-28 14:49 +0000 [r353126] 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.
+
+2012-01-28 04:25 +0000 [r353077] 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
+
+2012-01-27 19:12 +0000 [r352959] 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)
+
+2012-01-27 18:22 +0000 [r352955] Richard Mudgett <rmudgett at digium.com>
+
+ * 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, res/res_srtp.c, main/pbx.c, channels/chan_sip.c,
+ include/asterisk/indications.h: 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/
+
+2012-01-27 00:05 +0000 [r352862] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: 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 20:14 +0000 [r352807] 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)
+
+2012-01-26 19:06 +0000 [r352755] 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
+
+2012-01-26 06:27 +0000 [r352704] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/chan_sip.c: 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 22:21 +0000 [r352643] Paul Belanger <pabelanger at digium.com>
+
+ * apps/app_voicemail.c: Fix -Werror=unused-but-set-variable
+ compiler error (gcc 4.6.2)
+
+2012-01-25 21:16 +0000 [r352612] 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.
+
+2012-01-25 17:28 +0000 [r352514-352551] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Remove some extraneous debugging from
+ registry memleak fix
+
+ * 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/
+
+2012-01-25 16:39 +0000 [r352511] 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
+
+2012-01-24 22:17 +0000 [r352424] 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
+
+2012-01-24 20:33 +0000 [r352367] 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
+
+2012-01-24 16:59 +0000 [r352291] 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/
+
+2012-01-24 16:30 +0000 [r352287] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Move RTP timeout check to before bridged
+ channel check so it is actually executed. (issue ASTERISK-19179)
+ Reported by: TSAREGORODTSEV Yury (closes issue ASTERISK-14534)
+ Reported by: kriborgen Patches: chan_sip.patch uploaded by
+ kriborgen (license 6138)
+
+2012-01-23 20:30 +0000 [r352199-352230] Mark Michelson <mmichelson at digium.com>
+
+ * main/features.c: Fix grammar of comment.
+
+ * 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
+
+2012-01-23 19:12 +0000 [r352144] 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)
+
+2012-01-23 17:33 +0000 [r352090] 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
+
+2012-01-21 00:20 +0000 [r352029] Richard Mudgett <rmudgett at digium.com>
+
+ * main/app.c, funcs/func_timeout.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
+
+2012-01-21 00:08 +0000 [r352014-352016] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Remove XXX comment that is not necessary.
+
+ * 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/
+
+2012-01-20 19:34 +0000 [r351858-351860] 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.
+
+ * codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Allow ilbc
+ code to build under dev mode GCC 4.6.3 found some set/unused
+ variables in the ILBC code.
+
+2012-01-20 16:01 +0000 [r351765] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: Accidentally left off a semicolon only in
+ 1.8 somehow for previous patch.
+
+2012-01-20 15:48 +0000 [r351760] 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.
+
+2012-01-20 15:42 +0000 [r351759] 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)
+
+2012-01-20 12:59 +0000 [r351707] Stefan Schmidt <sst at sil.at>
+
+ * contrib/asterisk-ng-doxygen: enable doxygen build for files in
+ the channels/sip folder like reqresp_parser.c
+
+2012-01-19 23:17 +0000 [r351618] 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.
+
+2012-01-19 22:36 +0000 [r351611] 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)
+
+2012-01-19 21:46 +0000 [r351559] Jonathan Rose <jrose at digium.com>
+
+ * include/asterisk/netsock2.h, channels/chan_sip.c: 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/
+
+2012-01-19 21:11 +0000 [r351504] 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
+
+2012-01-18 20:54 +0000 [r351450] Matthew Jordan <mjordan at digium.com>
+
+ * 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), 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/gainquant.h (added), codecs/ilbc/LPCencode.c (added),
+ codecs/ilbc/hpInput.h (added), codecs/codec_ilbc.c,
+ codecs/ilbc/PATENTS (added), codecs/ilbc/StateSearchW.c (added),
+ codecs/ilbc/hpOutput.h (added),
+ contrib/scripts/get_ilbc_source.sh, codecs/ilbc/LICENSE (added),
+ codecs/ilbc/LPCencode.h (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): 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
+
+2012-01-18 14:57 +0000 [r351396] 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
+
+2012-01-17 17:22 +0000 [r351306] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_rtp_asterisk.c: Eliminate odd initialization of probation
+ variable.
+
+2012-01-17 16:55 +0000 [r351287] Jonathan Rose <jrose at digium.com>
+
+ * CHANGES, res/res_rtp_asterisk.c, configs/rtp.conf.sample: 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/
+
+2012-01-17 16:41 +0000 [r351284] 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)
+
+2012-01-17 16:06 +0000 [r351233] 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.
+
+2012-01-17 01:37 +0000 [r351182] Russell Bryant <russell at russellbryant.com>
+
+ * channels/chan_sip.c: 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 21:12 +0000 [r351080-351130] 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/
+
+ * 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)
+
+2012-01-16 19:09 +0000 [r351027] 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/
+
+2012-01-16 17:04 +0000 [r350975] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp_engine.c: Add missing code to set direct RTP setup
+ information during dialing.
+
+2012-01-15 20:07 +0000 [r350885-350888] 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/
+
+ * 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)
+
+2012-01-14 16:40 +0000 [r350788-350837] Kevin P. Fleming <kpfleming at digium.com>
+
+ * autoconf/libcurl.m4, configure, autoconf/ast_gcc_attribute.m4,
+ configure.ac: 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.
+
+ * addons/chan_mobile.c, channels/chan_h323.c: Correct some
+ 'set-but-not-used' variable warnings.
+
+ * contrib/scripts/install_prereq: 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.
+
+2012-01-13 22:05 +0000 [r350736] Kinsey Moore <kmoore at digium.com>
+
+ * configure, 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.
+
+2012-01-13 21:51 +0000 [r350733] 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
+
+2012-01-13 21:40 +0000 [r350730] Kinsey Moore <kmoore at digium.com>
+
+ * bootstrap.sh, main/asterisk.c, configure.ac: 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
+
+2012-01-13 20:29 +0000 [r350679] 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.
+
+2012-01-13 17:23 +0000 [r350555-350571] Richard Mudgett <rmudgett at digium.com>
+
+ * configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
+ cel/cel_pgsql.c, cel/cel_odbc.c, cel/cel_manager.c: 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)
+
+ * configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
+ main/cel.c, configs/cel_custom.conf.sample, cel/cel_pgsql.c,
+ configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c,
+ configs/cel.conf.sample, cel/cel_manager.c: 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
+
+2012-01-13 16:57 +0000 [r350552] 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)
+
+2012-01-12 15:57 +0000 [r350501] Jonathan Rose <jrose at digium.com>
+
+ * main/features.c: Adds peer to CEL report on CEL_BRIDGE_START and
+ CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic
+ Colledge Patches: features_18.patch uploaded by Nic Colledge
+ (license 6245)
+
+2012-01-11 22:50 +0000 [r350311-350452] Richard Mudgett <rmudgett at digium.com>
+
+ * main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a
+ CEL dummy channel. (closes issue ASTERISK-19180) Reported by:
+ Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license
+ #5909) patch uploaded by Corey Farrell
+
+ * CHANGES, apps/app_followme.c, apps/app_dial.c: Make FollowMe
+ optionally update connected line information when the accepting
+ endpoint is bridged. Like Dial and Queue, FollowMe needs to deal
+ with AST_CONTROL_CONNECTED_LINE information so when the parties
+ are initially bridged, the connected line information will be
+ correct. * Added the 'I' option just like the app_dial and
+ app_queue 'I' option. (closes issue ASTERISK-18969) Reported by:
+ rmudgett Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1656/
+
+ * funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK
+ function. The time passed by the LOCK function to an internal
+ function was relative time when the function expected absolute
+ time. * Don't use C++ keywords in get_lock(). (closes issue
+ ASTERISK-16868) Reported by: Andrey Solovyev Patches:
+ 20101102__issue18207.diff.txt (license #5003) patch uploaded by
+ Andrey Solovyev (modified)
+
+2012-01-09 21:54 +0000 [r350075-350220] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c: Fix joinable thread terminating without
+ joiner memory leak in chan_iax.c. The iax2_process_thread() can
+ exit without anyone waiting to join the thread. If noone is
+ waiting to join the thread then a large memory leak occurs. *
+ Made iax2_process_thread() deatach itself if nobody is waiting to
+ join the thread. (closes issue ASTERISK-17339) Reported by:
+ Tzafrir Cohen Patches:
+ asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch
+ (license #5617) patch uploaded by Alex Villacis Lasso (modified)
+ (closes issue ASTERISK-17825) Reported by: wangjin
+
+ * contrib/scripts/live_ast: live_ast: valgrind: run asterisk under
+ valgrind Adds a new sub-command, "valgrind" to live_ast. It runs
+ asterisk under valgrind. The extra command-line parameters are
+ passed to Asterisk as usual, and parameters to valgrind are
+ passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review:
+ https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636
+ from http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * contrib/scripts/live_ast, contrib/scripts/valgrind_compare
+ (added): Update contrib script live_ast to invoke Asterisk with
+ valgrind and suppression file. * Added valgrind_compare script to
+ compare two valgrind log files for differences. (issue
+ ASTERISK-17339) Reported by: Tzafrir Cohen Patches:
+ valgrind_compare (license #5035) script uploaded by Tzafrir Cohen
+ live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir
+ Cohen live_ast_valgrind_v2.diff (license #5185) patch uploaded by
+ Paul Belanger
+
+ * main/asterisk.c: Make Asterisk -x command line parameter imply -r
+ parameter presence. The Asterisk -x command line parameter is
+ documented inconsistently. * Made the -x documentation and
+ behavior consistent. * Since this is also a new year, updated the
+ copyright notices while here. (closes issue ASTERISK-19094)
+ Reported by: Eugene Patches:
+ issueA19094_correct_asterisk_option_x.patch (license #5674) patch
+ uploaded by Walter Doekes (modified) Tested by: Eugene
+
+2012-01-09 15:37 +0000 [r350023] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_meetme.c: Prevent SLA settings from getting wiped out on
+ reload If SLA was reloaded without the config file being changed,
+ current settings got wiped out before the SLA reload code decided
+ it wasn't going to reload the file since nothing was changed.
+ Moving the settings reset later in the reload process fixes this.
+ (closes issue AST-744)
+
+2012-01-06 23:17 +0000 [r349968] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Don't leak CID in From header when
+ presentation=unavailable When someone does
+ Set(CALLERPRES()=unavailable) (or
+ Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From
+ header shows "Anonymous" <anonymous at anonymous.invalid>. When
+ sendrpid=yes/pai, the From header will still display the callerid
+ info, even though we supply an rpid header with the anonymous
+ info. It seems like we shouldn't leak that info in any case.
+ Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04
+ seems to indicate that one shouldn't send identifying info in the
+ From in this case. This patch anonymizes the From header as well
+ even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review:
+ https://reviewboard.asterisk.org/r/1649/
+
+2012-01-06 16:46 +0000 [r349819-349872] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_followme.c: Fix memory leaks in app_followme
+ find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt
+ Jordan
+
+ * cel/cel_sqlite3_custom.c: Make not assume that the
+ cel_sqlite3_custom SQL table primary key is AcctId. If a table is
+ created by some other application and the primary key is not
+ named "AcctId", cel/cel_sqlite3_custom.c will always try to
+ create the table and fail because it already exists. * Change the
+ SQL table query to not require AcctId as the primary key. (closes
+ issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch
+ (license #6337) patch uploaded by socketpair
+
+2012-01-05 22:06 +0000 [r349731] Kinsey Moore <kmoore at digium.com>
+
+ * main/file.c: Allow playback of formats that don't support seeking
+ ast_streamfile previously did unconditional seeking on files that
+ broke playback of formats that don't support that functionality.
+ This patch avoids the seek that was causing the problem. This
+ regression was introduced in r158062. (closes issue
+ ASTERISK-18994) Patch-by: Timo Teras
+
+2012-01-05 21:46 +0000 [r349672-349728] Jonathan Rose <jrose at digium.com>
+
+ * main/dsp.c: Fix an issue where dsp.c would interpret multiple
+ dtmf events from a single key press. When receiving calls from a
+ mobile phone into a DISA system on a connection with significant
+ interference, the reporter's Asterisk system would interpret DTMF
+ incorrectly and replicate digits received. This patch resolves
+ that by increasing the number of frames a mismatch has to be
+ detected before assuming the DTMF is over by 1 frame and adjusts
+ dtmf_detect function to reset hits and misses only when an edge
+ is detected. (closes issue ASTERISK-17493) Reported by: Alec
+ Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis
+ (license 5546) Review: https://reviewboard.asterisk.org/r/1130/
+
+ * main/asterisk.c: Ensures Asterisk closes when receiving terminal
+ signals in 'no fork' mode. When catching a signal, in no fork
+ mode the console thread is identical to the thread responsible
+ for catching the signal and closing Asterisk, which requires it
+ to first dispense with the console thread. Prior to this patch,
+ if these threads were identical, upon receiving a killing signal,
+ the thread will send an URG signal to itself, which we also catch
+ and then promptly do nothing with. Obviously this isn't useful
+ behavior. (closes issue ASTERISK-19127) Reported By: Bryon Clark
+ Patches: quit_on_signals.patch uploaded by Bryon Clark (license
+ 6157)
+
+2012-01-04 20:46 +0000 [r349558] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Fix segfault in chan_dahdi for
+ CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private
+ pointer checks in the following chan_dahdi channel callbacks:
+ dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
+ dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by:
+ Diego Aguirre Tested by: rmudgett
+
+2012-01-04 20:23 +0000 [r349504-349529] Kinsey Moore <kmoore at digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Make debian init script
+ conform to the LSB standard Previously, this init script would
+ return 1 if Asterisk was already running. This is incorrect
+ behavior according to the LSB standard and has been fixed by
+ returning 0 instead. (closes issue ASTERISK-17958) Reported-by:
+ johnc
+
+ * contrib/scripts/autosupport, contrib/scripts/autosupport.8:
+ Update autosupport script and man page Added information
+ collection from the output of the utilities: top, free, uptime,
+ ifconfig Added information collection from the output of the
+ Asterisk command 'dahdi show status' Added option / flag '-n,
+ --non-interactive' Updated man page to reflect new option / flag
[... 36328 lines stripped ...]
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