[asterisk-commits] bebuild: tag certified-1.8.15-cert1-rc1 r375690 - /certified/tags/1.8.15-cert...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Nov 2 17:05:17 CDT 2012
Author: bebuild
Date: Fri Nov 2 17:05:13 2012
New Revision: 375690
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=375690
Log:
Importing files for 1.8.15-cert1-rc1 release.
Added:
certified/tags/1.8.15-cert1-rc1/.lastclean (with props)
certified/tags/1.8.15-cert1-rc1/.version (with props)
certified/tags/1.8.15-cert1-rc1/ChangeLog (with props)
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URL: http://svnview.digium.com/svn/asterisk/certified/tags/1.8.15-cert1-rc1/.lastclean?view=auto&rev=375690
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+2012-11-02 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.15-cert1-rc1 Released.
+
+2012-11-02 18:59 +0000 [r375630-375631] Richard Mudgett <rmudgett at digium.com>
+
+ * main/cel.c, /: Fix compiler warnings. gcc (GCC) 4.2.4 has
+ problems casting away constness. ........ Merged revisions 370275
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Multiple
+ revisions 375519-375524 ........ r375519 | rmudgett | 2012-10-30
+ 16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines chan_misdn: Timer
+ primitives must be handled first. The frm->addr is a different
+ "address space" than the stack/instance address of other Lx
+ primitives. The test for B channel instance address could fail.
+ Patches: patch01_timers.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2888 ........ r375520 | rmudgett |
+ 2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines
+ chan_misdn: Free memory in error paths and other memory leaks.
+ The one line commented with BUG is not easily fixable because
+ there is no de-init function one can call. Patches:
+ patch02_memory.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2888 ........ r375521 | rmudgett | 2012-10-30
+ 16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines chan_misdn: ISDN NT
+ L2 de-establish/establish * An NT-PTMP cannot de/establish L2
+ since it doesn't know the TEIs. * On NT-PTP L2 is started when L1
+ is finally active in handle_l1. * L2 deactivation logging
+ cleanup. * L2 aggregate link status is unknown for NT-PTMP, show
+ as "UNKN". * Removed unused functions and code for L2 handling.
+ Patches: patch03_L2estab.diff (license #6372) patch uploaded by
+ Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 |
+ rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22
+ lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH
+ prim via lower_id layer (3 or 1) simply does not work. For TE (3)
+ it returns an error (len=-6) which is not evaluated by
+ handle_l1(), so the L1 layer status ends up wrong. Instead PH
+ must be sent via L4, only then does it reach L1 without an error
+ message. And NT PH prims only reach L1 when they are sent to
+ layer 2 id. --> use upper_id to send PH primitives. * Check for
+ errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are
+ improved. * The lower_id is now not used for anything, except:
+ Why is lower_id layer deleted when it wasn't created? I removed
+ this code since it looks very wrong. Patches:
+ patch04_l1activation.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett |
+ 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
+ chan_misdn: Fix loss of B channels if L1 is down. If you make 2
+ calls out an NT PTMP port which is not connected to any phone,
+ the B channel associated with that call becomes unusable until
+ Asterisk is restarted. The problem is the EVENT_SETUP is queued
+ when L1 is not up in misdn_lib_send_event(). If L1 cannot be
+ activated the event won't be dequeued. It gets even worse when
+ the call is hung up. The queued EVENT_SETUP will be overwritten
+ by an EVENT_DISCONNECT. The reserved B channel then will never be
+ freed. If later someone connects a phone to the port, L1 will
+ eventually activate and the queued EVENT_DISCONNECT is sent down
+ the stack. However, it is ignored because it is the wrong call
+ state. The real fix would be that activation and queueing for a
+ new SETUP is done by the NT stack. But since it doesn't, the
+ workaround must be removed because it doesn't always work. Fix:
+ The event is no longer queued but immediately sent to the stack.
+ If L1 cannot be activated, the L3 state machine that was started
+ by the EVENT_SETUP will do its work, i.e. a timeout will release
+ the B channel properly. The SETUP possibly cannot be sent the
+ first time but is resent by T303 in case L1 could be activated.
+ Patches: patch05_bchan-loss.diff (license #6372) patch uploaded
+ by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 |
+ rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13
+ lines chan_misdn: Remove some calls to exit(). Try proper cleanup
+ when something goes wrong in misdn_lib_init(). Especially do not
+ call exit()! * Fix memory leak because stack_destroy() does not
+ free the stack struct. Patches: patch06_cleanup-init.diff
+ (license #6372) patch uploaded by Guenther Kelleter Modified JIRA
+ ABE-2888 ........ Merged revisions 375519-375524 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 375625 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-02 17:38 +0000 [r375584-375615] Matthew Jordan <mjordan at digium.com>
+
+ * main/app.c, /: core: Fix a memory leak in app.c from an early
+ return ast_app_group_match_get_count allocates memory with the
+ regcomp function and we previously forgot to free it when bailing
+ out due to a regex compilation failure against category. (closes
+ issue AST-1018) Reported by: Guenther Kelleter Patches:
+ regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372)
+ ........ Merged revisions 375299 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, /: chan_dahdi: Fix segfault dereferencing
+ a NULL tech_pvt. The tech support customer was using the AMI
+ Redirect action shortly after a call was placed. While the
+ channel tried to do an ast_read(), the masquerade resulting from
+ the channel redirect took place. The masquerade in the middle of
+ the ast_read() resulted in the segfault. (closes issue AST-1025)
+ Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch
+ (license #5621) patch uploaded by rmudgett ........ Merged
+ revisions 375361 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * UPGRADE.txt, apps/app_queue.c, /: Multiple revisions
+ 375216,375242 ........ r375216 | jrose | 2012-10-18 15:58:07
+ -0500 (Thu, 18 Oct 2012) | 12 lines app_queue: Make ordering of
+ rrmemory/rrordered persist over add/remove members Prior to this
+ patch, adding, removing or reloading members to rrmemory would
+ cause the order to become completely jumbled. Now it behaves more
+ or less like rrordered other than the fact that it stores the
+ members on a hash table rather than a linked list. This patch
+ also prevents removal of members and member reloads from jumbling
+ rrordered queues. (issue AST-989) Reported by: Thomas Arimont
+ Review: https://reviewboard.asterisk.org/r/2164/ ........ r375242
+ | jrose | 2012-10-18 16:30:13 -0500 (Thu, 18 Oct 2012) | 8 lines
+ app_queue: add upgrade notes for 375216 Adds notes describing
+ behavioral changes to rrmemory strategy caused by 375216 (issue
+ AST-989) Reported by: Thomas Arimont ........ Merged revisions
+ 375216,375242 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h,
+ channels/sip/sdp_crypto.c, /, channels/chan_sip.c: Multiple
+ revisions 372709,373165,373532,373652,374456 ........ r372709 |
+ mjordan | 2012-09-08 20:19:21 -0500 (Sat, 08 Sep 2012) | 38 lines
+ 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 ........
+ r373165 | file | 2012-09-19 11:02:18 -0500 (Wed, 19 Sep 2012) |
+ 10 lines Fix a regression where direct media was not permitted
+ for calls using SIP INFO DTMF. A change was committed to fix
+ direct media ACL support. This change wrongly assumed that only a
+ single channel technology structure exists for chan_sip. This is
+ in fact false as a second exists for calls using SIP INFO DTMF.
+ The code which performs direct media ACL checking now checks for
+ both the non-INFO DTMF and INFO DTMF channel technology
+ structures. (closes issue ASTERISK-20409) Reported by: michele
+ cicciotti privatewave ........ r373532 | file | 2012-09-24
+ 19:09:46 -0500 (Mon, 24 Sep 2012) | 5 lines Add missing checks
+ that I neglected. The SIP technology and SIP info technology
+ should be considered equal. (closes issue ASTERISK-20409)
+ Reported by: michele cicciotti privatewave ........ r373652 |
+ twilson | 2012-09-25 12:21:19 -0500 (Tue, 25 Sep 2012) | 18 lines
+ Properly handle UAC/UAS roles for SIP session timers The SIP
+ session timer mechanism contains a mandatory 'refresher'
+ parameter (included in the Session-Expires header) which is used
+ in the session timer offer/answer signaling within a SIP Invite
+ dialog. It looks like asterisk is interpreting the uac resp. uas
+ role only as the initial role of client and server (caller is
+ uac, callee is uas). The standard rfc 4028 however assigns the
+ client role to the ((RE)-Invite) requester, the server role to
+ the ((RE)-Invite) responder. This patch has Asterisk track the
+ actual refresher as "us" or "them" as opposed to relying on just
+ the configured "uas" or "uac" properties. (closes issue AST-922)
+ Reported by: Thomas Airmont Review:
+ https://reviewboard.asterisk.org/r/2118/ ........ r374456 | file
+ | 2012-10-04 12:39:18 -0500 (Thu, 04 Oct 2012) | 14 lines Fix a
+ regression from direct media ACLs where the directrtpsetup option
+ no longer works. A check was added for direct media ACLs that
+ immediately forbid remote bridging if there was no bridged
+ channel. This caused directrtpsetup to no longer function as it
+ needs this information before bridging actually occurs. Logic has
+ now been adjusted so if there is no bridged channel a remote
+ bridge will still be attempted. (closes issue ASTERISK-20511)
+ Reported by: kristoff Review:
+ https://reviewboard.asterisk.org/r/2146/ ........ Merged
+ revisions 372709,373165,373532,373652,374456 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * funcs/func_math.c, main/xmldoc.c, apps/app_dial.c, /,
+ channels/chan_sip.c: Multiple revisions
+ 371357,371469,371860,372628 ........ r371357 | jrose | 2012-08-16
+ 13:57:27 -0500 (Thu, 16 Aug 2012) | 8 lines 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 ........ r371469 | mjordan | 2012-08-17 13:51:43 -0500
+ (Fri, 17 Aug 2012) | 14 lines 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
+ ........ r371860 | rmudgett | 2012-08-29 13:22:24 -0500 (Wed, 29
+ Aug 2012) | 12 lines 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 ........
+ r372628 | rmudgett | 2012-09-07 17:06:29 -0500 (Fri, 07 Sep 2012)
+ | 5 lines Remove annoying unconditional debug message from
+ INC/DEC functions. (closes issue AST-1001) Reported by: Guenther
+ Kelleter ........ Merged revisions 371357,371469,371860,372628
+ 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
+
+ * main/cel.c, main/channel.c, /: Multiple revisions
+ 370205,370273,370360 ........ r370205 | kpfleming | 2012-07-18
+ 14:12:03 -0500 (Wed, 18 Jul 2012) | 18 lines Resolve severe
+ memory leak in CEL logging modules. A customer reported a
+ significant memory leak using Asterisk 1.8. They have tracked it
+ down to ast_cel_fabricate_channel_from_event() in main/cel.c,
+ which is called by both in-tree CEL logging modules (cel_custom.c
+ and cel_sqlite3_custom.c) for each and every CEL event that they
+ log. The cause was an incorrect assumption about how data
+ attached to an ast_channel would be handled when the channel is
+ destroyed; the data is now stored in a datastore attached to the
+ channel, which is destroyed along with the channel at the proper
+ time. (closes issue AST-916) Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/2053/ ........ r370273 |
+ mjordan | 2012-07-19 17:00:14 -0500 (Thu, 19 Jul 2012) | 14 lines
+ Fix compilation error when MALLOC_DEBUG is enabled To fix a
+ memory leak in CEL, a channel datastore was introduced whose
+ destruction function pointer was pointed to the ast_free macro.
+ Without MALLOC_DEBUG enabled this compiles as fine, as ast_free
+ is defined as free. With MALLOC_DEBUG enabled, however, ast_free
+ takes on a definition from a different place then utils.h, and
+ became undefined. This patch resolves this by using a reference
+ to ast_free_ptr. When MALLOC_DEBUG is enabled, this calls
+ ast_free; when MALLOC_DEBUG is not enabled, this is defined to be
+ ast_free, which is defined to be free. (issue AST-916) Reported
+ by: Thomas Arimont ........ r370360 | kpfleming | 2012-07-23
+ 09:41:03 -0500 (Mon, 23 Jul 2012) | 10 lines Free any datastores
+ attached to dummy channels. Revision 370205 added the use of a
+ datastore attached to a dummy channel to resolve a memory leak,
+ but ast_dummy_channel_destructor() in this branch did not free
+ datastores, resulting in a continued (but slightly smaller)
+ memory leak. This patch backports the change to free said
+ datastores from the Asterisk trunk. (related to issue AST-916)
+ ........ Merged revisions 370205,370273,370360 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h, /,
+ channels/chan_sip.c: Help mitigate potential reinvite glare
+ scenarios. When Asterisk servers are set up back-to-back, and
+ direct media is to be used betweeen endpoints, it is fairly
+ common for the two Asterisk servers to send direct media
+ reinvites to each other simultaneously. This results in 491s and
+ ACKs being exchanged between the servers. While the media
+ eventually gets set up properly, the problem is that there can be
+ a noticeable delay for the streams to stabilize. This patch adds
+ a new directmedia option called "outgoing". With this set, an
+ immediate direct media reinvite will only be sent if the call
+ direction is outgoing. For incoming dialogs, an immediate direct
+ media reinvite will not be sent, but further "reactionary" direct
+ media reinvites may be sent. For those who are having some deja
+ vu, that's because this patch was originally committed to trunk
+ since there is a new configuration option added. After seeing a
+ bug report about audio being slow to set up on SIP calls, it
+ became apparent that this patch would be the best solution for
+ resolving the issue. The patch is unintrusive and will have no
+ effect unless the option is explicitly enabled. (closes issue
+ AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857)
+ reported by Matt Jordan ........ Merged revisions 370618 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/ssl.c, main/tcptls.c, /, channels/chan_sip.c: Resolve memory
+ leaks in TLS initialization and TLS client connections This patch
+ resolves two sources of memory leaks when using TLS in Asterisk:
+ 1) It removes improper initialization (and multiple
+ re-initializations) of portions of the SSL library. Asterisk
+ calls SSL_library_init and SSL_load_error_strings during SSL
+ initialization; collectively this obviates the need for calling
+ any of the following during initialization or client connection
+ handling: * ERR_load_crypto_strings (handled by
+ SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
+ SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
+ SSL_library_init) 2) Failure to completely clean up all memory
+ allocated by Asterisk and by the SSL library for TLS clients.
+ This included not freeing the SSL_CTX object in the SIP channel
+ driver, as well as not clearing the error stack when the TLS
+ client exited. Note that these memory leaks were found by Thomas
+ Arimont, and this patch was essentially written by him with some
+ minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
+ Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
+ Arimont (license 5525) Review:
+ https://reviewboard.asterisk.org/r/2105 ........ Merged revisions
+ 373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-11 15:46 +0000 [r374848] Matthew Jordan <mjordan at digium.com>
+
+ * main/cdr.c, /: Fix incorrect billing duration reported when batch
+ mode is enabled Similar to r369351, the billing duration can be
+ skewed when batch mode is enabled. This happened much more rarely
+ than the duration, as it only occured when the call was answered
+ (thereby indicating an actual answer time) and immediately hung
+ up on (indicating a billsec of 0). Since a billing time of '0'
+ can either mean that the call immediately ended or that the CDR
+ was improperly answered, we have to use additional information to
+ know whether or not we can trust the CDR billsec value. Prior to
+ this patch, we looked to see if we had a valid answer time. If we
+ did, and billsec was zero, we used the current time to calculate
+ what billsec value we could from the CDR being written. If batch
+ mode is enabled, this will incorrectly report a billsec value
+ being much greater than the actual duration of the call. Instead
+ of relying on the presence of an answer time to know whether or
+ not we can re-calculate the billsec for the CDR, we now also use
+ the presence of the CDR's end time to know if we need to
+ re-calculate or whether we can trust the billsec value that we
+ have. This prevents erroneous jumps in the billsec value, while
+ still making sure that in the worst case, some billing time will
+ be calculated. (closes issue AST-1016) Reported by: Thomas
+ Arimont Tested by: Thomas Arimont ........ Merged revisions
+ 374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-10 21:16 +0000 [r374807] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c, /: app_queue: Made pass connected line updates
+ from the caller to ringing queue members. Party A calls Party B
+ Party B puts Party A on hold. Party B calls a queue. Ringing
+ queue member D sees Party B identification. Party B transfers
+ Party A to the queue. Queue member D does not get a connected
+ line update for Party A. Queue member D answers the call and
+ still sees Party B information. However, if Party A later
+ transfers the call to Party C then queue member D gets a
+ connected line update for Party C. * Made pass connected line
+ updates from the caller to queue members while the queue members
+ are ringing. (closes issue AST-1017) Reported by: Thomas Arimont
+ (closes issue ABE-2886) Reported by: Thomas Arimont Tested by:
+ rmudgett ........ Merged revisions 374801 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 374802 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-05 20:12 +0000 [r374569] dlee <dlee at localhost>:
+
+ * main/manager.c: Improve AMI long line error handling In AMI's
+ parser, when it receives a long line (> 1024 characters), it
+ discards that line, but continues to process the message
+ normally. Typically, this is not a problem because a) who has
+ lines that long and b) usually a discarded line results in an
+ invalid message. But if that line is specifying an optional
+ field, then the message will be processed, you get a 'Response:
+ Success', but things don't work the way you expected them to.
+ This patch changes the behavior when a line-too-long parse error
+ occurs. * Changes the log message to avoid way-too-long (and
+ truncated anyways) log messages * Adds a 'parsing' status flag to
+ Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if,
+ well, a line is too long * Responds with an appropriate error if
+ parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John
+ Bigelow Review: https://reviewboard.asterisk.org/r/2142/
+
+2012-10-05 19:02 +0000 [r374541] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
+ channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Multiple
+ revisions 370563,374536 ........ r370563 | rmudgett | 2012-07-30
+ 11:47:19 -0500 (Mon, 30 Jul 2012) | 2 lines Release B channel
+ allocation on error path in chan_misdn. ........ r374536 |
+ rmudgett | 2012-10-05 13:20:01 -0500 (Fri, 05 Oct 2012) | 159
+ lines Merged revisions 374515-374535 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
+ (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
+ Made setup_bc() static. Patches: patch1_unused-code.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
+ ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
+ (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
+ states Patches: patch2_unused-states.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
+ | 16 lines chan_misdn: Remove unnecessary null pointer checks and
+ checks for stack->nt * cleanup_bc() is always called with valid
+ bc (or it would've crashed before). * Value of stack->nt is known
+ in advance at some places. * Rename handle_event() to
+ handle_event_te(), handle_frm() to handle_frm_te(). Patches:
+ patch3_checks.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified JIRA ABE-2882 ................ r374518 |
+ rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Fix spelling in log messages Patches:
+ patch4_spelling.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
+ 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
+ chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
+ calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
+ emptied, cleaned and set not in use, although
+ misdn_lib_send_event() already did the same. This is bad. When
+ it's not in use we are not allowed to touch it. * Moved log
+ message in front of the resulting actions and fixed it to match
+ the case. Patches: patch5_bccleanup.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
+ | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
+ etc., really bad stuff. * Fix return codes of cb_events() for
+ EVENT_SETUP to use caller's cleanup mechanisms. * Move
+ cl_queue_chan() call after bearer check. Patches:
+ patch6_leaks.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
+ 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
+ chan_misdn: We must initialize cause on sending a DISCONNECT. We
+ must initialize cause on sending a DISCONNECT, so it is later
+ correctly indicated to ast_channel in case the answer
+ (RELEASE/RELEASE_COMPLETE) does not include one. Patches:
+ patch7_hangupcause.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374522 |
+ rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Remove unused code for upqueue Patches:
+ patch8_unused-upqueue.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374523 |
+ rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Improve debugging (port number, messages fixed, dups
+ removed) Patches: patch9_debug.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
+ | 8 lines chan_misdn: Better debug: we can print_bc_info even if
+ there's no ast leg. Patches: patch10_debug-bc-2.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified. JIRA
+ ABE-2882 ................ r374534 | rmudgett | 2012-10-05
+ 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
+ setup_bc() is called too early for an incoming SETUP on TE. This
+ prevents the B channel from being setup for HDLC mode when
+ requested by the bearer capability and config option hdlc=yes. It
+ violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
+ connect to the channel until a CONNECT ACKNOWLEDGE message has
+ been received." * Call setup_bc() on receipt of
+ CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
+ PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
+ Guenther Kelleter Modified. JIRA ABE-2881 ................
+ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
+ | 2 lines chan_misdn: Remove some more deadcode. ................
+ ........ Merged revisions 370563,374536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-04 15:11 +0000 [r374424] dlee <dlee at localhost>:
+
+ * res/res_agi.c, main/db.c: Fix DBDelTree error codes for AMI, CLI
+ and AGI The AMI DBDelTree command will return Success/Key tree
+ deleted successfully even if the given key does not exist. The
+ CLI command 'database deltree' had a similar problem, but was
+ saved because it actually responded with '0 database entries
+ removed'. AGI had a slightly different error, where it would
+ return success if the database was unavailable. This came from
+ confusion about the ast_db_deltree retval, which is -1 in the
+ event of a database error, or number of entries deleted
+ (including 0 for deleting nothing). * Adds a Doxygen comment to
+ process_db_keys explaining its retval * Changed some poorly named
+ res variables to num_deleted * Specified specific errors when
+ calling ast_db_deltree (database unavailable vs. entry not found
+ vs. success) * Fixed similar bug in AGI database deltree, where
+ 'Database unavailable' results in successful result (closes issue
+ AST-967) Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/2138/
+
+2012-09-25 22:59 +0000 [r373772] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_agent.c, configs/agents.conf.sample, /: Remove dead
+ code and documentation for nonexistent feature. multiplelogin was
+ removed from chan_agent back in 1.6.0 when AgentCallbackLogin()
+ was removed. (closes issue AST-948) reported by Steve Pitts
+ ........ Merged revisions 373768 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-20 19:13 +0000 [r373243] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_meetme.c: Fix incorrect MeetME conference bridge
+ reference count decrementing and sometimes premature destruction.
+ When using the 'e' or 'E' option to MeetMe the configured
+ conference bridges are loaded and examined to see if any are
+ empty. If no conference bridges are empty the caller is prompted
+ to enter the number of one. This operation left around a pointer
+ to the last created conference bridge still containing
+ participants. When the caller that was not able to find any empty
+ conference bridge hung up this pointer was disposed of and the
+ reference count of the conference bridge decremented. If there
+ was only a single participant in the conference bridge it was
+ ultimately destroyed prematurely. (closes issue AST-994) Reported
+ by: John Bigelow
+
+2012-09-11 21:02 +0000 [r372884] 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
+
+2012-08-30 18:48 +0000 [r372052] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_iax2.c, 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 reported 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 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
+
+2012-08-29 21:29 +0000 [r371948] 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 ........
+
+2012-08-27 21:48 +0000 [r371752-371788] 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
+
+2012-08-17 16:02 +0000 [r371397-371440] 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
+
+ * 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
+
+2012-08-13 20:42 +0000 [r371229] Kinsey Moore <kmoore at digium.com>
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