[svn-commits] bebuild: tag 10.9.0-rc1 r373020 - /tags/10.9.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Sep 13 13:27:45 CDT 2012
Author: bebuild
Date: Thu Sep 13 13:27:41 2012
New Revision: 373020
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=373020
Log:
Importing files for 10.9.0-rc1 release.
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tags/10.9.0-rc1/.version (with props)
tags/10.9.0-rc1/ChangeLog (with props)
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+2012-09-13 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.9.0-rc1 Released.
+
+2012-09-12 14:53 +0000 [r372933] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Add channel name to a warning to make
+ debugging easier. The "autodestruct with owner in place" message
+ is typically indicative of a channel reference leak. Printing out
+ the name of the channel in the message may be helpful when trying
+ to debug the issue. ........ Merged revisions 372932 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-11 22:23 +0000 [r372916] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_local.c, /: chan_local: Switch from using a random
+ 4 digit hex identifier to unique id Changes chan_local channels
+ to use an 8 digit hex identifier generated atomically and
+ sequentially in order to eliminate the chance of having multiple
+ channels with the same name during high call volume situations.
+ (issue ASTERISK-20318) Reported by: Dan Cropp Review:
+ https://reviewboard.asterisk.org/r/2104/ ........ Merged
+ revisions 372902 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-11 21:04 +0000 [r372885] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/_private.h, main/message.c, main/asterisk.c: Fix
+ inability to shutdown gracefully due to an unending channel
+ reference. message.c makes use of a special message queue channel
+ that exists in thread storage. This channel never goes away due
+ to the fact that the taskprocessor used by message.c does not get
+ shut down, meaning that it never ends the thread that stores the
+ channel. This patch fixes the problem by shutting down the
+ taskprocessor when Asterisk is shut down. In addition, the thread
+ storage has a destructor that will release the channel reference
+ when the taskprocessor is destroyed. (closes issue AST-937)
+ Reported by Jason Parker Patches: AST-937.patch uploaded by Mark
+ Michelson (License #5049) Tested by Jason Parker
+
+2012-09-11 17:14 +0000 [r372863] dlee <dlee at localhost>:
+
+ * Makefile: Corrects the astsbindir setting when installing the
+ sample asterisk.conf. (closes issue ASTERISK-20406)
+
+2012-09-11 15:30 +0000 [r372841] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/features.c: Fix bad channel application data reference.
+ When channels get bridged due to an AMI bridge action or a DTMF
+ attended transfer, the two channels that get bridged have their
+ application data pointing to the other channel's name. This means
+ that if one channel is hung up but the other moves on, it means
+ that the channel that moves on will have its application data
+ pointing at freed memory. (issue ASTERISK-20335) Reported by:
+ aragon ........ Merged revisions 372840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-10 20:56 +0000 [r372805] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_iax2.c: Ensure iax2 debug output is displayed
+ when expected When IAX2 debug was changed from iax_showframe to
+ iax_outputframe, some instances were missed (or added afterward).
+ This was causing debug output to not be displayed when expected.
+ (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
+ John Covert ........ Merged revisions 372804 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-10 18:41 +0000 [r372767] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Document that 'p' option will
+ continue in dialplan. (closes issue AST-991) Reported by John
+ Bigelow ........ Merged revisions 372765 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-10 18:32 +0000 [r372764] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: Warn on CLI when UDPTL init fails This
+ adds a CLI warning when a SDP offer is rejected due to UDPTL
+ initialization failure. Previously, there was no indication of
+ the reason for offer rejection in this case. (closes issue
+ ASTERISK-20357) Reported-by: Francesco Usseglio Gaudi ........
+ Merged revisions 372763 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-10 17:14 +0000 [r372737] Jonathan Rose <jrose at digium.com>
+
+ * main/channel.c, /: Masquerade: Retain parkinglot settings made by
+ CHANNEL function. Prior to this patch, the user would have a
+ parkinglot set on a channel that was parked and when the channel
+ was retrieved, any attempt by that channel to park would simply
+ use the default. This patch makes parkinglot values set in this
+ way be retained through the masquerade. (closes issue AST-990)
+ Reported by: Nick Huskinson Patches:
+ masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
+ (license 6182) ........ Merged revisions 372736 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-09 01:24 +0000 [r372710] Matthew Jordan <mjordan at digium.com>
+
+ * channels/sip/sdp_crypto.c, /: Only re-create an SRTP session when
+ needed In r356604, SRTP handling was fixed to accomodate multiple
+ crypto keys in an SDP offer and the ability to re-create an SRTP
+ session when the crypto keys changed. In certain circumstances -
+ most notably when a phone is put on hold after having been
+ bridged for a significant amount of time - the act of re-creating
+ the SRTP session causes problems for certain models of phones.
+ The patch committed in r356604 always re-created the SRTP session
+ regardless of whether or not the cryptographic keys changed.
+ Since this is technically not necessary, this patch modifies the
+ behavior to only re-create the SRTP session if Asterisk detects
+ that the remote key has changed. This allows models of phones
+ that do not handle the SRTP session changing to continue to work,
+ while also providing the behavior needed for those phones that do
+ re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported
+ by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
+ https://reviewboard.asterisk.org/r/2099 ........ Merged revisions
+ 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-08 05:21 +0000 [r372695] dlee <dlee at localhost>:
+
+ * /, main/Makefile: Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and
+ tcptls.c. Without this flag, those files will compile with the
+ system installed OpenSSL headers (if they exist). This is a real
+ bummer if a different path was specified using --with-ssl=
+ (closes issue ASTERISK-20392) ........ Merged revisions 372682
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-07 23:06 +0000 [r372621-372656] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup().
+ (closes issue ASTERISK-20349) Reported by: Brent Eagles ........
+ Merged revisions 372655 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, funcs/func_math.c: Remove annoying unconditional debug message
+ from INC/DEC functions. (closes issue AST-1001) Reported by:
+ Guenther Kelleter ........ Merged revisions 372628 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_queue.c: Fix exception path typo in app_queue.c
+ try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy
+ Pepper Patches: fix-local-channel-locking.patch (license #6350)
+ patch uploaded by Jeremy Pepper ........ Merged revisions 372624
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_voicemail.c, /: Fix VoicemailUserEntry event headers
+ ServerEmail and MailCommand reported values. The AMI action
+ VoicemailUsersList VoicemailUserEntry event headers ServerEmail
+ and MailCommand did not report the global values if they were not
+ overridden. The VoicemailUserEntry event header ServerEmail was
+ not populated with the global value if the voicemail user did not
+ override it. The VoicemailUserEntry event header MailCommand was
+ never populated with a value. * Removed unused struct ast_vm_user
+ member mailcmd[]. (closes issue AST-973) Reported by: John
+ Bigelow Tested by: rmudgett ........ Merged revisions 372620 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-07 02:25 +0000 [r372555-372582] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/app_minivm.c: Free ast_str objects when temp file fails
+ to be created in MiniVM The previous commit (r372554) was from a
+ patch that was written before r366880, which ensured that ast_str
+ objects allocated in the sendmail routine were free'd in off
+ nominal paths. This commit frees the string objects in the off
+ nominal path introduced in r372554. (issue ASTERISK-17133)
+ Reported by: Tzafrir Cohen ........ Merged revisions 372581 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_minivm.c: Fix file descriptor leak and pointer scope
+ issue in MiniVM when sending mail When MiniVM sends an e-mail and
+ it has the volgain option set, it will spawn sox in a separate
+ process to handle the manipulation of the sound file. In doing
+ so, it creates a temporary file. There are two problems here: 1)
+ The file descriptor returned from mkstemp is leaked 2) The
+ finalfilename character pointer points to a buffer that loses
+ scope once volgain processing is finished. Note that in r316265,
+ Russell fixed some gcc warnings by using the return value of the
+ mkstemp call. A warning was placed in minivm that the file
+ descriptor was going to be leaked. This patch reverts that
+ change, as it handles the leak and 'uses' the file descriptor
+ returned from mkstemp. (closes issue ASTERISK-17133) Reported by:
+ Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir
+ Cohen (license #5035) ........ Merged revisions 372554 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-06 22:10 +0000 [r372522] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: Fix loss of MOH on an ISDN channel when
+ parking a call for the second time. Using the AMI redirect action
+ to take an ISDN call out of a parking lot causes the MOH state to
+ get confused. The redirect action does not take the call off of
+ hold. When the call is subsequently parked again, the call no
+ longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on
+ repeated AST_CONTROL_HOLD frames if it is already in a state
+ where it is supposed to be sending MOH. The MOH may have been
+ stopped by other means. (Such as killing the generator.) This
+ simple fix is done rather than making the AMI redirect action
+ post an AST_CONTROL_UNHOLD unconditionally when it redirects a
+ channel and thus potentially breaking something with an
+ unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches:
+ jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by
+ rmudgett ........ Merged revisions 372521 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+
+2012-09-06 21:40 +0000 [r372518] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_queue.c: Ensure listed queues are not offered for
+ completion When using tab-completion for the list of queues on
+ "queue reset stats" or "queue reload
+ {all|members|parameters|rules}", the tab-completion listing for
+ further queues erroneously listed queues that had already been
+ added to the list. The tab-completion listing now only displays
+ queues that are not already in the list. (closes issue AST-963)
+ Reported-by: John Bigelow ........ Merged revisions 372517 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-06 18:54 +0000 [r372499] dsessions <dsessions at localhost>:
+
+ * channels/chan_sip.c, configs/res_ldap.conf.sample: LDAP Realtime
+ Peers Cannot Register Prior to 1.8, it was not necessary for an
+ explicit "type" to be set for an asterisk LDAP realtime peer. Now
+ the routine find_peer actually checks the type field during
+ registration and fails to find the peer if it is not set. The
+ attached patches make the realtime type equal whatever type is
+ being searched for if the type is 0 upon return from routine
+ build_peer. (closes issue ASTERISK-17222) Reported by: John
+ Covert Patch by: David Vossel Tested by: Darren Sessions Review:
+ https://reviewboard.asterisk.org/r/2095/
+
+2012-09-06 15:54 +0000 [r372472] Jonathan Rose <jrose at digium.com>
+
+ * /, UPGRADE-1.8.txt: chan_sip: Note change in behavior to how
+ directmediapermit/deny ACL works r366547 introduced a change to
+ the directmedia ACL for chan_sip which modified the behavior
+ significantly. Prior to the patch, this option would bridge peers
+ with directmedia if a peer's IP address matched its own
+ directmedia ACL. After that patch, the peer would check the
+ bridged peer's ACL instead. This change has been present since
+ 1.8.14.0. That patched failed to document the change in
+ Upgrade.txt, so this patch adds mention of that change to
+ UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876)
+ ........ Merged revisions 372471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-06 14:29 +0000 [r372445] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_queue.c: Ensure "rules" is tab-completable for "queue
+ show" Previously, tabbing at the end of "queue show" produced a
+ list of available queues about which information could be shown,
+ but did not include an alternative command, "rules", to access
+ information about queue rules. The "rules" item should now be
+ shown in the list of tab-completable items. (closes issue
+ AST-958) Reported-by: John Bigelow ........ Merged revisions
+ 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-06 02:49 +0000 [r372391-372418] Matthew Jordan <mjordan at digium.com>
+
+ * /, pbx/pbx_dundi.c: Fix DUNDi message routing bug when
+ neighboring peer is unreachable Consider a scenario where DUNDi
+ peer PBX1 has two peers that are its neighbors, PBX2 and PBX3,
+ and where PBX2 and PBX3 are also neighbors. If the connection is
+ temporarily broken between PBX1 and PBX3, PBX1 should not include
+ PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER
+ message, as it cannot send messages to PBX3. If it does, PBX2
+ will assume that PBX3 already received the message and fail to
+ forward the message on to PBX3 itself. This patch fixes this by
+ only including peers in a DPDISCOVER message that are reachable
+ by the sending node. This includes all peers with an empty
+ address (00:00:00:00:00:00) and that are have been reached by a
+ qualify message. This patch also prevents attempting to qualify a
+ dynamic peer with an empty address until that peer registers.
+ (closes issue ASTERISK-19309) Reported by: Peter Racz patches:
+ dundi_routing.patch uploaded by Peter Racz (license 6290) The
+ patch uploaded by Peter was modified slightly for this commit.
+ ........ Merged revisions 372417 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_followme.c: Allow configured numbers for FollowMe to
+ be greater than 90 characters When parsing a 'number' defined in
+ followme.conf, FollowMe previously parsed the number in the
+ configuration file into a buffer with a length of 90 characters.
+ This can artificially limit some parallel dial scenarios. This
+ patch allows for numbers of any length to be defined in the
+ configuration file. Note that Clod Patry originally wrote a patch
+ to fix this problem and received a Ship It! on the JIRA issue.
+ The patch originally expanded the buffer to 256 characters.
+ Instead, the patch being committed duplicates the string in the
+ config file on the stack before parsing it for consumption by the
+ application. (closes issue ASTERISK-16879) Reported by: Clod
+ Patry Tested by: mjordan patches: followme_no_limit.diff uploaded
+ by Clod Patry (license #5138) Slightly modified for this commit.
+ ........ Merged revisions 372390 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-05 19:42 +0000 [r372372] Richard Mudgett <rmudgett at digium.com>
+
+ * main/dsp.c: Fix compile error.
+
+2012-09-05 19:22 +0000 [r372358] Kinsey Moore <kmoore at digium.com>
+
+ * main/manager.c, /: Correct documentation for ModuleLoad AMI
+ action The documentation incorrectly listed 'rtp' as a reloadable
+ subsystem and left out many other reloadable subsystems. It is
+ now also documented that subsystems may only be reloaded, not
+ loaded or unloaded. (closes issue AST-977) Reported-by: John
+ Bigelow ........ Merged revisions 372354 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-05 18:43 +0000 [r372341] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/dsp.c, /: dsp.c: in ast_mf_detect_init incorrectly sets
+ goertzel samples to 160, should be MF_GSIZE Related
+ https://reviewboard.asterisk.org/r/2097/ ........ Merged
+ revisions 372339 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-05 18:30 +0000 [r372338] Kinsey Moore <kmoore at digium.com>
+
+ * main/pbx.c, /: Ensure counts generated in
+ manager_show_dialplan_helper are correct When
+ manager_show_dialplan_helper was written, the counter increment
+ for the total number of contexts was placed with the extensions
+ increment instead of in the enclosing loop. This function should
+ now generate correct context counts. (closes issue AST-970)
+ Reported-by: John Bigelow ........ Merged revisions 372337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-05 13:42 +0000 [r372288] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c, /: Fix memory leaks in app_voicemail when
+ using IMAP storage or realtime config This patch fixes two memory
+ leaks: 1. When find_user is called with NULL as its first
+ parameter, the voicemail user returned is allocated on the heap.
+ The inboxcount2 function uses find_user in such a fashion when
+ counting new messages, and fails to free the resulting voicemail
+ user object. 2. When populate_defaults is called on a voicemail
+ user, it wipes whatever flags have been set on the object by
+ copying over the global flags object. If the VM_ALLOCED flag was
+ ste on the voicemail user prior to doing so, that flag is
+ removed. This leaks the voicemail user when free_user is later
+ called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
+ patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
+ Patch slightly modified for this commit. Review:
+ https://reviewboard.asterisk.org/r/2096 ........ Merged revisions
+ 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-05 07:37 +0000 [r372213-372240] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/dsp.c, /: dsp.c: Fix multiple issues when no-interdigit
+ delay is present, and fast DTMF 50ms/50ms Revert DTMF hit/miss
+ detector to original -r349249 method with some changes, remove
+ unnecessary; 1. reseting of hits=0, when no signal, only need to
+ set it once. 2. incrementing of hits, when the hit is the same as
+ the current hit. 3. setting of lasthit, when it's the same as
+ before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3
+ spelling mistakes (closes issue ASTERISK-19610) alecdavis
+ (license 585) Reported by: Jean-Philippe Lord Tested by:
+ alecdavis Review: https://reviewboard.asterisk.org/r/2085/
+ ........ Merged revisions 372239 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/dsp.c, /: dsp.c: optimize goerztzel sample loops, in
+ dtmf_detect, mf_detect and tone_detect use a temporary short int
+ when repeatedly used to call goertzel_sample. alecdavis (license
+ 585) Reported by: alecdavis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/2093/ ........ Merged
+ revisions 372212 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-05 04:47 +0000 [r372198] Michael L. Young <elgueromexicano at gmail.com>
+
+ * res/res_rtp_asterisk.c, /: Fix Incrementing Sequence Number For
+ Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was put in
+ place to increment the sequence number for retransmitted DTMF end
+ packets. With the introduction of the RTP engine API in 1.8, the
+ sequence number was no longer being incremented. This patch fixes
+ this regression as well as cleans up a few lines that were not
+ doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh
+ Bansal Tested by: Michael L. Young Patches:
+ 01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license
+ 6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2083/ ........ Merged
+ revisions 372185 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-05 02:19 +0000 [r372165] Matthew Jordan <mjordan at digium.com>
+
+ * cel/cel_pgsql.c, /: Fix memory leak when CEL is successfully
+ written to PostgreSQL database PQClear is not called when the
+ result object of a call to PQExec has a status of
+ PGRES_COMMAND_OK. Interestingly enough, the off nominal case was
+ handled properly, so this memory leak only occurred when CEL
+ records were successfully written. This patch properly clears the
+ result in the nominal code path. (closes issue ASTERISK-19991)
+ Reported by: Etienne Lessard Tested by: Etienne Lessard patches:
+ mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license
+ #6394) ........ Merged revisions 372158 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-30 20:53 +0000 [r372049-372090] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_queue.c: Prevent crash on shutdown due to refcount
+ error on queues container. When app_queue is unloaded, the queues
+ container has its refcount decremented, potentially to 0. Then
+ the taskprocessor responsible for handling device state changes
+ is unreferenced. If the taskprocessor happens to be just about to
+ run its task, then it will create and destroy an iterator on the
+ queues container. This can cause the refcount on the queues
+ container to increase to 1 and then back to 0. Going back to 0 a
+ second time results in double frees. This failure was seen
+ periodically in the testsuite when Asterisk would shut down.
+ ........ Merged revisions 372089 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_queue.c: Help prevent ringing queue members from
+ being rung when ringinuse set to no. Queue member status would
+ not always get updated properly when the member was called, thus
+ resulting in the member getting multiple calls. With this change,
+ we update the member's status at the time of calling, and we also
+ check to make sure the member is still available to take the call
+ before placing an outbound call. (closes issue ASTERISK-16115)
+ reported by nik600 Patches: app_queue.c-svn-r370418.patch
+ uploaded by Italo Rossi (license #6409) ........ Merged revisions
+ 372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-30 16:22 +0000 [r371962-372020] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_iax2.c: AST-2012-013: Resolve ACL rules being
+ ignored during calls by some IAX2 peers When an IAX2 call is made
+ using the credentials of a peer defined in a dynamic Asterisk
+ Realtime Architecture (ARA) backend, the ACL rules for that peer
+ are not applied to the call attempt. This allows for a remote
+ attacker who is aware of a peer's credentials to bypass the ACL
+ rules set for that peer. This patch ensures that the ACLs are
+ applied for all peers, regardless of their storage mechanism.
+ (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by:
+ mjordan, Alan Frisch ........ Merged revisions 372015 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/manager.c, /, README-SERIOUSLY.bestpractices.txt:
+ AST-2012-012: Resolve AMI User Unauthorized Shell Access through
+ ExternalIVR The AMI Originate action can allow a remote user to
+ specify information that can be used to execute shell commands on
+ the system hosting Asterisk. This can result in an unwanted
+ escalation of permissions, as the Originate action, which
+ requires the "originate" class authorization, can be used to
+ perform actions that would typically require the "system" class
+ authorization. Previous attempts to prevent this permission
+ escalation (AST-2011-006, AST-2012-004) have sought to do so by
+ inspecting the names of applications and functions passed in with
+ the Originate action and, if those applications/functions matched
+ a predefined set of values, rejecting the command if the user
+ lacked the "system" class authorization. As noted by IBM X-Force
+ Research, the "ExternalIVR" application is not listed in the
+ predefined set of values. The solution for this particular
+ vulnerability is to include the "ExternalIVR" application in the
+ set of defined applications/functions that require "system" class
+ authorization. Unfortunately, the approach of inspecting fields
+ in the Originate action against known applications/functions has
+ a significant flaw. The predefined set of values can be bypassed
+ by creative use of the Originate action or by certain dialplan
+ configurations, which is beyond the ability of Asterisk to
+ analyze at run-time. Attempting to work around these scenarios
+ would result in severely restricting the applications or
+ functions and prevent their usage for legitimate means. As such,
+ any additional security vulnerabilities, where an
+ application/function that would normally require the "system"
+ class authorization can be executed by users with the "originate"
+ class authorization, will not be addressed. Instead, the
+ README-SERIOUSLY.bestpractices.txt file has been updated to
+ reflect that the AMI Originate action can result in commands
+ requiring the "system" class authorization to be executed. Proper
+ system configuration can limit the impact of such scenarios.
+ (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM
+ X-Force Research ........ Merged revisions 371998 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * doc/CODING-GUIDELINES (added), /: Restore CODING-GUIDELINES to
+ doc folder In r294740, the CODING-GUIDELINES was removed from the
+ doc folder in favor of the content on the Asterisk wiki. Some
+ folks still look in the doc folder initially for coding guideline
+ suggestions; as such, this patch adds a CODING-GUIDELINES file
+ back into the doc folder. The content of the file merely points
+ to the correct page on the Asterisk wiki where the coding
+ guidelines currently live. (closes issue ASTERISK-20279) Reported
+ by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by
+ Andrew Latham (license 5985) ........ Merged revisions 371961
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-29 20:58 +0000 [r371920] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Adding test events for
+ following activity in MeetMe. ........ Merged revisions 371919
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-29 19:40 +0000 [r371861-371890] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, /: Initialize file descriptors for dummy channels
+ to -1. Dummy channels usually aren't read from, but functions
+ like SHELL and CURL use autoservice on the channel. (closes issue
+ ASTERISK-20283) Reported by: Gareth Palmer Patches:
+ svn-371580.patch (license #5169) patch uploaded by Gareth Palmer
+ (modified) ........ Merged revisions 371888 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_dial.c, /: Fix hangup cause passthrough regression. The
+ v1.8 -r369258 change to fix the F and F(x) action logic
+ introduced a regression in passing the hangup cause from the
+ called channel to the caller channel. (closes issue
+ ASTERISK-20287) Reported by: Konstantin Suvorov Patches:
+ app_dial_hangupcause.patch (license #6421) patch uploaded by
+ Konstantin Suvorov (modified) Tested by: rmudgett ........ Merged
+ revisions 371860 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-29 17:07 +0000 [r371825] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Send 408 on retransmit timeout
+ instead of 603 (closes issue ASTERISK-20124) Reported by: Walter
+ Doekes ........ Merged revisions 371824 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-27 21:49 +0000 [r371748-371789] Mark Michelson <mmichelson at digium.com>
+
+ * configs/agents.conf.sample, /: Fix misleading documentation in
+ agents.conf.sample regarding ackcall usage. The documentation
+ made it sound as if the DTMF acknowledgment was needed at the
+ time the agent logs in, rather than when the agent is called.
+ This is likely a relic from the days when there were multiple
+ ways of logging in agents. (closes issue AST-962) reported by
+ Steve Pitts ........ Merged revisions 371787 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/manager.c, /: Fix incorrect documentation of the
+ MailboxStatus manager command. The "Waiting" field was
+ misdocumented as reporting the number of messages waiting. In
+ reality, it simply indicated the presence or absence of waiting
+ messages. ........ Merged revisions 371782 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configs/queues.conf.sample: Fix incorrectly documented option
+ in queues.conf sharedlastcall defaults to "no" not "yes" (closes
+ issue AST-979) reported by Steve Pitts ........ Merged revisions
+ 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-27 16:43 +0000 [r371719] dlee <dlee at localhost>:
+
+ * main/lock.c, /: Fixes ast_rwlock_timed[rd|wr]lock for BSD and
+ variants. The original implementations simply wrap pthread
+ functions, which take absolute time as an argument. The spinlock
+ version for systems without those functions treated the argument
+ as a delta. This patch fixes the spinlock version to be
+ consistent with the pthread version. (closes issue
+ ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch
+ uploaded by Egor Gorlin (license 6416) ........ Merged revisions
+ 371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-27 13:57 +0000 [r371691] Kinsey Moore <kmoore at digium.com>
+
+ * /, main/utils.c: Implement workaround for BETTER_BACKTRACES crash
+ When compiling with BETTER_BACKTRACES enabled, Asterisk will
+ sometimes crash when "core show locks" is run. This happens
+ regularly in the testsuite since several tests run "core show
+ locks" to help with debugging. This seems to be a fault with
+ libraries on certain operating systems (notably CentOS 6.2/6.3)
+ running on virtual machines and utilizing gcc 4.4.6. (closes
+ issue ASTERISK-20090) ........ Merged revisions 371690 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-26 23:06 +0000 [r371663] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/dsp.c, /: mf_detect: incorrectly used DTMF_GSIZE instead of
+ MF_GSIZE ........ Merged revisions 371662 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-21 20:40 +0000 [r371591] Mark Michelson <mmichelson at digium.com>
+
+ * cdr/cdr_tds.c, main/xmldoc.c, apps/app_dial.c,
+ channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c,
+ main/file.c, main/utils.c, apps/app_queue.c, pbx/pbx_config.c,
+ res/res_jabber.c, apps/app_stack.c, channels/chan_oss.c,
+ res/res_config_sqlite.c: Fix misuses of asprintf throughout the
+ code. This fixes three main issues * Change asprintf() uses to
+ ast_asprintf() so that it pairs properly with ast_free() and no
+ longer causes MALLOC_DEBUG to freak out. * When ast_asprintf()
+ fails, set the pointer NULL if it will be referenced later. * Fix
+ some memory leaks that were spotted while taking care of the
+ first two points. (Closes issue ASTERISK-20135) reported by
+ Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071
+ ........ Merged revisions 371590 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-20 15:27 +0000 [r371545] Kinsey Moore <kmoore at digium.com>
+
+ * main/udptl.c, /: Ignore recovered zero-length secondary UDPTL
+ packets In some cases, recovering lost packets using the
+ secondary packet recovery mechanism with UDPTL/T.38 can result in
+ the recovery of zero-length packets. These must be ignored or the
+ frame generated from them can cause segfaults and allocation
+ failures. (closes issue ASTERISK-19762) (closes issue
+ ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob
+ Gagnon (rgagnon) ........ Merged revisions 371544 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-18 02:34 +0000 [r371491-371529] Matthew Jordan <mjordan at digium.com>
+
+ * main/http.c: Remove old debug code from http configuration
+ loading (closes issue ASTERISK-20254) Reported by: Andrew Latham
+ Patches: http.diff uploaded by Andrew Latham (license #5985)
+
+ * main/xmldoc.c, /: Fix memory leak in XML documentation When
+ formatting documentation fields, the XML documentation parser
+ calls xmldoc_get_formatted. This function allocates a string
+ buffer at the beginning of its routine. Unfortunately, on certain
+ code paths, it also calls xmldoc_string_cleanup, which assumes
+ that it will create the string buffer. The previously allocated
+ string buffer is then leaked by the xmldoc_string_cleanup
+ routine. Now: we don't do that. (closes issue AST-932) Reported
+ by: Alexander Homig ........ Merged revisions 371469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-17 15:51 +0000 [r371437] Kinsey Moore <kmoore at digium.com>
+
+ * main/loader.c, /: Add instrumentation to subsystem reloads When
+ Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
+ generate TestEvent AMI events on subsystem reloads such as cdr,
+ dnsmgr, extconfig, etc. (issue PQ-1126) ........ Merged revisions
+ 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-16 22:50 +0000 [r371398] Terry Wilson <twilson at digium.com>
+
+ * /, main/config.c: Handle integer over/under-flow in
+ ast_parse_args The strtol family of functions will return
+ *_MIN/*_MAX on overflow. To detect when an overflow has happened,
+ errno must be set to 0 before calling the function, then checked
+ afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged
+ revisions 371392 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-16 22:42 +0000 [r371394] Kinsey Moore <kmoore at digium.com>
+
+ * main/loader.c, /: Add module reload instrumentation for
+ TEST_FRAMEWORK This adds AMI events for module reloads when
+ Asterisk is built with TEST_FRAMEWORK enabled and corrects
+ generation of the module load AMI event. (issue PQ-1126) ........
+ Merged revisions 371393 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-16 19:05 +0000 [r371338-371358] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Use pvt outgoing_call variable
+ to set Remote-Party-ID Header Previously the pvt SIP_OUTGOING
+ flag was used instead, which will frequently flip during
+ reinvites. (closes issue AST-897) Reported by: Thomas Arimont
+ ........ Merged revisions 371357 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: chan_sip: Trigger reinvite if the SDP
+ answer is included in the SIP ACK Under certain conditions, a SIP
+ transaction involving directmedia wouldn't trigger a re-invite
+ because the SDP answer was included in an ACK instead of in a
+ message that we would have triggered the invite with. This patch
+ just queues a source change control frame if the dialog is using
+ directmedia when we find sdp for an ACK. (closes issue AST-913)
+ Reported by: Thomas Arimont ........ Merged revisions 371337 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-15 23:19 +0000 [r371313] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_queue.c: Fix bug where final queue member would not
+ be removed from memory. If a static queue had realtime members,
+ then there could be a potential for those realtime members not to
+ be properly deleted from memory. If the queue's members were
+ loaded from realtime and then all the members were deleted from
+ the backend, then the queue would still think these members
+ existed. The reason was that there was a short- circuit in code
+ such that if there were no members found in the backend, then the
+ queue would not be updated to reflect this. Note that this only
+ affected static queues with realtime members. Realtime queues
+ with realtime members were unaffected by this issue. (closes
+ issue ASTERISK-19793) reported by Marcus Haas ........ Merged
+ revisions 371306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-15 20:15 +0000 [r371271] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: Avoid unconditional NULLing of mwipvt on
+ relatedpeer on SIP dialog destruction The other instance of this
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