[svn-commits] bebuild: tag 1.8.17.0-rc1 r373016 - /tags/1.8.17.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Sep 13 13:09:51 CDT 2012
Author: bebuild
Date: Thu Sep 13 13:09:46 2012
New Revision: 373016
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=373016
Log:
Importing files for 1.8.17.0-rc1 release.
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+2012-09-13 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.17.0-rc1 Released.
+
+2012-09-12 15:42 +0000 [r372959] Matthew Jordan <mjordan at digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Constify
+ __ao2_ref_debug in astobj2 When REF_DEBUG is enabled in certain
+ files - most notably ccss.c - the 'tag' parameter passed to
+ __ao2_ref_debug will be a const char *. The function currently
+ expects that parameter to not be const. This causes a warning
+ when compiling, as the const qualifier is being discarded. With
+ dev-mode enabled, this prevents compiling Asterisk. This patch
+ makes __ao2_ref_debug's tag and file parameters const. (closes
+ issue ASTERISK-20408) Reported by: mjordan
+
+2012-09-12 14:51 +0000 [r372932] 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.
+
+2012-09-11 22:11 +0000 [r372902] 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/
+
+2012-09-11 15:26 +0000 [r372840] 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
+
+2012-09-10 20:53 +0000 [r372804] 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
+
+2012-09-10 18:35 +0000 [r372765] 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
+
+2012-09-10 18:31 +0000 [r372763] 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
+
+2012-09-10 17:07 +0000 [r372736] 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)
+
+2012-09-09 01:19 +0000 [r372709] Matthew Jordan <mjordan at digium.com>
+
+ * channels/sip/sdp_crypto.c: Only re-create an SRTP session when
+ needed; respond with correct crypto policy 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. In addition, in Asterisk 1.8
+ only, it was found that phones that offer AES_CM_128_HMAC_SHA1_32
+ will end up with no audio if the phone is the initiator of the
+ call. The phone will send an INVITE request specifying that
+ AES_CM_128_HMAC_SHA1_32 be used for the cryptographic policy;
+ Asterisk will set its policy to that value. Unfortunately, when
+ the call is Answered and a 200 OK is sent back to the UA, the
+ policy sent in the response's SDP will be the hard coded value
+ AES_CM_128_HMAC_SHA1_80. This potentially results in Asterisk
+ using the INVITE request's policy of AES_CM_128_HMAC_SHA1_32,
+ while the phone uses Asterisk's response of
+ AES_CM_128_HMAC_SHA1_80. Hilarity ensues as both endpoints think
+ the other is crazy. This patch fixes that by caching the policy
+ from the request and responding with it. Note that this is not a
+ problem in Asterisk 10 and later, as the ability to configure the
+ policy was added in that version. (issue ASTERISK-20194) Reported
+ by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
+ https://reviewboard.asterisk.org/r/2099
+
+2012-09-08 03:54 +0000 [r372682] 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)
+
+2012-09-07 23:05 +0000 [r372620-372655] Richard Mudgett <rmudgett at digium.com>
+
+ * main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup(). (closes
+ issue ASTERISK-20349) Reported by: Brent Eagles
+
+ * funcs/func_math.c: Remove annoying unconditional debug message
+ from INC/DEC functions. (closes issue AST-1001) Reported by:
+ Guenther Kelleter
+
+ * 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
+
+ * 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
+
+2012-09-07 02:24 +0000 [r372554-372581] 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
+
+ * 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)
+
+2012-09-06 21:38 +0000 [r372517] 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
+
+2012-09-06 18:54 +0000 [r372498] dsessions <dsessions at localhost>:
+
+ * configs/res_ldap.conf.sample, channels/chan_sip.c: 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 15:52 +0000 [r372471] Jonathan Rose <jrose at digium.com>
+
+ * UPGRADE.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)
+
+2012-09-06 14:28 +0000 [r372444] 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
+
+2012-09-06 02:48 +0000 [r372390-372417] 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.
+
+ * 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.
+
+2012-09-05 19:20 +0000 [r372354] 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
+
+2012-09-05 18:34 +0000 [r372339] 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/
+
+2012-09-05 18:29 +0000 [r372337] 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
+
+2012-09-05 13:13 +0000 [r372268] 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
+
+2012-09-05 07:35 +0000 [r372212-372239] 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/
+
+ * 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/
+
+2012-09-05 03:45 +0000 [r372185] 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/
+
+2012-09-05 02:16 +0000 [r372158] 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)
+
+2012-08-30 20:51 +0000 [r372048-372089] 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.
+
+ * 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)
+
+2012-08-30 16:21 +0000 [r371961-372015] 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
+
+ * 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
+
+ * 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)
+
+2012-08-29 20:42 +0000 [r371919] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_meetme.c: app_meetme: Adding test events for following
+ activity in MeetMe.
+
+2012-08-29 19:38 +0000 [r371860-371888] 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)
+
+ * 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
+
+2012-08-29 16:59 +0000 [r371824] 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
+
+2012-08-27 21:47 +0000 [r371747-371787] 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
+
+ * 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. (closes
+ issue AST-975) reported by John Bigelow
+
+ * 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
+
+2012-08-27 16:40 +0000 [r371718] 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)
+
+2012-08-27 13:43 +0000 [r371690] 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)
+
+2012-08-26 23:03 +0000 [r371662] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/dsp.c: mf_detect: incorrectly used DTMF_GSIZE instead of
+ MF_GSIZE
+
+2012-08-21 20:35 +0000 [r371590] Mark Michelson <mmichelson at digium.com>
+
+ * 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, 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: 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
+
+2012-08-20 15:25 +0000 [r371544] 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)
+
+2012-08-17 18:51 +0000 [r371469] Matthew Jordan <mjordan at digium.com>
+
+ * 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
+
+2012-08-17 15:49 +0000 [r371393-371436] 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)
+
+ * 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)
+
+2012-08-16 22:30 +0000 [r371392] 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/
+
+2012-08-16 18:57 +0000 [r371337-371357] 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
+
+ * 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
+
+2012-08-15 23:10 +0000 [r371306] 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
+
+2012-08-15 20:14 +0000 [r371270] 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
+ bug was fixed by jcolp/file in r121496. If we are destroying a
+ dialog only set the MWI dialog pointer on the related peer to
+ NULL if it is the dialog currently being destroyed. (closes issue
+ ASTERISK-20119) Patch-by: Misha Vodsedalek
+
+2012-08-13 20:00 +0000 [r371201] Kinsey Moore <kmoore at digium.com>
+
+ * main/loader.c, apps/app_meetme.c: Add test instrumentation This
+ adds test instrumentation for loading and unloading of modules
+ and for certain actions in MeetMe to be used in the testsuite or
+ any other consumer of AMI events. These will only be generated
+ when Asterisk is built with TEST_FRAMEWORK enabled. (issue
+ PQ-1131) (issue PQ-1133)
+
+2012-08-13 19:49 +0000 [r371198] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix problem where incorrect pointer was
+ checked for nullity.
+
+2012-08-10 21:21 +0000 [r371141] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Fix a couple of documentation problems in
+ app_queue.c * The RemoveQueueMember app made mention of options
+ that could be passed in, but no options are supported. I have
+ removed the listing of options from the documentation. * The
+ RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value
+ that could be set. (closes issue AST-949) reported by Steve Pitts
+ (closes issue AST-954) reported by Steve Pitts
+
+2012-08-10 16:40 +0000 [r371060-371089] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c: remove ALREADYGONE flag on ooh323 call data
+ by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone
+ there really. This indication arrive from asterisk core not h.323
+ stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov
+ Patches: ASTERISK-19308.patch
+
+ * addons/ooh323c/src/ooGkClient.c: Send re-register packets by GRQ
+ (gatekeeper request) interval (close issue ASTERISK-20094)
+ Patches: ASTERISK-20094-2.patch
+
+2012-08-09 18:58 +0000 [r371012] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, channels/sig_ss7.c, channels/chan_dahdi.c,
+ configure, include/asterisk/autoconfig.h.in, configure.ac: Use
+ better libss7 detection test and move libpri compile test.
+
+2012-08-09 18:58 +0000 [r370988-371011] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c: Fix to resend GRQ/RRQ if RRJ
+ (registration reject) is received (close issue ASTERISK-20094)
+ Patches: ASTERISK-20094.patch
+
+ * addons/ooh323c/src/ooh323ep.c: change opening h323 logfile with
+ append mode instead of overwrite
+
+2012-08-09 17:39 +0000 [r370985] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_meetme.c: Correct documentation for the MeetMe x flag
+ The documentation for the x flag for MeetMe incorrectly described
+ its function as closing down the conference when the last marked
+ user left. It actually causes the users with that flag to leave
+ the conference when the last marked user exits. The functionality
+ of this flag is not changing.
+
+2012-08-08 22:40 +0000 [r370952] Michael L. Young <elgueromexicano at gmail.com>
+
+ * apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When a
+ channel hangs up while being spied upon and the option to exit
+ the ChanSpy application when the spied on channel hangs up is
+ set, ast_autochan_destroy is not being called and therefore a
+ reference to the spied upon channel is not removed. The symptom
+ being reported was that when using func_group in the dialplan and
+ calling "group show channels" at the cli, the spied upon channel
+ was still being shown while "core show channels" showed that the
+ channel was not up. This patch calls ast_autochan_destroy when a
+ spied upon channel hangs up and the option to exit the ChanSpy
+ application is set, removing the reference to the channel
+ allowing the count for the group that the spied channel was part
+ of to be decremented. (closes issue ASTERISK-17515) Reported by:
+ Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young
+ Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael
+ L. Young (license 5026)
+
+2012-08-08 20:28 +0000 [r370923] Kinsey Moore <kmoore at digium.com>
+
+ * main/channel.c: Do not define a cause that doesn't actually exist
+ AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no
+ cause information. As such, it should not be defined and
+ translatable as a cause.
+
+2012-08-08 19:58 +0000 [r370900] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/sig_analog.h: Fix the analog dial *0 flash-hook of
+ bridged peer feature. The flash-hook the bridged peer feature now
+ correctly determines if the bridged peer is another chan_dahdi
+ channel, that it is an analog channel, and that it has the
+ correct signaling for an FXO port. It now also flash-hooks the
+ correct channel.
+
+2012-08-07 19:19 +0000 [r370856] Kinsey Moore <kmoore at digium.com>
+
+ * main/channel.c: Add missing AST_CAUSE_* -> text translations
+
+2012-08-06 15:00 +0000 [r370797] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Improve debug message for temporary outbound
+ proxies. Thanks to Paul Belanger for pointing this out.
+
+2012-08-03 21:43 +0000 [r370769-370771] Mark Michelson <mmichelson at digium.com>
+
+ * channels/sip/config_parser.c: Seriously? Another compilation
+ error fixed. Somebody beat me.
+
+ * channels/chan_sip.c: Remove unused variable.
+
+ * channels/sip/config_parser.c, channels/sip/include/sip.h,
+ channels/chan_sip.c: Fix error in the "IPorHost" section of a SIP
+ dialstring. This is based on the review request posted by Walter
+ Doekes (referenced lower in the commit message) The main fix here
[... 41135 lines stripped ...]
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