[asterisk-commits] bebuild: tag 11.1.0-rc1 r375961 - /tags/11.1.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Nov 6 10:34:18 CST 2012
Author: bebuild
Date: Tue Nov 6 10:34:14 2012
New Revision: 375961
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=375961
Log:
Importing files for 11.1.0-rc1 release.
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tags/11.1.0-rc1/.version (with props)
tags/11.1.0-rc1/ChangeLog (with props)
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+2012-11-06 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.1.0-rc1 Released.
+
+2012-11-06 12:09 +0000 [r375925] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_motif.c: Fix a bug where our Motif ICE candidates
+ were not quite proper, and make us more forgiving. An issue was
+ reported on the mailing list where calling would result in an
+ "Incomplete ICE-UDP candidate received on session" error message.
+ This is the result of the ICE-UDP candidate code not placing a
+ "network" attribute within the candidates. This is now done. To
+ increase compatibility though I have removed the requirement for
+ the "network" attribute to exist within ICE-UDP candidates that
+ are received since we don't actually require the value. Reported
+ on the mailing list by Jean-Denis Girard.
+
+2012-11-05 23:09 +0000 [r375895] Matthew Jordan <mjordan at digium.com>
+
+ * main/timing.c, main/channel.c, /, res/res_timing_pthread.c,
+ res/res_timing_dahdi.c, res/res_timing_timerfd.c,
+ bridges/bridge_softmix.c, funcs/func_jitterbuffer.c,
+ include/asterisk/timing.h, res/res_musiconhold.c,
+ channels/chan_iax2.c, res/res_fax_spandsp.c,
+ res/res_timing_kqueue.c: Refactor ast_timer_ack to return an
+ error and handle the error in timer users Currently, if an
+ acknowledgement of a timer fails Asterisk will not realize that a
+ serious error occurred and will continue attempting to use the
+ timer's file descriptor. This can lead to situations where errors
+ stream to the CLI/log file. This consumes significant resources,
+ masks the actual problem that occurred (whatever caused the timer
+ to fail in the first place), and can leave channels in odd
+ states. This patch propagates the errors in the timing resource
+ modules up through the timer core, and makes users of these
+ timers handle acknowledgement failures. It also adds some
+ defensive coding around the use of timers to prevent using bad
+ file descriptors in off nominal code paths. Note that the patch
+ created by the issue reporter was modified slightly for this
+ commit and backported to 1.8, as it was originally written for
+ Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/
+ (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches:
+ jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license
+ 6358) ........ Merged revisions 375893 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375894 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-05 21:41 +0000 [r375864] Richard Mudgett <rmudgett at digium.com>
+
+ * main/loader.c, /: Add safety NULL pointer check in module user
+ references. Made __ast_module_user_remove() check for NULL
+ pointers. ........ Merged revision 375860 from C.3 ........
+ Merged revisions 375862 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375863 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-05 17:59 +0000 [r375847] Jonathan Rose <jrose at digium.com>
+
+ * /, UPGRADE.txt: chan_sip: Document a change to user-field
+ encoding introduced with r303509 The change in question was added
+ to improve compliance with RFC3261, but at the time of commit, it
+ wasn't adequately documented in the UPGRADE notes. (closes issue
+ ASTERISK-20561) Reported by: Deniz Review:
+ https://reviewboard.asterisk.org/r/2177/ ........ Merged
+ revisions 375846 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-04 03:09 +0000 [r375729-375802] Matthew Jordan <mjordan at digium.com>
+
+ * main/manager.c, /: Don't attempt to purge sessions when no
+ sessions exist Manager's tcp/tls objects have a periodic function
+ that purge old manager sessions periodically. During shutdown,
+ the underlying container holding those sessions can be disposed
+ of and set to NULL before the tcp/tls periodic function is
+ stopped. If the periodic function fires, it will attempt to
+ iterate over a NULL container. This patch checks for whether or
+ not the sessions container exists before attempting to purge
+ sessions out of it. If the sessions container is NULL, we simply
+ return. Note that this error was also caught by the Asterisk Test
+ Suite. ........ Merged revisions 375800 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375801 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, res/res_fax.c: Only deref a reserved gateway session if we
+ actually reserved one Its perfectly acceptable to have a gateway
+ session unreserved when we go to first allocate one. Unreffing
+ the reserved gateway session - when its NULL - will result in an
+ assertion error. This problem was caught by the Asterisk Test
+ Suite (once we had enough of the debugging flags enabled)
+ ........ Merged revisions 375797 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/manager.c, /: Properly clean up manager resources on exit
+ This patch does two things: 1) It properly unregisters the
+ manager CLI commands 2) It cleans up AMI users on exit. Prior to
+ this patch, the AMI users were not being disposed of properly,
+ resulting in a memory leak. (closes issue ASTERISK-20646)
+ Reported by: Corey Farrell patches: manager_shutdown.patch
+ uploaded by Corey Farrell (license 5909) ........ Merged
+ revisions 375793 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375794 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/db.c, /: Properly finalize prepared SQLite3 statements to
+ prevent memory leak The AstDB uses prepared SQLite3 statements to
+ retrieve data from the SQLite3 database. These statements should
+ be finalized during Asterisk shutdown so that the SQLite3
+ database can be properly closed. Failure to finalize the
+ statements results in a memory leak and a failure when closing
+ the database. This patch fixes those issues by ensuring that all
+ prepared statements are properly finalized at shutdown. (closes
+ issue ASTERISK-20647) Reported by: Corey Farrell patches:
+ astdb-sqlite3_close.patch uploaded by Corey Farrell (license
+ 5909) ........ Merged revisions 375761 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/xmldoc.c: Fix memory leaks in XML documentation This patch
+ fixes two memory leaks: 1) When building XML documentation items,
+ the 'name' attribute was extracted from XML elements but not
+ properly freed after being copied into the item being built. 2)
+ When unloading XML documentation, the doctree container objects
+ were not properly freed. This patch corrects these memory leaks.
+ Note that this patch was modified slightly for this commmit, as
+ the case where the 'name' attribute doesn't exist also wasn't
+ handled in the item construction. This patch also checks for that
+ attribute not existing. (closes issue ASTERISK-20648) Reported
+ by: Corey Farrell Tested by: mjordan patches:
+ xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909)
+
+ * main/cdr.c, /: Prevent multiple CDR batches from conflicting when
+ scheduling the CDR write The Asterisk Test Suite caught an error
+ condition where a scheduled CDR batch write can be deleted twice
+ if two channels attempt to post their CDRs at the same time. The
+ batch CDR mutex is locked while the CDRs are appended to the
+ current batch list; however, it is unlocked prior to actually
+ scheduling the CDR write. As such, two threads can attempt to
+ remove the currently scheduled batch write at the same time,
+ resulting in an assertion error. This patch extends the time that
+ the mutex is locked to encompass actually scheduling the write.
+ This prevents two threads from unscheduling the currently
+ scheduled write at the same time. ........ Merged revisions
+ 375727 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375728 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-03 03:17 +0000 [r375702] Andrew Latham <lathama at gmail.com>
+
+ * README, include/asterisk/doxyref.h: Doxygen Updates Replace links
+ to missing text files removed in the 1.6.x series with links to
+ the wiki. Doxygen can handle URLs fine, don't atempt to quote
+ them. Also update the wiki link in the Readme to get everyone on
+ the same page. (issue ASTERISK-20259) ........ Merged revisions
+ 375698 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375699 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-02 20:59 +0000 [r375661] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, channels/chan_misdn.c, /, main/ccss.c,
+ main/format_pref.c: Things don't need to be that const. ........
+ Merged revisions 375658 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375659 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-02 20:56 +0000 [r375660] Damien Wedhorn <voip at facts.com.au>
+
+ * channels/chan_skinny.c: Fix for chan_skinny leaving RTP ports
+ open Skinny wasn't closing RTP sockets. This patch includes
+ ast_rtp_instance_stop before ast_rtp_instance_destroy which fixes
+ the problem. Also add destroy for VRTP (which I believe is
+ unused, but exists). Review:
+ https://reviewboard.asterisk.org/r/2176/
+
+2012-11-02 18:44 +0000 [r375627] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_lib.h, /, channels/misdn/isdn_lib.c: 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 ........ Merged
+ revisions 375626 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-02 17:24 +0000 [r375613] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, channels/chan_sip.c: Fix Wrong Result In Debug Message For SDP
+ Origin Processing While looking at some debug logs, I noticed
+ that it was being reported that the SDP origin line was
+ unsupported or failed. Upon looking into this on my local
+ machine, I found that I too was getting this debug message yet
+ everything seemed to be getting processed properly. What was
+ discovered is, that, the variable to determine what is displayed
+ in the debug message for the SDP line that was processed, was not
+ being set for the origin line when the result was successful.
+ This patch fixes this and was tested on local machine. ........
+ Merged revisions 375594 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375601 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-11-01 14:52 +0000 [r375575] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Fix a bug
+ causing SIP reloads to remove all entries from the registry A
+ regression was introduced in chan_sip by changes to sip reload
+ introduced by r349097. That patch moved peer purging from the
+ beginning of the reload to after the general configuration was
+ finished. This patch fixes that by undoing the repositioning of
+ the original peer purging code and using a similar function after
+ performing general configuration that purges only autocreated
+ peers that were created when persist mode isn't enabled. (closes
+ issue ASTERISK-20611) Reported by: Alisher Review:
+ https://reviewboard.asterisk.org/r/2171/
+
+2012-10-31 18:00 +0000 [r375559] Joshua Colp <jcolp at digium.com>
+
+ * res/res_http_websocket.exports.in: Fix an issue with
+ res_http_websocket where the chan_sip WebSocket handler could not
+ be registered. On some systems the optional API support uses the
+ GCC compiler attribute "weakref" to provide its functionality.
+ This code changes the function names and prefixes "__" to the
+ front. The res_http_websocket exports file did not take this into
+ account, thereby not allowing those functions to be global and
+ ultimately found. (closes issue ASTERISK-20631) Reported by:
+ danjenkins
+
+2012-10-31 14:49 +0000 [r375532] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_calendar_ews.c, /: Properly extract the Body information
+ of an EWS calendar item Unlike all other calendar modules,
+ res_calendar_ews fails to extract the Body information for a
+ calendar item. This is due, in part, to a quirk in the schema in
+ the XML - not only does a CalendarItem contain a Body element,
+ but the CalendarItem exists as a descendant of a different Body
+ element. The neon parser was erroneously skipping all Body
+ elements. This patch fixes that by bypassing Body elements that
+ are not a child of CalendarItem, and parsing the Body element out
+ if it is a child. Note that the original patch by Terry Wilson
+ only needed slight modifications to make it properly pull the
+ Body information out; as such, while I've linked to the patch
+ that I uploaded for Dmitry, I've attributed the patch to Terry.
+ (closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested
+ by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff
+ uploaded by Terry Wilson (license 6283) ........ Merged revisions
+ 375528 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375531 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-30 19:23 +0000 [r375506] Richard Mudgett <rmudgett at digium.com>
+
+ * /, bridges/bridge_softmix.c: Fix ConfBridge crash if no timing
+ module loaded. (closes issue ASTERISK-19448) Reported by: feyfre
+ Patches: smfix.patch (license #6099) patch uploaded by feyfre
+ Modified for coding guidelines. ........ Merged revisions 375496
+ from http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-30 19:09 +0000 [r375471-375486] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_mixmonitor.c: mixmonitor: Add a test event This test
+ event is being used to fix the mixmonitor_audiohook_inherit test.
+ ........ Merged revisions 375484 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375485 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, apps/app_confbridge.c: confbridge: Fix a bug which made
+ conferences not record with AMI/CLI commands When confbridge was
+ changed to handle conference status with a state machine in
+ r374658. The function responsible for starting recording for a
+ conference was refactored with the function actually responsible
+ for launching the recording thread being split into a function
+ with another name. The old function name was still used for
+ manually started recordings through AMI or CLI. This patch fixes
+ that by switching which function is used to start recording the
+ conference. (closes issue ASTERISK-20601) Reported by: Vilius
+ Patches: confbridge_mixmonitor.diff uploaded by Jonathan Rose
+ (license 6182) ........ Merged revisions 375470 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-30 02:22 +0000 [r375469] Matthew Jordan <mjordan at digium.com>
+
+ * /, apps/app_queue.c: Ensure that the Queue application tracks
+ busy members in off nominal situations There are a few code paths
+ where the Queue application fails to count a paused or in use
+ queue member as being 'busy'. This can cause callers to get stuck
+ in the Queue until a paused agent unpauses themselves. (closes
+ issue ASTERISK-20623) Reported by: Bryan Walters patches:
+ app_queue.patch uploaded by Bryan Walters (license 5851) ........
+ Merged revisions 375450 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375451 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-29 21:23 +0000 [r375437] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Prevent resetting of NATted realtime peer
+ address on reload. If a "sip reload" is issued for a SIP peer,
+ then his IP address will be cleared, thus resulting in forgetting
+ the public IP address. Asterisk will then attempt to route SIP
+ traffic to the private IP address. The fix here is to make "sip
+ reload" ignore realtime peers when "host = dynamic" is spotted.
+ Realtime peers can now only have their IP address reset if they
+ have gone from being not dynamic to being dynamic. (closes issue
+ ASTERISK-18203) reported by daren ferreira (closes issue
+ ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff
+ uploaded by JoshE (license #6075) ........ Merged revisions
+ 375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375417 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-29 19:29 +0000 [r375363-375390] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/features.c: Fix the Park 'r' option when a channel parks
+ itself. When a channel uses the Park appliation to park itself
+ with the 'r' option, the channel hears music-on-hold instead of
+ the requested ringing. * Added a missing check for the 'r' option
+ when a channel parks itself. (closes issue ASTERISK-19382)
+ Reported by: James Stocks Patches by: dsessions Review:
+ https://reviewboard.asterisk.org/r/2148/ ........ Merged
+ revisions 375388 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375389 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * 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 ........ Merged
+ revisions 375362 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-23 16:22 +0000 [r375288-375327] Jonathan Rose <jrose at digium.com>
+
+ * contrib/scripts/ast_tls_cert, /: ast_tls_cert script: Better
+ response for various exit conditions to openssl (closes issue
+ ASTERISK-20260) Reported by: Daniel O'Connor Patches:
+ ast_tls_cert-update.diff uploaded by Daniel O'Connor (license
+ 6419) ........ Merged revisions 375325 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375326 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, 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 ........ Merged
+ revisions 375300 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, codecs/gsm/src/code.c: GSM: Fix encoding problems with GSM
+ (closes issue ASTERISK-20457) Reported by: Richard Miller
+ Patches: code.patch uploaded by Richard Miller (license 5685)
+ ........ Merged revisions 375272 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375273 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-18 21:44 +0000 [r375219-375247] Jonathan Rose <jrose at digium.com>
+
+ * UPGRADE.txt: app_queue: add upgrade notes for 375216 Adds UPGRADE
+ notes describing behavioral changes to rrmemory strategy caused
+ by 375216 (issue AST-989) Reported by: Thomas Arimont
+
+ * /, 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 ........ Merged
+ revisions 375217 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-18 20:02 +0000 [r375191] Richard Mudgett <rmudgett at digium.com>
+
+ * Makefile, /, build_tools/make_version, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
+ build_tools: Allow Asterisk to report git SHAs in version string.
+ Make git more attractive for managing work-in-progress.
+ Especially convenient when a potential patch set needs to be
+ tested on multiple platforms since one can use git to keep all
+ the test environments in sync independent of a subversion server.
+ Now the Asterisk version will show the exact git SHA5 that was
+ used when building (still appended by "M" if there are local
+ modifications) from a git clone of the Asterisk repository so the
+ developer can more easily know what is actually under test. You
+ will now get this: $ asterisk -V Asterisk GIT-1698298 Instead of
+ this: $ asterisk -V Asterisk UNKNOWN__and_probably_unsupported
+ This has zero impact for those not using git with the exception
+ of an extra test in the configure script to gather git's path.
+ This is necessary to prevent "sudo make install" from failing
+ since git may not be in the path in make's shell environment.
+ (closes issue ASTERISK-20483) Reported by: Shaun Ruffell Patches:
+ 0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch
+ (license #5417) patch uploaded by Shaun Ruffell Modified ........
+ Merged revisions 375189 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375190 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-17 19:00 +0000 [r375148] Kinsey Moore <kmoore at digium.com>
+
+ * main/tcptls.c, /: Ensure Asterisk fails TCP/TLS SIP calls when
+ certificate checking fails When placing a call to a TCP/TLS SIP
+ endpoint whose certificate is not signed by a configured CA
+ certificate, Asterisk would issue a warning and continue to
+ process the call as if there was not an issue with the
+ certificate. Asterisk now properly fails the call if the
+ certificate fails verification or if the certificate does not
+ exist when certificate checking is enabled (the default
+ behavior). (closes issue ASTERISK-20559) Reported by: kmoore
+ Review: https://reviewboard.asterisk.org/r/2163/ ........ Merged
+ revisions 375146 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375147 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-16 21:44 +0000 [r375079-375113] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: Fixes to the fd-oriented SIP TCP reads.
+ Don't crash on large user input. Allow SIP headers without space.
+ Optimize code a bit. Review:
+ https://reviewboard.asterisk.org/r/2162 ........ Merged revisions
+ 375111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 375112 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * /, channels/chan_sip.c: Update sip_request_call SIP dial string
+ documentation. This was missed when merging review r1859.
+ ........ Merged revisions 375074 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375078 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-16 14:08 +0000 [r375051] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: Remove a log message that was left in
+ accidentally from call-id logging development.
+
+2012-10-15 21:15 +0000 [r375027] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_dial.c, /, main/ccss.c, include/asterisk/strings.h,
+ channels/chan_iax2.c: Fix some potential misuses of ast_str in
+ the code. Passing an ast_str pointer by value that then calls
+ ast_str_set(), ast_str_set_va(), ast_str_append(), or
+ ast_str_append_va() can result in the pointer originally passed
+ by value being invalidated if the ast_str had to be reallocated.
+ This fixes places in the code that do this. Only the example in
+ ccss.c could result in pointer invalidation though since the
+ other cases use a stack-allocated ast_str and cannot be
+ reallocated. I've also updated the doxygen in strings.h to
+ include notes about potential misuse of the functions mentioned
+ previously. Review: https://reviewboard.asterisk.org/r/2161
+ ........ Merged revisions 375025 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 375026 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-15 08:11 +0000 [r375016] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c: Fix underscreen buttons warnings apeared
+ while transfer process
+
+2012-10-14 11:57 +0000 [r374995] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * config.guess, config.sub, /: Update config.guess and config.sub:
+ 2012-10-10 Update config.guess and config.sub to revision
+ fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the
+ savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM
+ 64bit). config.guess:timestamp='2012-09-25'
+ config.sub:timestamp='2012-10-10' ........ Merged revisions
+ 374977 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 374991 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-12 21:57 +0000 [r374932] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_voicemail.c: Avoid a segfault on invalid format names If
+ a format name was not found by ast_getformatbyname, a NULL
+ pointer would be passed into ast_format_rate and immediately
+ dereferenced. This ensures that a valid pointer is used since the
+ structure is already allocated on the stack. (closes issue
+ DPH-523) Reported-by: Steve Pitts
+
+2012-10-12 16:20 +0000 [r374914] Mark Michelson <mmichelson at digium.com>
+
+ * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
+ Do not use a FILE handle when doing SIP TCP reads. This is used
+ to solve an issue where a poll on a file descriptor does not
+ necessarily correspond to the readiness of a FILE handle to be
+ read. This change makes it so that for TCP connections, we do a
+ recv() on the file descriptor instead. Because TCP does not
+ guarantee that an entire message or even just one single message
+ will arrive during a read, a loop has been introduced to ensure
+ that we only attempt to handle a single message at a time. The
+ tcptls_session_instance structure has also had an overflow buffer
+ added to it so that if more than one TCP message arrives in one
+ go, there is a place to throw the excess. Huge thanks goes out to
+ Walter Doekes for doing extensive review on this change and
+ finding edge cases where code could fail. (closes issue
+ ASTERISK-20212) reported by Phil Ciccone Review:
+ https://reviewboard.asterisk.org/r/2123 ........ Merged revisions
+ 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ Merged revisions 374906 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-11 21:18 +0000 [r374850-374877] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_motif.c: Fix a bug where audio on Google Voice
+ would not work due to ignoring candidates. Instead of ignoring
+ parts of the message that are not known just ignore the ones we
+ know may be present and that would cause a problem.
+
+ * res/res_rtp_asterisk.c: Remove code that should not have gotten
+ in. (issue ASTERISK-20554)
+
+ * res/res_rtp_asterisk.c, channels/chan_motif.c: Fix an issue where
+ outgoing calls would fail to establish audio due to ICE
+ negotiation failures. This change removes the requirement for
+ ufrag and pwd in the transport stanza and also makes us the
+ controlling agent. (closes issue ASTERISK-20554) Reported by:
+ mmichelson
+
+2012-10-11 15:44 +0000 [r374845] 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
+ ........ Merged revisions 374844 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-11 15:31 +0000 [r374842] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c, include/asterisk/sip_api.h,
+ channels/chan_sip.exports.in (removed), main/sip_api.c (added):
+ Don't make chan_sip export global symbols. During testing, it was
+ discovered that having chan_sip export global symbols was
+ problematic. The biggest problem was that load order was
+ affected. Trying to use realtime could be problematic since in
+ all likelihood the necessary realtime driver(s) would not be
+ loaded before chan_sip. In addition, it was found that it was
+ impossible to use the Digium Phone Module for Asterisk since it
+ must be loaded before chan_sip since it must hook into chan_sip's
+ configuration parsing. The solution is to use a virtual table in
+ the same manner that other modules in Asterisk do, like
+ app_voicemail. (closes issue ASTERISK-20545) Reported by: kmoore
+
+2012-10-11 13:33 +0000 [r374833] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_motif.c: Consider the Google Talk content stanza
+ name (jin:content) valid.
+
+2012-10-10 21:03 +0000 [r374804] 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 ........ Merged
+ revisions 374803 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-10 13:35 +0000 [r374792] Kinsey Moore <kmoore at digium.com>
+
+ * main/manager.c: Fix segfault regression from r370681 Due to usage
+ of ast_hook_send_action, AMI action handling code should be able
+ to handle a NULL mansession->session. This would cause a crash on
+ NULL dereference if action_originate was called from
+ ast_hook_send_action. (closes issue ASTERISK-20544)
+
+2012-10-09 22:21 +0000 [r374771] Richard Mudgett <rmudgett at digium.com>
+
+ * main/pbx.c, /: Fix execution of 'i' extension due to
+ uninitialized variable. The fix for ASTERISK-18243 added code
+ that could potentially use dst_exten[] uninitialized. As a result
+ the 'i' exten may not be executed when it should. (closes issue
+ ASTERISK-20455) Reported by: Richard Miller Patches:
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