[asterisk-commits] bebuild: tag 11.4.0-rc1 r384254 - /tags/11.4.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Mar 28 08:43:30 CDT 2013
Author: bebuild
Date: Thu Mar 28 08:43:26 2013
New Revision: 384254
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=384254
Log:
Importing files for 11.4.0-rc1 release.
Added:
tags/11.4.0-rc1/.lastclean (with props)
tags/11.4.0-rc1/.version (with props)
tags/11.4.0-rc1/ChangeLog (with props)
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+2013-03-28 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.4.0-rc1 Released.
+
+2013-03-27 19:51 +0000 [r384163] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c, main/format_pref.c: Address uninitialized
+ conditional that valgrind found ........ Merged revisions 384162
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-27 18:51 +0000 [r384119] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/http.c: Fix a file descriptor leak in off nominal path
+ While looking at the security vulnerability in ASTERISK-20967,
+ Walter noticed a file descriptor leak and some other issues in
+ off nominal code paths. This patch corrects them. Note that this
+ patch is not related to the vulnerability in ASTERISK-20967, but
+ the patch was placed on that issue. (closes issue ASTERISK-20967)
+ Reported by: wdoekes patches:
+ issueA20967_file_leak_and_unused_wkspace.patch uploaded by
+ wdoekes (License 5674) ........ Merged revisions 384118 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-27 17:06 +0000 [r384049] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_rtp_asterisk.c, /: Fix white noise on SRTP decryption
+ When res_rtp_asterisk.c was altered to avoid attempting to apply
+ unprotect algorithms to non-audio RTP packets, the test used was
+ incorrect. This caused the audio packets to not be decrypted and
+ resulted in loud white noise on the other endpoint (or both
+ endpoints depending on the call legs involved). The test now
+ properly checks the version field in the RTP header to ensure
+ that RTP and RTCP are decrypted while other types of packets are
+ not. (closes issue ASTERISK-21323) Reported by: andrea Tested by:
+ Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff
+ uploaded by Kinsey Moore ........ Merged revisions 384048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-27 15:23 +0000 [r383973-384003] Matthew Jordan <mjordan at digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c,
+ channels/sip/security_events.c: AST-2013-003: Prevent username
+ disclosure in SIP channel driver When authenticating a SIP
+ request with alwaysauthreject enabled, allowguest disabled, and
+ autocreatepeer disabled, Asterisk discloses whether a user exists
+ for INVITE, SUBSCRIBE, and REGISTER transactions in multiple
+ ways. The information is disclosed when: * A "407 Proxy
+ Authentication Required" response is sent instead of a "401
+ Unauthorized" response * The presence or absence of additional
+ tags occurs at the end of "403 Forbidden" (such as "(Bad Auth)")
+ * A "401 Unauthorized" response is sent instead of "403
+ Forbidden" response after a retransmission * Retransmission are
+ sent when a matching peer did not exist, but not when a matching
+ peer did exist. This patch resolves these various vectors by
+ ensuring that the responses sent in all scenarios is the same,
+ regardless of the presence of a matching peer. This issue was
+ reported by Walter Doekes, OSSO B.V. A substantial portion of the
+ testing and the solution to this problem was done by Walter as
+ well - a huge thanks to his tireless efforts in finding all the
+ ways in which this setting didn't work, providing automated
+ tests, and working with Kinsey on getting this fixed. (closes
+ issue ASTERISK-21013) Reported by: wdoekes Tested by: wdoekes,
+ kmoore patches: AST-2013-003-1.8 uploaded by kmoore, wdoekes
+ (License 6273, 5674) AST-2013-003-10 uploaded by kmoore, wdoekes
+ (License 6273, 5674) AST-2013-003-11 uploaded by kmoore, wdoekes
+ (License 6273, 5674)
+
+ * main/http.c: AST-2013-002: Prevent denial of service in HTTP
+ server AST-2012-014, fixed in January of this year, contained a
+ fix for Asterisk's HTTP server for a remotely-triggered crash.
+ While the fix put in place fixed the possibility for the crash to
+ be triggered, a denial of service vector still exists with that
+ solution if an attacker sends one or more HTTP POST requests with
+ very large Content-Length values. This patch resolves this by
+ capping the Content-Length at 1024 bytes. Any attempt to send an
+ HTTP POST with Content-Length greater than this cap will not
+ result in any memory allocation. The POST will be responded to
+ with an HTTP 413 "Request Entity Too Large" response. This issue
+ was reported by Christoph Hebeisen of TELUS Security Labs (closes
+ issue ASTERISK-20967) Reported by: Christoph Hebeisen patches:
+ AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
+ AST-2013-002-10.diff uploaded by mmichelson (License 5049)
+ AST-2013-002-11.diff uploaded by mmichelson (License 5049)
+
+ * res/res_format_attr_h264.c: AST-2013-001: Prevent buffer overflow
+ through H.264 format negotiation The format attribute resource
+ for H.264 video performs an unsafe read against a media attribute
+ when parsing the SDP. The value passed in with the format
+ attribute is not checked for its length when parsed into a fixed
+ length buffer. This patch resolves the vulnerability by only
+ reading as many characters from the SDP value as will fit into
+ the buffer. (closes issue ASTERISK-20901) Reported by: Ulf
+ Harnhammar patches: h264_overflow_security_patch.diff uploaded by
+ jrose (License 6182)
+
+2013-03-26 02:28 +0000 [r383840-383878] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Resolve deadlock between SIP registration
+ and channel based functions In r373424, several reentrancy
+ problems in chan_sip were addressed. As a result, the SIP channel
+ driver is now properly locking the channel driver private
+ information in certain operations that it wasn't previously. This
+ exposed two latent problems either in register_verify or by
+ functions called by register_verify. This includes: * Holding the
+ private lock while calling sip_send_mwi_to_peer. This can create
+ a new sip_pvt via sip_alloc, which will obtain the channel
+ container lock. This is a locking inversion, as any channel
+ related lock must be obtained prior to obtaining the SIP channel
+ technology private lock. Note that this issue was already fixed
+ in Asterisk 11. * Holding the private lock while calling
+ sip_poke_peer. In the same vein as sip_send_mwi_to_peer,
+ sip_poke_peer can create a new SIP private, causing the same
+ locking inversion. Note that this locking inversion typically
+ occured when CLI commands were run while a SIP REGISTER request
+ was being processed, as many CLI commands (such as 'sip show
+ channels', 'core show channels', etc.) have to obtain the channel
+ container lock. (issue ASTERISK-21068) Reported by: Nicolas
+ Bouliane (issue ASTERISK-20550) Reported by: David Brillert
+ (issue ASTERISK-21314) Reported by: Badalian Vyacheslav (issue
+ ASTERISK-21296) Reported by: Gabriel Birke ........ Merged
+ revisions 383863 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/cdr.c, /: Resolve deadlock between pending CDR and batch CDR
+ locks r375757 attempted to resolve a race condition between
+ multiple submissions of CDRs while in batch mode from attempting
+ to destroy the scheduled batch submission by extending the batch
+ CDR lock. Unfortunately, this causes a deadlock between the
+ pending CDR lock and the batch CDR lock. This patch resolves the
+ intent of r375757 by simply providing a new lock that protects
+ the scheduling of the batches. The original batch CDR lock is
+ kept to protect manipulation of the batch CDR settings, but has
+ been placed such that it is not held when the pending lock is
+ held. Thanks to Chase Venters for providing lock analysis on the
+ issue. (issue ASTERISK-21162) Reported by: Chase Venters ........
+ Merged revisions 383839 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-26 01:36 +0000 [r383836] Russell Bryant <russell at russellbryant.com>
+
+ * /, apps/app_meetme.c: Fix multi-station answer race condition.
+ When an SLA trunk is ringing (inbound call on the trunk) Asterisk
+ will make outbound calls to the stations that have that trunk. If
+ more than one station answers the call at the same time, all
+ channels other than the first one to answer are left in a bad
+ state. The channel gets leaked, is not connected to anything, and
+ there's no way to get rid of it. We now properly clean up these
+ losing channels by hanging up on them. Since they lost the race,
+ as we process their answer, there is no ringing trunk for them to
+ answer. ........ Merged revisions 383835 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-25 23:24 +0000 [r383798] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/sig_pri.c: Set the CALLERID(dnid-num-plan) for
+ incoming ISDN calls. The CALLEDTON channel variable is set for
+ incoming ISDN calls to the lower 7 bits of the Q.931
+ type-of-number/numbering-plan octet. The CALLERID(dnid-num-plan)
+ should have the same value. (closes issue ASTERISK-21248)
+ Reported by: rmudgett ........ Merged revisions 383796 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-25 12:36 +0000 [r383668] Sean Bright <sean at malleable.com>
+
+ * res/res_config_curl.c, /: Properly delimit post data in
+ res_config_curl. ........ Merged revisions 383667 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-22 20:41 +0000 [r383631] Michael L. Young <elgueromexicano at gmail.com>
+
+ * apps/app_mixmonitor.c: Fix StopMixMonitor Hanging Up When Unable
+ To Stop MixMonitor On A Channel A regression was accidentally
+ introduced when allowing an optional ID to be used when calling
+ StopMixMonitor. When we are unable to stop MixMonitor on a
+ channel, -1 is being returned which triggers the hangup of the
+ channel. This patch restores the prior behavior by returning 0
+ whether we were successful or not. It also allows the call from
+ the manager to use the return code when the action fails. (closes
+ issue ASTERISK-21294) Reported by: daroz Tested by: daroz
+ Patches: asterisk-21294-stop_mixmonitor_hangingup.diff Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2404/
+
+2013-03-20 20:25 +0000 [r383457-383461] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * funcs/func_curl.c, /: Have func_curl log a warning when a curl
+ request fails. Review: https://reviewboard.asterisk.org/r/2403/
+ ........ Merged revisions 383460 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * funcs/func_curl.c: Minor cleanup in func_curl near hashcompat
+ code. Review: https://reviewboard.asterisk.org/r/2402/
+
+2013-03-19 15:58 +0000 [r383341-383342] David M. Lee <dlee at digium.com>
+
+ * codecs/Makefile: Remove codecs/speex/*.i on make clean
+
+ * codecs/Makefile, /: Removed codecs/g722/*.i on make clean
+ ........ Merged revisions 383340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-16 15:14 +0000 [r383266] Joshua Colp <jcolp at digium.com>
+
+ * res/res_xmpp.c: Fix a bug where resources were not found due to
+ hashing on the priority itself.
+
+2013-03-15 12:51 +0000 [r383166] Kinsey Moore <kmoore at digium.com>
+
+ * main/tcptls.c, main/manager.c, /, channels/chan_sip.c,
+ main/http.c: tcptls: Prevent unsupported options from being set
+ AMI, HTTP, and chan_sip all support TLS in some way, but none of
+ them support all the options that Asterisk's TLS core is capable
+ of interpreting. This prevents consumers of the TLS/SSL layer
+ from setting TLS/SSL options that they do not support. This also
+ gets tlsverifyclient closer to a working state by requesting the
+ client certificate when tlsverifyclient is set. Currently, there
+ is no consumer of main/tcptls.c in Asterisk that supports this
+ feature and so it can not be properly tested. Review:
+ https://reviewboard.asterisk.org/r/2370/ Reported-by: John
+ Bigelow Patch-by: Kinsey Moore (closes issue AST-1093) ........
+ Merged revisions 383165 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-15 01:34 +0000 [r383121-383125] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: When a session timer expires during a
+ T.38 call, re-invite with correct SDP When a session timer
+ expires during a dialog that has re-negotiated to T.38 and
+ Asterisk is the refresher, Asterisk will send a re-INVITE with an
+ SDP containing audio media only. This causes some hilarity with
+ the poor fax session under weigh. This patch corrects that by
+ sending T.38 parameters if we are in the middle of a T.38
+ session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal
+ patches:
+ dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch
+ uploaded by nbansal (License 6418) ........ Merged revisions
+ 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * pbx/pbx_spool.c, /: Fix processing of call files when using
+ KQueue on OS X In certain situations, call files are not
+ processed when using KQueue with pbx_spool. Asterisk was sending
+ an invalid timeout value when the spool directory is empty,
+ causing the call to kevent to error immediately. This can create
+ a tight loop, increasing the CPU load on the system. (closes
+ issue ASTERISK-21176) Reported by: Carlton O'Riley patches:
+ kqueue_osx.patch uploaded by coriley (License 6473) ........
+ Merged revisions 383120 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-14 16:57 +0000 [r383062] Jason Parker <jparker at digium.com>
+
+ * autoconf/ast_ext_lib.m4, /: Fix whitespace in AST_EXT_LIB_CHECK
+ macro. ........ Merged revisions 383061 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-12 21:17 +0000 [r382940-382943] Michael L. Young <elgueromexicano at gmail.com>
+
+ * addons/res_config_mysql.c, /: Fix Sorting Order For Parking Lots
+ Stored In Static Realtime When retrieving the parking lots from a
+ MySQL database table, the current order is "filename, cat_metric
+ desc, var_metric asc, category". If there are multiple parking
+ lots with the same cat_metric but different categories,
+ everything is being sorted on cat_metric first resulting in
+ errors when loading the parking lots. This patch fixes the
+ problem by sorting on the category field first, then the
+ cat_metric field. (closes issue ASTERISK-21035) Reported by: Alex
+ Epshteyn Patches: asterisk-21035-orderby.diff Michael L. Young
+ (license 5026) ........ Merged revisions 382942 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * contrib/realtime/mysql/sippeers.sql, /,
+ contrib/realtime/postgresql/realtime.sql: Update Contributed
+ Realtime Schema Files - IPv6 Addresses This commit updates some
+ fields in the contributed realtime schema files to handle IPv6
+ addresses. (closes issue ASTERISK-21173) Reported by: Torrey
+ Searle Patches: realtime_sql.patch Torrey Searle (license 5334)
+ asterisk-21173-update-ip-fields.diff Michael L. Young (license
+ 5026) ........ Merged revisions 382939 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-12 20:06 +0000 [r382923] Joshua Colp <jcolp at digium.com>
+
+ * res/res_xmpp.c: Fix a crash when res_xmpp is configured using a
+ username without a domain. (closes issue ASTERISK-21156) Reported
+ by: amsoft2001
+
+2013-03-12 16:23 +0000 [r382848] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c, UPGRADE.txt: Include the Username field
+ in SIP Registry events when Status is registered In
+ ASTERISK-17888, the AMI Registry event during SIP registrations
+ was supposed to include the Username field. Somehow, one of the
+ events was missed. This patch corrects that - the Username field
+ should be included in all AMI Registry events involving SIP
+ registrations. (issue ASTERISK-17888) (closes issue
+ ASTERISK-21201) Reported by: Dmitriy Serov patches:
+ chan_sip.c.diff uploaded by Dmitriy Serov (license 6479) ........
+ Merged revisions 382847 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-12 08:53 +0000 [r382827] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c: Fix core dump on CLI usage Fix issue
+ with 'unistim show info' CLI command when device connected not
+ configured
+
+2013-03-08 20:16 +0000 [r382739] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Update the via header when
+ relaying SMS MESSAGE Prior to this change, certain conditions for
+ sending the message would result in an address of '(null)' being
+ used in the via header of the SIP message because a NULl value of
+ pvt->ourip was used when initially generating the via header.
+ This is fixed by adding a call to build_via when the address is
+ set before sending the message. (closes issue ASTERISK-21148)
+ Reported by: Zhi Cheng Patches: 700-sip_msg_send_via_fix.patch
+ uploaded by Zhi Cheng (license 6475)
+
+2013-03-07 17:57 +0000 [r382617] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c: Let vm_mailbox_snapshot combine "Urgent"
+ when no folder is specified r381835 fixed a bug in
+ vm_mailbox_snapshot where combining INBOX and Old forgot that
+ Urgent also "counts" as new messages. This fixed the problem when
+ any of the three folders was specified and the combine option was
+ used. It missed the case where the folder isn't specified and we
+ build a snapshot of all folders. This patch corrects that.
+
+2013-03-07 15:08 +0000 [r382574] Kinsey Moore <kmoore at digium.com>
+
+ * main/logger.c: Ensure that logmsgs are freed properly Messages
+ sent while the logger thread is shutting down will now have their
+ associated callid freed properly.
+
+2013-03-07 14:58 +0000 [r382573] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c: Add a 'secret' probation strictrtp mode
+ to handle delayed changes in RTP source Often, Asterisk may
+ realize that a change in the source of an RTP stream is about to
+ occur and ask that the RTP engine reset it's lock on the current
+ RTP source. In certain scenarios, it may take awhile for the new
+ remote system to send RTP packets, while the old remote system
+ may continue providing RTP during that time period. This causes
+ Asterisk to re-lock onto the old source, thereby rejecting the
+ new source when the old source stops sending RTP and the new
+ source begins. This patch prevents that by having a constant
+ secondary, 'secret' probation mode enabled when an RTP source has
+ been chosen. RTP packets from other sources are always
+ considered, but never chosen unless the current RTP source stops
+ sending RTP. Review: https://reviewboard.asterisk.org/r/2364
+ (closes issue AST-1124) Reported by: John Bigelow Tested by: John
+ Bigelow (closes issue AST-1125) Reported by: John Bigelow Tested
+ by: John Bigelow
+
+2013-03-06 18:28 +0000 [r382514] Kinsey Moore <kmoore at digium.com>
+
+ * /: Recorded merge of revisions 382513 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ Correct app_page documentation The 'A' and 'n' options for Page()
+ mention that the announcement will be played simultaneously. This
+ is not necessarily the case.
+
+2013-03-05 03:51 +0000 [r382410] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c, /: Fix several unreleased mutex locks
+ that cause problem with processing calls Reported by: Daniel
+ Bohling Tested by: Daniel Bohling (Closes issue ASTERISK-21119)
+ ........ Merged revisions 382409 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-03-04 21:12 +0000 [r382390] Jason Parker <jparker at digium.com>
+
+ * /, main/event.c: Fix comparison of presence state in event
+ subsystem. Several new IEs were not given types (or names),
+ causing the comparison function to improperly succeed. This adds
+ those. (closes issue AST-1128)
+
+2013-03-04 20:03 +0000 [r382385] kharwell <kharwell at localhost>:
+
+ * apps/app_confbridge.c: Confbridge CLI new record file name check.
+ This fix checks to make sure that if a confbridge record start
+ command is issued from the CLI it will always use the file name
+ given on the CLI even if it changes between start/stop records
+ for a conference. Previously it had been reusing the same file
+ between start/stops even if a new filename was given. (issue
+ AST-1088) Reported by: John Bigelow
+
+2013-03-01 04:28 +0000 [r382322] Michael L. Young <elgueromexicano at gmail.com>
+
+ * contrib/realtime/mysql/sippeers.sql, channels/chan_sip.c,
+ contrib/realtime/postgresql/realtime.sql, CHANGES: Fix / Clean Up
+ Some Items To Handle The New auto_* NAT Options The original
+ report had to do with a realtime peer behind NAT being pruned and
+ the peer's private address being used instead of its external
+ address. Upon debugging, it was discovered that this was being
+ caused by the addition of the auto_force_rport and auto_comedia
+ settings. This patch does the following: * Adds a missing note to
+ the CHANGES file indicating that the default global nat setting
+ is auto_force_rport * Constify the 'req' parameter for
+ check_via() * Add calls to check_via() in a couple of places in
+ order for the auto_* settings to do their job in attempting to
+ determine if NAT is involved * Set the flags SIP_NAT_FORCE_RPORT
+ and SIP_PAGE2_SYMMETRICRTP if the auto_* settings are in use
+ where it was needed * Moves the copying of peer flags up in
+ build_peer() to before they are used; this fixes the realtime
+ prune issue * Update the contrib/realtime schemas to allow the
+ nat column to handle the different nat setting combinations we
+ have This patch received a review and "Ship It!" on the issue
+ itself. (closes issue ASTERISK-20904) Reported by: JoshE Tested
+ by: JoshE, Michael L. Young Patches:
+ asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young
+ (license 5026)
+
+2013-02-28 21:58 +0000 [r382296-382298] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c: While the ICE negotiation is occurring
+ leave strictrtp in an open state, media can and will come from
+ different places.
+
+ * res/res_rtp_asterisk.c: Fix a bug with ICE and strictrtp where
+ media could get dropped. If the end result of the ICE negotiation
+ resulted in the path for media changing it was possible for the
+ strictrtp code to discard the RTP packets. This change causes
+ strictrtp to enter learning mode once again when the ICE
+ negotiation has completed successfully.
+
+2013-02-28 17:16 +0000 [r382230-382234] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_iax2.c: Prevent deadlock in chan_iax2 when
+ attempting to set caller ID A deadlock can occur in chan_iax2
+ when it attempts to set the caller ID, as it already holds the
+ iax2 private lock and improperly fails to obtain the channel lock
+ before calling ast_set_callerid. By not safely obtaining the
+ channel lock, a locking inversion can take place, causing a
+ deadlock. This patch solves this by calling the required deadlock
+ avoidance functions that obtain the channel lock before setting
+ the caller ID. Thanks to Pavel for fixing my syntax errors and
+ testing this patch out. (closes issue ASTERISK-21128) Reported
+ by: Pavel Troller Tested by: Pavel Troller patches:
+ ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
+ ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller
+ (license 6302) ........ Merged revisions 382233 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_meetme.c, UPGRADE.txt: Let channels joining a MeetMe
+ conference opt out of the denoiser For some channel drivers,
+ specifically those that have a varying rate in the number of
+ audio samples, the audio quality for a MeetMe conference can be
+ exceedingly poor. This is due to a unilateral application of the
+ DENOISE function in func_speex to channels joining the
+ conference. The denoiser function in the speex library is
+ initialized with the number of audio samples in each sample that
+ will be provided to it. If the number of audio samples changes,
+ the denoiser has to be thrown away and re-initialized. While this
+ could be worked around by removing func_speex, that doesn't help
+ if you actually use the denoiser with other channels on the
+ system. This patches does the following: * Checks for the
+ presence of func_speex as opposed to codec_speex when determining
+ if the DENOISE function is present (which is where the function
+ is actually implemented) * Adds an option to MeetMe 'n' that
+ causes the denoiser to not be applied to a channel when it joins.
+ This keeps the current behavior the default, but let's users
+ disable the denoiser if it causes problems on their system.
+ Review: https://reviewboard.asterisk.org/r/2358 (closes issue
+ AST-1062) Reported by: Thomas Arimont ........ Merged revisions
+ 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-27 16:17 +0000 [r382151-382174] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Relax dialog checking in
+ get_sip_pvt_byid_locked so it works when the dialog is forked.
+ (closes issue ASTERISK-20638) Reported by: eelcob Patches:
+ pedantic-call-pickup-from-tag.patch uploaded by eelcob (license
+ 6442) ........ Merged revisions 382171 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configure, include/asterisk/autoconfig.h.in: Regenerate the
+ configure script. The one in the tree was not working for me at
+ all.
+
+2013-02-26 19:45 +0000 [r382111] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * /, configure, configure.ac: Consider linux-gnuspe as linux-gnu *
+ The powerpcspe Linux port uses linux-gnuspe as the OS string. *
+ Our build system shouldn't really care for that, so just call it
+ linux-gnu. * Original report: Roland Stigge ,
+ http://bugs.debian.org/701505 Review:
+ https://reviewboard.asterisk.org/r/2357/ ........ Merged
+ revisions 382110 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-26 19:34 +0000 [r382108] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: Correct RPID parsing for unquoted
+ display-name. Parsing Remote-Party-ID will now succeed if
+ display-name is of the *(token LWS) kind and not just the
+ quoted-string kind. Review:
+ https://reviewboard.asterisk.org/r/2341/ ........ Merged
+ revisions 382107 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-26 19:19 +0000 [r382096] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * /, main/Makefile: Remove unneeded linux-gnueabi* As of r380521
+ the configure scripts converts the value of linux-gnueabi* of
+ OSARCH to "linux-gnu". So no point in testing for those values.
+ ........ Merged revisions 382087 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-26 15:38 +0000 [r382066-382069] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_confbridge.c: Fix typo in r382068 Well, that was
+ embarrassing. Removed an '-l' that somehow got in there.
+
+ * apps/app_confbridge.c: Clean up ConfBridge commands to account
+ for wait_marked users When ConfBridge was refactored to better
+ handle the concept of marked, wait_marked, and normal users
+ co-existing in a conference (thereby implementing a state machine
+ for the conference), the wait_marked users were put into their
+ own list of conference participants, separate from the active
+ users. This list is used for wait_marked users when they are
+ waiting in a conference but no marked user has joined; normal
+ users may have joined at this point however. There are several
+ AMI/CLI commands that affect conference users that were not
+ checking the wait_marked users list: * CLI/AMI commands that
+ mute/unmute a participant. In this case, wait_marked users have
+ to remain in their particular state and should not be affected -
+ however, the commands would return "Channel not found" as opposed
+ to the appropriate error condition. * CLI/AMI commands that kick
+ a participant. An admin should always be able to kick a
+ participant out of the conference. This patch fixes both sets of
+ commands, and cleans up the CLI commands slightly by allowing
+ them to complete a participant name (this was supposed to have
+ been added, but the function call was commented out and wasn't
+ implemented). Review: https://reviewboard.asterisk.org/r/2346/
+ (closes issue AST-1114) Reported by: John Bigelow Tested by: John
+ Bigelow
+
+ * apps/confbridge/conf_config_parser.c,
+ configs/confbridge.conf.sample: Ensure that the default
+ bridge/user profiles are always available ConfBridge and Page
+ require that there always be a default bridge and user profile
+ available. While properties of the default profiles can be
+ overriden in the configuration file, removing them can create
+ situations where neither application can function properly. This
+ patch ensures that if an administrator removes the profiles from
+ the confbridge.conf configuration file, the profiles are added
+ upon load. Documentation clarifying this has been added to the
+ confbridge.conf.sample file. Review:
+ https://reviewboard.asterisk.org/r/2356/ (closes issue AST-1115)
+ Reported by: John Bigelow Tested by: John Bigelow
+
+2013-02-25 12:50 +0000 [r381917-382022] Matthew Jordan <mjordan at digium.com>
+
+ * addons/res_config_mysql.c, /: Clean up use of va_end/va_args in
+ res_config_mysql There were several problems using variadic
+ argument macros in res_config_mysql. * Improper use of va_end.
+ Multiple calls to va_end were possible resulting in an unbalanced
+ matching of va_start/va_end. * Calls to va_arg after a possible
+ encounter of a SENTINEL value. This patch corrects those errors.
+ (closes issue ASTERISK-19451) Reported by: wdoekes patches:
+ ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674)
+ ........ Merged revisions 382021 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_jingle.c, /: Set the sin_family on the bind address
+ socket during initialization Somehow, chan_jingle has managed to
+ operate for years without setting the sin_family on its bindaddr
+ socket. This patch properly sets the field during initial module
+ load to AF_INET. Note that the patch on the issue was modified
+ slightly to change the initialization of the socket from
+ allocation of a chan_jingle private to the module initialization,
+ as the bindaddr object (which is static) only needs to have the
+ address set once. (closes issue ASTERISK-19341) Reported by:
+ andre valentin patches: 0105-chan_jingle.patch uploaded by
+ avalentin (License 6064) ........ Merged revisions 381975 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/manager.c, /: Don't display the AMI ALL class authorization
+ for users if they don't have it When converting AMI class
+ authorizations to a string representation, the method always
+ appends the ALL class authorization. This is especially important
+ for events, as they should always communicate that class
+ authorization - even if the event itself does not specify ALL as
+ a class authorization for itself. (Events have always assumed
+ that the ALL class authorization is implied when they are raised)
+ Unfortunately, this did mean that specifying a user with
+ restricted class authorizations would show up in the 'manager
+ show user' CLI command as having the ALL class authorization.
+ Rather then modifying the existing string manipulation function,
+ this patch adds a function that will only return a string if the
+ field being compared explicitly matches class authorization field
+ it is being compared against. This prevents ALL from being
+ returned unless it is actually specified for the user. (closes
+ issue ASTERISK-20397) Reported by: Johan Wilfer ........ Merged
+ revisions 381939 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_parkandannounce.c, /: Make ParkAndAnnounce return to
+ priority + 1 when return context is not defined The
+ ParkAndAnnounce application documentation for the optional
+ return_context parameter states the following: return_context The
+ goto-style label to jump the call back into after timeout.
+ Default 'priority+1'. Unfortunately, the application was sending
+ the channel back into the dialplan at 'priority', which is the
+ ParkAndAnnounce application call. This causes an infinite loop of
+ the channel constantly being parked, announced, timed out,
+ parked, announced, timed out... while fun, especially for those
+ callers you wish to drive to the end of madness, this was not the
+ intent of the application. (closes issue ASTERISK-20113) Reported
+ by: serginuez patches: app_parkandannounce.diff uploaded by
+ serginuez (License 6405) ........ Merged revisions 381916 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-22 19:38 +0000 [r381893] Michael L. Young <elgueromexicano at gmail.com>
+
+ * res/res_agi.c: Fix FastAGI To Properly Check For A Connection
+ When IPv6 support was added to FastAGI, the intent was to have
+ the ability to check all addresses resolved for a host since we
+ might receive an IPv4 address and an IPv6 address. The problem
+ with the current code, is that, since we are doing O_NONBLOCK, we
+ get EINPROGRESS when calling ast_connect() but are ignoring this
+ instead of handling it. We break out of the loop and continue on.
+ When we later call ast_poll(), it succeeds but we never check if
+ we have a connection or not on the socket level. We then attempt
+ to send data to the host address that we think is setup and it
+ fails. We then check the errno and see that we have "connection
+ refused" and then return with agi failed. This patch does the
+ following: * Handles EINPROGRESS by creating the function
+ handle_connection() - ast_poll() was moved into this function -
+ This function checks the results of the connection on the socket
+ level after calling ast_poll() * Continues to the next address if
+ the above fails to create a connection * Once all addresses
+ resolved are tried and we still are unable to establish a
+ connection, then we return that the FastAGI call failed (closes
+ issue ASTERISK-21065) Reported by: Jeremy Kister Tested by:
+ Jeremy Kister, Michael L. Young Patches:
+ asterisk-21065_poll_correctly_v4.diff Michael L. Young (license
+ 5026) Review: https://reviewboard.asterisk.org/r/2330/
+
+2013-02-22 15:41 +0000 [r381880] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_dial.c: app_dial: Honor the 'c' flag when the calling
+ party hangs up Apparently this feature became broken in 11,
+ probably as a result of the Hangup Cause project. (closes issue
+ ASTERISK-21113) Reprted by: Heiko Wundram Patches: app_dial.patch
+ uploaded by Heiko Wundram (license 5822)
+
+2013-02-21 22:48 +0000 [r381848] Matthew Jordan <mjordan at digium.com>
+
+ * /, configure, configure.ac: Properly detect launchd Asterisk was
+ a little too pro-active in claiming that it found launchd. On
+ systems without launchd - such as FreeBSD - this resulted in
+ certain items in Asterisk that conflict with launchd to not be
+ selectable, such as res_timing_kqueue. (closes issue
+ ASTERISK-20749) Reported by: Oleg Baranov ........ Merged
+ revisions 381847 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-02-20 19:14 +0000 [r381835] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c: Let vm_mailbox_snapshot_create's combine
+ option apply to "Urgent" as well The vm_mailbox_snapshot_create
+ function has an option that combines the contents of INBOX and
+ Old into a single snapshot. The intent of this is that both 'new'
+ messages and 'deleted' messages are given in a single snapshot,
+ as some applications prefer this view of the voicemail world.
+ Unfortunately, the initial implementation ignored the "Urgent"
+ folder. The "Urgent" folder is a pseudo-INBOX, in that new
+ messages left with the 'U' flag will be placed in that folder as
+ opposed to INBOX. Thus, the option failed the intent with which
+ it was added. This patch makes it so that the "Urgent" folder is
+ included in the snapshot when that option is used.
+
+2013-02-19 19:44 +0000 [r381702-381791] kharwell <kharwell at localhost>:
+
+ * /, main/features.c: Write the correct callid to the data1 field
+ in queue_log for transfer events. The incorrect callid was being
+ written to the "data1" field in queue_log table for transfer
+ events. The callid of the queue was being written instead of the
+ transfer target's callid. This now gets the correct "transfer to"
+ number and places that in the "data1" field of the queue_log
+ table when a transfer event is triggered. (closes issue
+ ASTERISK-19960) Reported by: vladimir shmagin ........ Merged
+ revisions 381770 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_confbridge.c: Confbridge channels staying active when
+ all participants leave. If you started/stopped recording of a
+ conference multiple times channels would remain active even when
+ all participants left the conference. This was due to the fact
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