[asterisk-commits] bebuild: tag certified-1.8.11-cert8 r375572 - /certified/tags/1.8.11-cert8/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Oct 31 15:34:09 CDT 2012
Author: bebuild
Date: Wed Oct 31 15:34:05 2012
New Revision: 375572
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=375572
Log:
Importing files for 1.8.11-cert8 release.
Added:
certified/tags/1.8.11-cert8/.lastclean (with props)
certified/tags/1.8.11-cert8/.version (with props)
certified/tags/1.8.11-cert8/ChangeLog (with props)
Added: certified/tags/1.8.11-cert8/.lastclean
URL: http://svnview.digium.com/svn/asterisk/certified/tags/1.8.11-cert8/.lastclean?view=auto&rev=375572
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--- certified/tags/1.8.11-cert8/ChangeLog (added)
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+2012-10-31 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.11-cert8 Released.
+
+2012-10-31 20:26 +0000 [r375568-375570] Matthew Jordan <mjordan at digium.com>
+
+ * /, UPGRADE.txt: Multiple revisions 375242,375244 ........ 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 ........ r375244 | jrose |
+ 2012-10-18 16:36:59 -0500 (Thu, 18 Oct 2012) | 4 lines Correct
+ version number in Upgrade.txt release notes pertaining to queue
+ order Showed 1.8.17 to 1.8.18, needs to be 1.8.18 to 1.8.19
+ ........ Merged revisions 375242,375244 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_queue.c, /: 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/ ........ Merged
+ revisions 375216 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/sip/include/sip.h, apps/app_voicemail.c, /,
+ channels/chan_sip.c, funcs/func_math.c: Multiple revisions
+ 369436,369557,369579,369626,369652,372628 for 1.8.11-cert8
+ ........ r369436 | twilson | 2012-06-27 15:58:51 -0500 (Wed, 27
+ Jun 2012) | 17 lines AST-2012-010: Clean up after a reinvite that
+ never gets a final response The basic problem is that if a
+ re-INVITE is sent by Asterisk and it receives a provisional
+ response, but no final response, then the dialog is never torn
+ down. In addition to leaking memory, this also leaks file
+ descriptors and will eventually lead to Asterisk no longer being
+ able to process calls. This patch just keeps track of whether
+ there is an outstanding re-INVITE, and if there is goes ahead and
+ cleans up everything as though there was no outstanding reinvite.
+ Review: https://reviewboard.asterisk.org/r/2009/ (closes issue
+ ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
+ Davies, Terry Wilson ........ r369557 | twilson | 2012-07-03
+ 09:27:02 -0500 (Tue, 03 Jul 2012) | 11 lines Better handle
+ re-INVITEs with provisional but no final repsonses A previous
+ attempt at fixing this issue had negative side effects related to
+ attended transfers which this patch should resolve. Many thanks
+ to Steve Davies for all of the good suggestions and testing.
+ (closes issue ASTERISK-19992) Reported by: Steve Davies Tested
+ by: Steve Davies, Terry Wilson Review:
+ https://reviewboard.asterisk.org/r/2009/ ........ r369579 |
+ twilson | 2012-07-03 11:58:16 -0500 (Tue, 03 Jul 2012) | 8 lines
+ More improvements to re-INVITEs timing out after a provisional
+ response There is no need to call check_pendings() on a final
+ response to an INVITE when destroying the scheduler entry as it
+ will be done later during normal processing. (issue
+ ASTERISK-19992) ........ r369626 | mjordan | 2012-07-05 12:01:52
+ -0500 (Thu, 05 Jul 2012) | 16 lines Do not send a BYE when a
+ provisional response arrives during a re-INVITE Commits r369557
+ and r369579 were done to improve handling of re-INVITEs when the
+ UA that was supposed to receive the re-INVITE fails to respond. A
+ limitation of those patches occurred when a UA sent a provisional
+ response to the re-INVITE. This triggered a sending of a BYE in
+ check_pending. This patch tweaks the handling of the re-INVITE
+ such that a BYE is not sent in response to those messages. (issue
+ ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
+ patches: (reinvite_tweak.diff license #5012 by Steve Davies)
+ ........ r369652 | kmoore | 2012-07-05 14:01:52 -0500 (Thu, 05
+ Jul 2012) | 22 lines AST-2012-011: Resolve heap corruption issue
+ with voicemail The heard and deleted arrays in the voicemail
+ state structure were not handled properly following the memory
+ leak fix in r354890 and a fix for an invalid free in r356797.
+ This could result in accessing and writing into freed memory. The
+ allocation for these arrays has been reworked to avoid the
+ possibility of invalid frees, access of freed memory, and crashes
+ that were occurring as a result of this. Locking around accesses
+ and modifications of the voicemail state structure members
+ dh_arraysize, heard, and deleted has been added to prevent
+ simultaneous modification and access when IMAP storage is in use.
+ If IMAP storage is not in use, this locking is not compiled in.
+ Review: https://reviewboard.asterisk.org/r/1994/ (closes issue
+ ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney,
+ Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license
+ 6273) ........ 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
+ 369436,369557,369579,369626,369652,372628 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-15 14:11 +0000 [r375024] Matthew Jordan <mjordan at digium.com>
+
+ * configs/sip.conf.sample, channels/sip/include/sip.h, /,
+ channels/chan_sip.c: 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/ ........ Merged
+ revisions 373652 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-11 15:46 +0000 [r374847] 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 21:31 +0000 [r374806-374832] 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
+
+ * 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 18:56 +0000 [r374540] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.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
+
+2012-09-14 20:25 +0000 [r373088] Matthew Jordan <mjordan at digium.com>
+
+ * include/asterisk/tcptls.h, main/tcptls.c, /, channels/chan_sip.c,
+ main/ssl.c: Multiple revisions 366052,367002,367416,369731,373061
+ ........ r366052 | mmichelson | 2012-05-10 11:10:18 -0500 (Thu,
+ 10 May 2012) | 7 lines Close the proper tcptls_session when
+ session creation fails. (issue AST-998) Reported by: Thomas
+ Arimont Tested by: Thomas Arimont ........ r367002 | mmichelson |
+ 2012-05-18 11:53:47 -0500 (Fri, 18 May 2012) | 17 lines Fix
+ memory leak of SSL_CTX structures in TLS core. SSL_CTX structures
+ were allocated but never freed. This was a bigger issue for
+ clients than servers since new SSL_CTX structures could be
+ allocated for each connection. Servers, on the other hand,
+ typically set up a single SSL_CTX for their lifetime. This is
+ solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an
+ ssl_ctx on it, it is freed so that a new one can take its place.
+ 2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
+ been added so that servers can properly free their SSL_CTXs.
+ (issue ASTERISK-19278) ........ r367416 | mmichelson | 2012-05-23
+ 15:27:47 -0500 (Wed, 23 May 2012) | 5 lines Only call
+ SSL_CTX_free if DO_SSL is defined. Thanks to Paul Belanger for
+ pointing out this error. ........ r369731 | mmichelson |
+ 2012-07-06 13:40:06 -0500 (Fri, 06 Jul 2012) | 19 lines Remove a
+ superfluous and dangerous freeing of an SSL_CTX. The problem here
+ is that multiple server sessions share a SSL_CTX. When one
+ session ended, the SSL_CTX would be freed and set NULL, leaving
+ the other sessions unable to function. The code being removed is
+ superfluous because the SSL_CTX structures for servers will be
+ properly freed when ast_ssl_teardown is called. (closes issue
+ ASTERISK-20074) Reported by Trevor Helmsley Patches:
+ ASTERISK-20074.diff uploaded by Mark Michelson (license #5049)
+ Testers: Trevor Helmsley ........ r373061 | mjordan | 2012-09-14
+ 14:07:20 -0500 (Fri, 14 Sep 2012) | 28 lines 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
+ 366052,367002,367416,369731,373061 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-08-30 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.11-cert7 Released.
+
+ * AST-2012-013: Resolve ACL rules being ignored during calls by some
+ IAX2 peers
+
+ * AST-2012-012: Resolve AMI User Unauthorized Shell Access through
+ ExternalIVR
+
+2012-08-24 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.11-cert6 Released.
+
+2012-08-24 13:55 +0000 [r371650-371651] Matthew Jordan <mjordan at digium.com>
+
+ * configs/sip.conf.sample, main/cel.c, channels/sip/include/sip.h,
+ main/xmldoc.c, main/channel.c, /, channels/chan_sip.c: Merge
+ patches for 1.8.11-cert6 This includes the following * r369351
+ for AST-883 * r368807 for AST-884 * r356604, r356650, r364203 for
+ AST-890 * r370618 for AST-896 * r370205, r370273, r370360 for
+ AST-916 * r371469 for AST-932
+
+ * main/cdr.c, /: Merge r369351 for AST-883
+
+2012-08-07 15:40 +0000 [r370843] Mark Michelson <mmichelson at digium.com>
+
+ * 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
+ is to treat the IPorHost portion of the dial string as a
+ temporary outbound proxy. This ensures requests get sent to the
+ proper location. Due to the age of the request, some parts were
+ no longer relevant. For instance, the request moved outbound
+ proxy parsing code into a single method. This is done in a
+ previous commit, so it was not necessary to do again. Also, the
+ review request fixed some errors with regards to request routing
+ for CANCEL and ACK requests. This has also been fixed in more
+ recent commits. (closes issue ASTERISK-19677) reported by Walter
+ Doekes Review https://reviewboard.asterisk.org/r/1859 ........
+ Merged revisions 370769 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-07-27 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.11-cert5 Released.
+
+2012-07-11 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.11-cert5-rc2 Released.
+
+ * channels/sig_analog.c: Fix bad merge of r368759 in sig_analog
+
+ The patch for r368759 in Asterisk 1.8 relied upon the methods
+ analog_unlock_private/analog_lock_private. In earlier versions of
+ 1.8, including the Certified Asterisk 1.8.11 branch, those two
+ methods were unused, and hence were undefined out of the source.
+ When the patch was made for 1.8.11-cert5, those two functions were
+ not re-defined back in. This caused linking errors when sig_analog
+ was loaded.
+
+ This patch properly restores those two methods, such that the fix
+ for AST-891 works correctly.
+
+ (issue AST-891)
+
+2012-07-09 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.11-cert5-rc1 Released.
+
+2012-07-09 19:59 +0000 [r369848] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
+ channels/sig_analog.c, /, channels/chan_sip.c,
+ include/asterisk/channel.h: Fix deadlock between bridged channels
+ that attempt to set the hangup source Calling
+ ast_set_hangupsource with the channel lock held can result in a
+ deadlock because the function also locks the bridged channel.
+ (issue AST-891)
+
+2012-07-09 19:50 +0000 [r369845] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Add support for exposing the received
+ contact URI and also for setting the request URI in messages.
+ (closes issue AST-911)
+
+2012-07-09 19:06 +0000 [r369839-369840] Jason Parker <jparker at digium.com>
+
+ * include/asterisk/app_voicemail.h (removed): Remove file that
+ should no longer exist.
+
+ * apps/app_mixmonitor.c, apps/app_voicemail.c,
+ include/asterisk/callerid.h, include/asterisk/app.h,
+ channels/chan_sip.c, apps/app_voicemail.exports.in,
+ tests/test_voicemail_api.c, main/callerid.c, main/app.c,
+ include/asterisk/app_voicemail.h: Re-merge changes that were
+ reverted.
+ ------------------------------------------------------------------------
+ r365395 | qwell | 2012-05-04 16:17:08 -0500 (Fri, 04 May 2012) |
+ 7 lines Add support for folders in MixMonitor 'm' option.
+ Backport manager actions. The manager actions are needed, so
+ MixMonitor can be executed on existing channels. (issue DPMA-68)
+ ------------------------------------------------------------------------
+ r364761 | qwell | 2012-05-01 12:25:14 -0500 (Tue, 01 May 2012) |
+ 6 lines Remove folder_dir from voicemail snapshots API. It was
+ both unused (except in tests, where it was fudged) and
+ unnecessary. (closes issue AST-842)
+ ------------------------------------------------------------------------
+ r367161 | mmichelson | 2012-05-21 14:05:52 -0500 (Mon, 21 May
+ 2012) | 21 lines Add "send to voicemail" Digium phone
+ functionality to Asterisk. This change accommodates two methods
+ by which calls can be directed to a user's voicemail. * Incoming
+ calls can be redirected to any user's voicemail. * Established
+ calls can be blind transferred to any user's voicemail. Digium
+ phones indicate the desire to direct a call to voicemail by using
+ a Diversion header with a reason parameter of "send_to_vm". This
+ patch adds the "send_to_vm" reason as a valid redirecting reason.
+ In addition, chan_sip.c has been modified to update redirecting
+ information on the transferred channel by reading a Diversion
+ header on a REFER request. (closes issue AST-871) Reported by
+ Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925
+ ------------------------------------------------------------------------
+ r368790 | mjordan | 2012-06-12 08:44:36 -0500 (Tue, 12 Jun 2012)
+ | 18 lines Fix deadlock in SIP transfers that involve a REFER
+ request In r367163, "send to voicemail" functionality was added
+ to the SIP channel driver. This required updating the party
+ redirecting information for the channel based on the headers
+ provided in the REFER request. When the redirecting party
+ information is updated on the channel, a call to
+ ast_indicate_data occurs. Because handle_request_refer still had
+ the sip_pvt locked, a deadlock could occur between the pbx_thread
+ and the do_monitor thread servicing the REFER request. This patch
+ preserves the proper locking order between the channel and the
+ sip_pvt by ensuring that the sip_pvt is unlocked prior to
+ updating the party redirecting information on the channel.
+ (closes issue AST-903) Reported by: Matt Jordan patches:
+ jira_ast_903_trunk.patch by rmudgett (license 5621)
+ ------------------------------------------------------------------------
+ r368962 | qwell | 2012-06-14 13:38:48 -0500 (Thu, 14 Jun 2012) |
+ 11 lines Remove global symbol requirement from app_voicemail.
+ This uses the existing "function installation" stuff that already
+ existed for other functions, like getting message counts. (closes
+ issue AST-807) (issue AST-901) (issue AST-908) Review:
+ https://reviewboard.asterisk.org/r/1965/
+ ------------------------------------------------------------------------
+ r368964 | qwell | 2012-06-14 14:03:24 -0500 (Thu, 14 Jun 2012) |
+ 8 lines These functions that were moved need to be static. Also
+ wrap test functions in a #ifdef. (issue AST-807) (issue AST-901)
+ (issue AST-908)
+ ------------------------------------------------------------------------
+ r368998 | qwell | 2012-06-15 10:31:43 -0500 (Fri, 15 Jun 2012) |
+ 6 lines Remove some symbol exports that got missed in the removal
+ of global symbols. (issue AST-807) (issue AST-901) (issue
+ AST-908)
+ ------------------------------------------------------------------------
+ r369024 | qwell | 2012-06-15 11:29:40 -0500 (Fri, 15 Jun 2012) |
+ 2 lines Fix voicemail API tests by using the correct argument
+ order for create/destroy.
+ ------------------------------------------------------------------------
+
+2012-07-05 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.11-cert4 Released.
+
+ * AST-2012-010
+
+ * AST-2012-011
+
+2012-05-29 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.11-cert2 Released.
+
+ * AST-2012-007
+
+ * AST-2012-008
+
+2012-04-25 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Certified Asterisk 1.8.11-cert1 Released.
+
+2012-04-25 16:53 +0000 [r363674] Jason Parker <jparker at digium.com>
+
+ * / (added): Asterisk 1.8-digiumphones branch has become Certified
+ Asterisk 1.8.11. For more details about Certified Asterisk, see
+ http://tinyurl.com/7pfp639
+
+2012-04-24 20:57 +0000 [r363374] Jason Parker <jparker at digium.com>
+
+ * /res/res_smdi.c,
+ /apps/app_osplookup.c,
+ /channels/chan_misdn.c,
+ /channels/chan_skinny.c,
+ /funcs/func_frame_trace.c,
+ /cdr/cdr_sqlite.c,
+ /pbx/pbx_realtime.c,
+ /apps/app_amd.c,
+ /pbx/pbx_dundi.c,
+ /apps/app_url.c,
+ /channels/chan_nbs.c,
+ /apps/app_externalivr.c,
+ /apps/app_zapateller.c,
+ /cdr/cdr_odbc.c,
+ /res/res_fax_spandsp.c,
+ /channels/chan_mgcp.c,
+ /cel/cel_pgsql.c,
+ /apps/app_readfile.c,
+ /apps/app_test.c,
+ /apps/app_ices.c,
+ /channels/chan_gtalk.c,
+ /cdr/cdr_csv.c,
+ /channels/chan_phone.c,
+ /funcs/func_pitchshift.c,
+ /apps/app_waitforring.c,
+ /formats/format_vox.c,
+ /res/res_timing_pthread.c,
+ /apps/app_minivm.c,
+ /channels/chan_h323.c,
+ /cel/cel_sqlite3_custom.c,
+ /apps/app_confbridge.c,
+ /res/res_config_ldap.c,
+ /apps/app_nbscat.c,
+ /cdr/cdr_sqlite3_custom.c,
+ /res/res_snmp.c,
+ /apps/app_dictate.c,
+ /apps/app_waitforsilence.c,
+ /apps/app_dahdiras.c,
+ /pbx/pbx_lua.c,
+ /apps/app_alarmreceiver.c,
+ /apps/app_image.c,
+ /res/res_ael_share.c,
+ /cdr/cdr_tds.c,
+ /apps/app_setcallerid.c,
+ /apps/app_mp3.c,
+ /channels/chan_alsa.c,
+ /res/res_timing_kqueue.c,
+ /channels/chan_unistim.c,
+ /apps/app_dahdibarge.c,
+ /res/res_config_pgsql.c,
+ /res/res_adsi.c,
+ /res/res_phoneprov.c,
+ /apps/app_morsecode.c,
+ /cdr/cdr_pgsql.c,
+ /res/res_config_sqlite.c,
+ /channels/chan_jingle.c,
+ /pbx/pbx_ael.c,
+ /apps/app_sms.c,
+ /formats/format_jpeg.c,
+ /apps/app_jack.c,
+ /apps/app_adsiprog.c,
+ /cel/cel_radius.c,
+ /res/res_ais.c,
+ /cel/cel_tds.c,
+ /apps/app_festival.c,
+ /apps/app_chanisavail.c,
+ /channels/chan_console.c,
+ /apps/app_talkdetect.c,
+ /res/res_jabber.c,
+ /cdr/cdr_radius.c,
+ /apps/app_getcpeid.c,
+ /channels/chan_oss.c: Disable extended
+ and deprecated modules by default. Users can still enable any of
+ these using menuselect if they so choose. (closes issue AST-873)
+
+2012-04-23 15:17 +0000 [r363161] Jason Parker <jparker at digium.com>
+
+ * /main/manager.c,
+ ,
+ /channels/chan_sip.c,
+ /channels/chan_skinny.c: Multiple
+ revisions 363102,363106,363141 ........ r363102 | mjordan |
+ 2012-04-23 08:37:55 -0500 (Mon, 23 Apr 2012) | 16 lines
+ AST-2012-005: Fix remotely exploitable heap overflow in keypad
+ button handling When handling a keypad button message event, the
+ received digit is placed into a fixed length buffer that acts as
+ a queue. When a new message event is received, the length of that
+ buffer is not checked before placing the new digit on the end of
+ the queue. The situation exists where sufficient keypad button
+ message events would occur that would cause the buffer to be
+ overrun. This patch explicitly checks that there is sufficient
+ room in the buffer before appending a new digit. (closes issue
+ ASTERISK-19592) Reported by: Russell Bryant ........ Merged
+ revisions 363100 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ r363106 | mjordan | 2012-04-23 09:05:02 -0500 (Mon, 23 Apr 2012)
+ | 17 lines AST-2012-006: Fix crash in UPDATE handling when no
+ channel owner exists If Asterisk receives a SIP UPDATE request
+ after a call has been terminated and the channel has been
+ destroyed but before the SIP dialog has been destroyed, a
+ condition exists where a connected line update would be attempted
+ on a non-existing channel. This would cause Asterisk to crash.
+ The patch resolves this by first ensuring that the SIP dialog has
+ an owning channel before attempting a connected line update. If
+ an UPDATE request is received and no channel is associated with
+ the dialog, a 481 response is sent. (closes issue ASTERISK-19770)
+ Reported by: Thomas Arimont Tested by: Matt Jordan Patches:
+ ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license
+ 6283) ........ r363141 | jrose | 2012-04-23 09:33:16 -0500 (Mon,
+ 23 Apr 2012) | 20 lines AST-2012-004: Fix an error that allows
+ AMI users to run shell commands sans authorization. As detailed
+ in the advisory, AMI users without write authorization for SYSTEM
+ class AMI actions were able to run system commands by going
+ through other AMI commands which did not require that
+ authorization. Specifically, GetVar and Status allowed users to
+ do this by setting their variable/s options to the SHELL or EVAL
+ functions. Also, within 1.8, 10, and trunk there was a similar
+ flaw with the Originate action that allowed users with originate
+ permission to run MixMonitor and supply a shell command in the
+ Data argument. That flaw is fixed in those versions of this
+ patch. (closes issue ASTERISK-17465) Reported By: David Woolley
+ Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) ........ Merged revisions 363117 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 363102,363106,363141 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-19 20:31 +0000 [r362673] Mark Michelson <mmichelson at digium.com>
+
+ * /channels/chan_sip.c: Add a test
+ application for sending custom SIP INFO messages. When
+ TEST_FRAMEWORK is enabled, SIPSendCustomInfo is available to test
+ sending custom INFO requests. Review:
+ https://reviewboard.asterisk.org/r/1866
+
+2012-04-13 17:19 +0000 [r362042-362132] Matthew Jordan <mjordan at digium.com>
+
+ * : Rename property branches-1.8-merged to
+ branch-1.8-merged
+
+ * : Update properties on 1.8-digiumphones
+ Change the merge property tag from svnmerge-integrated to
+ branches-1.8-merged. Added merged revisions from r362042.
+
+ * ,
+ /channels/chan_sip.c,
+ /main/features.c: Merge of several
+ needed fixes for 1.8-digiumphones This merges fixes for the
+ following issues into the 1.8-digiumphones branch: *
+ ASTERISK-19355 - Call transfer with consultation frequently fails
+ in cross- linked Asterisk scenario (directmedia & sendrpid
+ active) * ASTERISK 19365 - Remote SIP Call legs are frequently
+ not released in a cross-linked Asterisk scenario (directmedia &
+ sendrpid) * ASTERISK-19183 - Sporadically missing connectedline
+ event to caller channel in directed pickup app
+
+2012-04-09 20:40 +0000 [r361704] Mark Michelson <mmichelson at digium.com>
+
+ * /apps/app_voicemail.c,
+ /apps/app_voicemail.exports.in,
+ /tests/test_voicemail_api.c (added),
+ /include/asterisk/app_voicemail.h: Fix
+ bugs in voicemail APIs and add unit tests. There were several
+ crashes that could occur due to NULL inputs, invalid inputs, and
+ the like. This fixes all known ones and adds unit tests to
+ exercise the APIs.
+
+2012-04-06 19:08 +0000 [r361502] Richard Mudgett <rmudgett at digium.com>
+
+ * /main/message.c: Update Func MESSAGE()
+ and AMI MessageSend documentation. * Document
+ MESSAGE(custom_data) * Update AMI MessageSend documentation *
+ Eliminate a shadowed variable name in msg_func_write() for
+ custom_data.
+
+2012-04-05 17:24 +0000 [r361283] Mark Michelson <mmichelson at digium.com>
+
+ * /funcs/func_presence_state.c,
+ /tests/test_config.c: Add additional
+ configuration and presence unit tests. These were originally
+ written while merging features into trunk, but these tests apply
+ just as much for the 1.8 version of Digium phones, so might as
+ well have them here, too.
+
+2012-04-03 21:03 +0000 [r361088] Jonathan Rose <jrose at digium.com>
+
+ * /apps/app_mixmonitor.c: Make m option
+ for mixmonitor delete the source file once it is finished copying
+ to vm. Review: https://reviewboard.asterisk.org/r/1842/
+
+2012-03-29 21:49 +0000 [r360826] Jason Parker <jparker at digium.com>
+
+ * /main/manager.c,
+ ,
+ /main/utils.c,
+ /include/asterisk/manager.h,
+ /apps/app_milliwatt.c: Multiple
+ revisions 359656,359706,359979 ........ r359656 | mjordan |
+ 2012-03-15 13:35:59 -0500 (Thu, 15 Mar 2012) | 22 lines Fix
+ remotely exploitable stack overrun in Milliwatt Milliwatt is
+ vulnerable to a remotely exploitable stack overrun when using the
+ 'o' option. This occurs due to the milliwatt_generate function
+ not accounting for AST_FRIENDLY_OFFSET when calculating the
+ maximum number of samples it can put in the output buffer. This
+ patch resolves this issue by taking into account
+ AST_FRIENDLY_OFFSET when determining the maximum number of
+ samples allowed. Note that at no point is remote code execution
+ possible. The data that is written into the buffer is the
+ pre-defined Milliwatt data, and not custom data. (closes issue
+ ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt
+ Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell
+ Bryant (license 6283) Note that this patch was written by
+ Russell, even though Matt uploaded it ........ Merged revisions
+ 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+ ........ r359706 | mjordan | 2012-03-15 14:01:22 -0500 (Thu, 15
+ Mar 2012) | 16 lines Fix remotely exploitable stack overflow in
+ HTTP manager There exists a remotely exploitable stack buffer
+ overflow in HTTP digest authentication handling in Asterisk. The
+ particular method in question is only utilized by HTTP AMI. When
+ parsing the digest information, the length of the string is not
+ checked when it is copied into temporary buffers allocated on the
+ stack. This patch fixes this behavior by parsing out pre-defined
+ key/value pairs and avoiding unnecessary copies to the stack.
+ (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
+ by: Matt Jordan ........ r359979 | rmudgett | 2012-03-20 12:21:16
+ -0500 (Tue, 20 Mar 2012) | 28 lines Allow AMI action callback to
+ be reentrant. Fix AMI module reload deadlock regression from
+ ASTERISK-18479 when it tried to fix the race between calling an
+ AMI action callback and unregistering that action. Refixes
+ ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2
+ object guaranteed that there were no active callbacks that
+ mattered when ast_manager_unregister() was called. Unfortunately,
+ this causes the deadlock situation. The patch stops locking the
[... 38079 lines stripped ...]
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