[svn-commits] lmadsen: tag 1.4.38-rc1 r295103 - /tags/1.4.38-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Nov 15 13:22:10 CST 2010
Author: lmadsen
Date: Mon Nov 15 13:22:06 2010
New Revision: 295103
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=295103
Log:
Importing files for 1.4.38-rc1 release.
Added:
tags/1.4.38-rc1/.lastclean (with props)
tags/1.4.38-rc1/.version (with props)
tags/1.4.38-rc1/ChangeLog (with props)
Added: tags/1.4.38-rc1/.lastclean
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--- tags/1.4.38-rc1/ChangeLog (added)
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+2010-11-15 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.38-rc1 Released.
+
+2010-11-15 18:05 +0000 [r295026-295031] Tilghman Lesher <tlesher at digium.com>
+
+ * tests/test_expr.c: Err, oops. Made it const to verify that it
+ wasn't altered, but forgot to revert before commit.
+
+ * tests/test_expr.c (added): Create test verifying results of
+ expression parser
+
+2010-11-12 20:49 +0000 [r294903] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Fix regression causing abort in voicemail
+ after opening a mailbox with no mesgs. In order to be more safe,
+ some error handling code was changed to respect more error
+ conditions including the potential memory allocation failure for
+ deleted and heard message tracking introduced in 293004. However,
+ last_message_index returns -1 for zero messages (perhaps as
+ expected) and was triggering the stricter error checking. Because
+ last_message_index is only called directly in one place, just
+ return 0 from open_mailbox (for file based storage) when no
+ messages are detected unless a real error has occurred. (closes
+ issue #18240) Reported by: leobrown Patches:
+ bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
+ Tested by: pabelanger
+
+2010-11-12 02:41 +0000 [r294821] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Asterisk is getting a "No D-channels
+ available!" warning message every 4 seconds. Asterisk is just
+ whining too much with this message: "No D-channels available!
+ Using Primary channel XXX as D-channel anyway!". Filtered the
+ message so it only comes out once if there is no D channel
+ available without an intervening D channel available period.
+ (closes issue #17270) Reported by: jmls
+
+2010-11-11 22:11 +0000 [r294641-294739] Jeff Peeler <jpeeler at digium.com>
+
+ * main/pbx.c: I didn't mean to merge this, sorry
+
+ * channels/chan_sip.c: Fix problem with qualify option packets for
+ realtime peers never stopping. The option packets not only never
+ stopped, but if a realtime peer was not in the peer list multiple
+ options dialogs could accumulate over time. This scenario has the
+ potential to progress to the point of saturating a link just from
+ options packets. The fix was to ensure that the poke scheduler
+ checks to see if a peer is in the peer list before continuing to
+ poke. The reason a peer must be in the peer list to be able to
+ properly manage an options dialog is because otherwise the call
+ pointer is lost when the peer is regenerated from the database,
+ which is how existing qualify dialogs are detected. (closes issue
+ #16382) Reported by: lftsy Patches: bug16382-3.patch uploaded by
+ jpeeler (license 325) Tested by: zerohalo
+
+ * main/pbx.c: One small addition to 294384 found while very
+ carefully merging to 1.6.
+
+2010-11-09 17:37 +0000 [r294384] Jeff Peeler <jpeeler at digium.com>
+
+ * main/asterisk.c, include/asterisk.h, main/pbx.c: Fix a deadlock
+ in device state change processing. Copied from some notes from
+ the original author (Russell): Deadlock scenario: Thread 1:
+ device state change thread Holds - rdlock on contexts Holds -
+ hints lock Waiting on channels container lock Thread 2: SIP
+ monitor thread Holds the "iflock" Holds a sip_pvt lock Holds
+ channel container lock Waiting for a channel lock Thread 3: A
+ channel thread (chan_local in this case) Holds 2 channel locks
+ acquired within app_dial Holds a 3rd channel lock it got inside
+ of chan_local Holds a local_pvt lock Waiting on a rdlock of the
+ contexts lock A bunch of other threads waiting on a wrlock of the
+ contexts lock To address this deadlock, some locking order rules
+ must be put in place and enforced. Existing relevant rules: 1)
+ channel lock before a pvt lock 2) contexts lock before hints lock
+ 3) channels container before a channel What's missing is some
+ enforcement of the order when you involve more than any two. To
+ fix this problem, I put in some code that ensures that (at least
+ in the code paths involved in this bug) the locks in (3) come
+ before the locks in (2). To change the operation of thread 1 to
+ comply, I converted the storage of hints to an astobj2 container.
+ This allows processing of hints without holding the hints
+ container lock. So, in the code path that led to thread 1's
+ state, it no longer holds either the contexts or hints lock while
+ it attempts to lock the channels container. (closes issue #18165)
+ Reported by: antonio ABE-2583
+
+2010-11-08 18:59 +0000 [r294163] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Modify our handling of 491 responses to drop
+ any pending reinvite retry scheduler entries if we get a new 491.
+ This prevents a scheduler entry from leaking if we receive a 491
+ response when one is pending. If a scheduler entry leaks, the pvt
+ it is associated my get destroyed before the scheduler entry
+ fires, and then memory corruption and crashes can occur when the
+ scheduled reinvite attempts to access and modify the memory of
+ the destroyed pvt. ABE-2543
+
+2010-11-05 00:02 +0000 [r293968] Shaun Ruffell <sruffell at digium.com>
+
+ * codecs/codec_dahdi.c: codecs/codec_dahdi: Prevent "choppy" audio
+ when receiving unexpected frame sizes. dahdi-linux 2.4.0
+ (specifically commit 9034) added the capability for the wctc4xxp
+ to return more than a single packet of data in response to a
+ read. However, when decoding packets, codec_dahdi was still
+ assuming that the default number of samples was in each read. In
+ other words, each packet your provider sent you, regardless of
+ size, would result in 20 ms of decoded data (30 ms if decoding
+ G723). If your provider was sending 60 ms packets then
+ codec_dahdi would end up stripping 40 ms of data from each
+ transcoded frame resulting in "choppy" audio. This would only
+ affect systems where G729 packets are arriving in sizes greater
+ than 20ms or G723 packets arriving in sizes greater than 30ms.
+ DAHDI-744.
+
+2010-11-04 21:28 +0000 [r293922] David Vossel <dvossel at digium.com>
+
+ * res/res_features.c: Fixes ringback tone on feature semi-attended
+ transfer ABE-2168
+
+2010-11-03 18:23 +0000 [r293805] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Party A in an analog 3-way call would
+ continue to hear ringback after party C answers. All parties are
+ analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A
+ flash hooks to bring C into 3-way call before C answers. (A and B
+ hear ringback) 4) C answers 5) A continues to hear ringback
+ during the 3-way call. (All parties can hear each other.) * Fixed
+ use of wrong variable in dahdi_bridge() that stopped ringback on
+ the wrong subchannel. * Made several debug messages have more
+ information. A similar issue happens if B and C are SIP channels.
+ B continues to hear ringback. For some reason this only affects
+ v1.8 and trunk. * Don't start ringback on the real and 3-way
+ subchannels when creating the 3-way conference. Removing this
+ code is benign on v1.6.2 and earlier.
+
+2010-11-02 23:02 +0000 [r293722] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: Add enabled/disabled information for
+ rtautoclear sip show settings output. When setting to zero/"no",
+ the numeric default was shown making it not obvious the disabled
+ setting was respected. (closes issue #18123) Reported by:
+ zerohalo
+
+2010-11-02 21:24 +0000 [r293639] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Make warning message have more useful
+ information in it. Change "Unable to get index, and nullok is not
+ asserted" to "Unable to get index for '<channel-name>' on channel
+ <number> (<function>(), line <number>)".
+
+2010-10-30 01:45 +0000 [r293339-293416] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Remove some more code that serves no
+ purpose.
+
+ * channels/chan_dahdi.c: Remove some code that serves no purpose.
+
+2010-10-28 19:44 +0000 [r293194] Tilghman Lesher <tlesher at digium.com>
+
+ * main/ast_expr2.c, main/ast_expr2.h, main/ast_expr2.y: "!00"
+ evaluated as false, which is incorrect. Fixing. Reported (though
+ the reporter did not understand he was reporting a bug) on the
+ asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+
+2010-10-25 22:55 +0000 [r293004] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Fix inprocess_container in voicemail to
+ correctly restrict max messages. The comparison function logic
+ was off, so the number of sessions for a given mailbox were not
+ being incremented properly. This problem caused the maximum
+ number of messages per folder to not be respected when
+ simultaneously leaving multiple voicemails just below the
+ threshold. These problems should be fixed by the above, but just
+ in case: Fixed resequence_mailbox to rely on the actual number of
+ detected number of files in a directory rather than just assuming
+ only 10 messages more than the maximum had been left. Also if
+ more messages than the maximum are deleted they are actually
+ removed now. The second purpose of this commit should have been
+ separated out probably, but is related to the above. Again, if
+ the number of messages in a given voicemail folder exceeds the
+ maximum set limit make sure to allocate enough space for the
+ deleted and heard index tracking array. A few random fixes: There
+ was a forgotten decrement of the inprocess count in
+ imap_store_file. When using IMAP storage, do not look in the
+ directory where file based storage messages may still reside and
+ influence the message count. Ensure to use only the first format
+ in sendmail. ABE-2516
+
+2010-10-25 19:05 +0000 [r292866] David Vossel <dvossel at digium.com>
+
+ * channels/chan_local.c: This patch turns chan_local pvts into
+ astobj2 objects. chan_local does some dangerous things involving
+ deadlock avoidance. tech_pvt functions like hangup and
+ queue_frame are provided with a locked channel upon entry. Those
+ functions are completely safe as long as you don't attempt to
+ give up that channel lock, but that is impossible to guarantee
+ due to the required deadlock avoidance necessary to lock both the
+ tech_pvt and both channels involved. In the past, we have tried
+ to account for this by doing things like setting a "glare" flag
+ that indicates what function should destroy the pvt. This was
+ used in local_hangup and local_queue_frame to decided who should
+ destroy the pvt if they collided in separate threads. I have
+ removed the need to do this by converting all chan_local
+ tech_pvts to astobj2. This means we can ref a pvt before deadlock
+ avoidance and not have to worry about that pvt possibly getting
+ destroyed under us. It also cleans up where we destroy the
+ tech_pvt. The only unlink from the tech_pvt container occurs in
+ local_hangup now, which is where it should occur. Since there
+ still may be thread collisions on some functions like
+ local_hangup after deadlock avoidance, I have added some checks
+ to detect those collisions and exit appropriately. I think this
+ patch is going to solve quite a bit of weirdness we have had with
+ local channels in the past.
+
+2010-10-21 00:00 +0000 [r292411] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * apps/app_dial.c: Record priv-recordintro as sln, not gsm This
+ removes the gsm->sln step when transcoding priv-recordintro.
+ (closes issue #18176) Reported by: pabelanger Patches:
+ chan_sip.diff uploaded by pabelanger (license 224)
+
+2010-10-18 21:50 +0000 [r292223] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Fix improper operator key acceptance and
+ clean up temp recording files. This is a fix for when pressing
+ the operator key after recording an unavailable, busy, name, or
+ temporary message in mailbox options. The operator key should not
+ be accepted here, but should be allowed during the message
+ recording. If the operator key is pressed during ensure the file
+ is saved or deleted as apporopriate. Also, ensure removal of
+ temporary recorded files after an early hang up or when message
+ acceptance confirmation times out. ABE-2518
+
+2010-10-18 21:47 +0000 [r292222] Leif Madsen <lmadsen at digium.com>
+
+ * sounds/sounds.xml, sounds/Makefile: Add support for the new
+ English (Australian Accent) sound files. (closes issue #17426)
+ Reported by: camsown Patches: core-sounds-en_AU.txt uploaded by
+ camsown (license 1050) add_AU_sounds.patch.txt uploaded by
+ lmadsen (license 10) Tested by: camsown, lmadsen, jtodd, qwell
+
+2010-10-15 19:30 +0000 [r291938] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * configs/gtalk.conf.sample: Clean up formatting.
+
+2010-10-15 02:13 +0000 [r291862] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_oss.c: Don't access o->next after freeing o on
+ unload
+
+2010-10-13 23:29 +0000 [r291643] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Deadlock between dahdi_exception() and
+ dahdi_indicate(). There is a deadlock between dahdi_exception()
+ and dahdi_indicate() for analog ports. The call-waiting and
+ three-way-calling feature can experience deadlock if these
+ features are trying to do something and an event from the bridged
+ channel happens at the same time. Deadlock avoidance code added
+ to obtain necessary channel locks before attemting an operation
+ with call-waiting and three-way-calling. (closes issue #16847)
+ Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ Review: https://reviewboard.asterisk.org/r/971/
+
+2010-10-13 22:45 +0000 [r291577] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c: Don't ignore frames that have been queued when
+ softhangup'd When an outgoing call is answered and hung up by the
+ far end *very* quickly, we may not read any frames and therefor
+ end up with a call that displays the wrong
+ disposition/DIALSTATUS. The reason is because ast_queue_hangup()
+ immediately sets the _softhangup flag on the channel and then
+ queues the HANGUP control frame, but __ast_read refuses to read
+ any frames if ast_check_hangup() indicates that a hangup request
+ has been made (which it will if _softhangup is set). So, we end
+ up losing control frames. This change makes __ast_read continue
+ to read frames even if a soft hangup has been requested. It
+ queues a hangup frame to make sure that __ast_read() will still
+ eventually return NULL. Much thanks to David Vossel for all of
+ the reviews, discussion, and help! (closes issue #16946) Reported
+ by: davidw Review: https://reviewboard.asterisk.org/r/740/
+
+2010-10-13 15:23 +0000 [r291392] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Lock pvt so pvt->owner can't disappear when
+ queueing up a frame. This fixes a crash due to a hangup race
+ condition. ABE-2601
+
+2010-10-12 16:55 +0000 [r291263] Tilghman Lesher <tlesher at digium.com>
+
+ * main/acl.c: Oops, incorrect range (although unallocated at ARIN)
+
+2010-10-11 18:29 +0000 [r291109] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Add missing unlock to an exception condition
+ in reload_config().
+
+2010-10-08 02:35 +0000 [r290862] Jeff Peeler <jpeeler at digium.com>
+
+ * main/asterisk.c: Ensure editline cleanup occurs when Ctrl-C is
+ pressed at control console. A recent change was made to avoid a
+ race condition on shutdown which only called the end functions
+ from the console thread. However, when pressing Ctrl-C the quit
+ handler is called from the signal handler thread. (closes issue
+ #17698) Reported by: jmls
+
+2010-10-07 20:56 +0000 [r290750] Jason Parker <jparker at digium.com>
+
+ * autoconf/ast_ext_lib.m4, configure,
+ include/asterisk/autoconfig.h.in: Allow PRI to build properly
+ when using --with-pri. Use the directories found for the parent
+ when using lib dependencies. (closes issue #17314) Reported by:
+ tzafrir Patches: 17314-withdeps.diff uploaded by qwell (license
+ 4)
+
+2010-10-05 20:20 +0000 [r290392] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_jabber.c: Fix a crash by ensuring that we don't alter
+ memory after it's freed. (closes issue #17387) Reported by: jmls
+ Patches: 20100726__issue17387.diff.txt uploaded by tilghman
+ (license 14) Tested by: jmls
+
+2010-10-05 17:41 +0000 [r290323] Richard Mudgett <rmudgett at digium.com>
+
+ * contrib/valgrind.supp: Merged revision 258974 from
+ https://origsvn.digium.com/svn/asterisk/trunk .......... r258974
+ | diruggles | 2010-04-26 14:05:47 -0500 (Mon, 26 Apr 2010) | 4
+ lines Line 24 missed in compatibility fix in revision 233577
+ added a "fun:" prefix line 24 ..........
+
+2010-10-04 20:15 +0000 [r290100-290177] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, configure.ac: Fixing Mac OS X auto-builder.
+
+ * configure, configure.ac: Automatically re-run configure test for
+ menuselect, when the relevant makeopts settings change.
+
+2010-10-02 08:50 +0000 [r289949] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c: Add documentation for undocumented option to AMI
+ action originate
+
+2010-10-02 04:42 +0000 [r289873] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: When forwarding a message, a prepend means
+ that the filesystem will always have a better copy. (closes issue
+ #17803) Reported by: dpetersen Patches:
+ 20100923__issue17803.diff.txt uploaded by tilghman (license 14)
+ Tested by: dpetersen
+
+2010-10-01 22:58 +0000 [r289797] Jeff Peeler <jpeeler at digium.com>
+
+ * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Change
+ RFC2833 DTMF event duration on end to report actual elapsed time.
+ The scenario here is with a non P2P early media session. The
+ reported time length of DTMF presses are coming up short when
+ sending to the remote side. Currently the event duration is a
+ running total that is incremented when sending continuation
+ packets. These continuation packets are only triggered upon
+ incoming media from the remote side, which means that the running
+ total probably is not going to end up matching the actual length
+ of time Asterisk received DTMF. This patch changes the end event
+ duration to be lengthened if it is detected that the end event is
+ going to come up short. Review:
+ https://reviewboard.asterisk.org/r/957/ ABE-2476
+
+2010-10-01 17:03 +0000 [r289703] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * configs/jabber.conf.sample, res/res_jabber.c: Disable debugging
+ by default and reformat .config file. Review:
+ https://reviewboard.asterisk.org/r/929/
+
+2010-10-01 16:20 +0000 [r289699] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: Ensure user portion of SIP URI matches
+ dialplan when using encoded characters. This commit takes a
+ simliar approach to 288112 and checks the dialplan to determine
+ the proper action for an incoming contact header as to whether or
+ not it should be decoded or not. sip_new was blindly always
+ decoding the extension, which also caused the outgoing contact
+ header to be incorrect as well as failing to match the encoded
+ extension in the dialplan. (closes issue #17892) Reported by:
+ wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license
+ 325) Tested by: wdoekes
+
+2010-10-01 09:42 +0000 [r289622] schmitds <schmitds at localhost>:
+
+ * channels/chan_sip.c: don't iterate through all dialogs to find
+ and delete old subscribes On every incoming subscribe there is a
+ iteration through all dialogs to find old subscribes and delete
+ them. This is slow and not RFC conform. This was only needed in
+ 1.2 cause a subscribe was not deleted when a dialog was
+ destroyed, after 1.4 a subscribe get removed when its dialog is
+ destroyed. (closes issue #17950) Reported by: schmidts Tested by:
+ schmidts Review: https://reviewboard.asterisk.org/r/901/
+
+2010-09-30 17:08 +0000 [r289500] Brett Bryant <bbryant at digium.com>
+
+ * res/res_agi.c: res_agi.c:handle_getvariablefull() could
+ recursively lock a channel and not release it if an argument is
+ the current channel's name. (closes issue #17970) Reported by:
+ mdu113 Patches: res_agi.c.diff3 uploaded by mdu113 (license 582)
+ Tested by: mdu113 Review: https://reviewboard.asterisk.org/r/947/
+
+2010-09-30 15:34 +0000 [r289424] Russell Bryant <russell at digium.com>
+
+ * apps/app_sms.c: Fix a crash in app_sms. Since the data being
+ passed to the generator callback is on the stack of the SMS()
+ application, we must ensure that the generator is stopped before
+ the application exits. ABE-2587
+
+2010-09-29 20:56 +0000 [r289338] Jason Parker <jparker at digium.com>
+
+ * main/channel.c, res/res_features.c: Allow a manager originate to
+ succeed on forwarded devices. The timeout to wait for an answer
+ was being set to 0 when a device forwarded to another extension.
+ We don't always need the timeout set like this, so make it an
+ optional parameter, and don't use it in this case. ABE-2544
+
+2010-09-29 15:03 +0000 [r289177] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c: Set the caller id on CDRs when it is set on the
+ parent channel. (closes issue #17569) Reported by: tbelder
+ Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
+ tbelder
+
+2010-09-28 18:10 +0000 [r289094] Brett Bryant <bbryant at digium.com>
+
+ * main/channel.c: Fixes an issue with the Newchannel AMI event
+ during the Masquerading process. Fixes an issue with the
+ Newchannel AMI event during the Masquerading process, where no
+ Newchannel AMI event was generated for the psuedo channel used
+ during the masquerading process. (closes issue #17987) Reported
+ by: RadicAlish Patches: newchannel.patch.txt uploaded by
+ RadicAlish (license 1122) Tested by: RadicAlish Review:
+ https://reviewboard.asterisk.org/r/937/
+
+2010-09-24 15:26 +0000 [r288746] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c: Don't fail a masquerade if it is already
+ being hung up This avoids noise on some Local channel situations
+ where we don't use /n. Thanks to Alec Davis for the suggestion.
+
+2010-09-24 03:20 +0000 [r288636] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.exports, include/asterisk/compat.h,
+ main/strcompat.c, include/asterisk/channel.h: Solaris
+ compatibility fixes
+
+2010-09-22 23:00 +0000 [r288499] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c: Don't let a Local channel get bridged to
+ itself If a local channel gets bridged to itself, it becomes
+ orphaned with no devices left to actually tell it to hang up.
+ This patch modifies local_fixup() to detect this case and deny
+ it. Review: https://reviewboard.asterisk.org/r/934
+
+2010-09-22 17:48 +0000 [r288416] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: RFC3261 section 12.2 explicitly says out of
+ order requests are responded with a 500 Server Internal Error
+ response. ABE-2458
+
+2010-09-22 17:39 +0000 [r288412] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Handle media specific T.38 SDP information
+ (closes issue #16647) Reported by: kwemheuer
+
+2010-09-22 16:49 +0000 [r288343] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: During check_pendings, if the dialog is
+ terminated with a CANCEL, change the invitestate to INV_CANCEL
+ like in sip_hangup.
+
+2010-09-22 16:39 +0000 [r288339] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c: Fix a 100% CPU consumption problem when setting
+ console=yes in asterisk.conf. The handling of -c and console=yes
+ should be the same, but they were not. When you specify -c, it
+ sets both a flag for console module and for asterisk not to
+ fork() off into the background. The handling of console=yes only
+ set console mode, so you would end up with a background process()
+ trying to run the Asterisk console and freaking out since it
+ didn't have anything to read input from. Thanks to beagles for
+ reporting and helping debug the problem!
+
+2010-09-22 15:04 +0000 [r288265-288266] Tilghman Lesher <tlesher at digium.com>
+
+ * UPGRADE.txt: Document addition of encoding parameter. (issue
+ #16940) Reported by: jamicque
+
+ * configs/cdr_pgsql.conf.sample, cdr/cdr_pgsql.c: Allow the
+ encoding to be set, in case local charset does not agree with
+ database. (closes issue #16940) Reported by: jamicque Patches:
+ 20100827__issue16940.diff.txt uploaded by tilghman (license 14)
+ 20100921__issue16940__1.6.2.diff.txt uploaded by tilghman
+ (license 14) Tested by: jamicque
+
+2010-09-21 23:55 +0000 [r288192] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c: In chan_iax2.c:schedule_delivery() calls
+ ast_bridged_channel() on an unlocked channel. Near the beginning
+ of schedule_delivery(), ast_bridged_channel() is called on
+ iaxs[fr->callno]->owner. However, the channel is not locked,
+ which can result in ast_bridged_channel() crashing should
+ owner->tech change to a technology that doesn't implement
+ bridged_channel. I also fixed the other calls to
+ ast_bridged_channel() in chan_iax2.c since the owner lock was not
+ held there either. Converted the existing channel deadlock
+ avoidance to use iax2_lock_owner(). Using the new function
+ simplified some awkward code. In the process of fixing the
+ locking on ast_bridged_channel(), I also found a memory leak in
+ socket_process() for v1.6.2 and v1.8. The local struct variable
+ ies.vars is not freed on early/abnormal function exits. (closes
+ issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
+ uploaded by rmudgett (license 664) Review:
+ https://reviewboard.asterisk.org/r/926/
+
+2010-09-21 22:17 +0000 [r288112-288116] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/stringfields.h: Underquoted definition of
+ stringfield macro.
+
+ * channels/chan_sip.c: Try both the encoded and unencoded
+ subscription URI for a match in hints. When a phone sends an
+ encoded URI for a subscription, the URI is not matched with the
+ actual hint that is in decoded format. For example, if we have an
+ extension with a hint that is named: "#5601" or "*5601", the
+ subscription will work fine if the phone subscribes with an
+ already decoded URI, but when it's decoded like "%255601" or
+ "%2A5601", Asterisk is unable to match it with the correct hint.
+ (closes issue #17785) Reported by: ramonpeek Patches:
+ 20100831__issue17785.diff.txt uploaded by tilghman (license 14)
+ Tested by: ramonpeek
+
+2010-09-21 19:43 +0000 [r288005] Brett Bryant <bbryant at digium.com>
+
+ * main/channel.c: Add a check to fix a rare segmentation fault
+ you'd get if ast_frdup couldn't allocate memory on the first
+ frame being queued in ast_queue_frame. (closes issue #17882)
+ Reported by: seanbright Tested by: seanbright
+
+2010-09-21 19:07 +0000 [r287933] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Less than zero is an error, not any non-zero
+ value.
+
+2010-09-20 23:57 +0000 [r287758] Brett Bryant <bbryant at digium.com>
+
+ * apps/app_meetme.c: Fix misvalidation of meetme pins in
+ conjunction with the 'a' MeetMe flag. When using the 'a' MeetMe
+ flag and having a user and admin pin setup for your conference,
+ using the user pin would gain you admin priviledges. Also, when
+ no user pin was set, an admin pin was, the 'a' MeetMe flag wasn't
+ used, and the user tried to enter a conference then they were
+ still prompted for a pin and forced to hit #. (closes issue
+ #17908) Reported by: kuj Patches: pins_2.patch uploaded by kuj
+ (license 1111) Tested by: kuj Review: [full review board URL with
+ trailing slash]
+
+2010-09-20 23:15 +0000 [r287682-287684] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/channel.c: ast_channel_masquerade: remove extra else if
+ (closes issue #17363,#16057) Reported by:
+ amorsen;davidw,alecdavis Patches: based on bug16057.diff4.txt
+ uploaded by alecdavis (license 585) Tested by: ramonpeek, davidw,
+ alecdavis
+
+ * main/channel.c: ast_channel_masquerade: Avoid recursive
+ masquerades. Check all 4 combinations of (original/clonechan) *
+ (masq/masqr). Initially original->masq and clonechan->masqr were
+ only checked. It's possible with multiple masq's planned - and
+ not yet executed, that the 'original' chan could already have
+ another masq'd into it - thus original->masqr would be set, that
+ masqr would lost. Likewise for the clonechan->masq. (closes issue
+ #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
+ bug16057.diff4.txt uploaded by alecdavis (license 585) Tested by:
+ ramonpeek, davidw, alecdavis
+
+2010-11-02 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.37 Released.
+
+2010-09-20 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.37-rc1 Released.
+
+2010-09-20 15:48 +0000 [r287555] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c: Use ast_dynamic_str when processing hint state
+ changes (related to issue #17928) Reported by: mdu113
+
+2010-09-19 15:56 +0000 [r287469] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c: Make sure we always free variables properly in
+ manager originate. (closes issue #17891) reported, solved and
+ tested by oej Review: https://reviewboard.asterisk.org/r/869/
+
+2010-09-17 21:06 +0000 [r287386] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: Blank columns should get set on reload, not
+ ignored. (closes issue #16893) Reported by: haakon Patches:
+ 20100818__issue16893.diff.txt uploaded by tilghman (license 14)
+
+2010-09-17 13:34 +0000 [r287307] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c: Use ast_strdup() instead of ast_strdupa() while
+ processing in ast_hint_state_changed(). (related to issue #17928)
+ Reported by: mdu113
+
+2010-09-16 22:12 +0000 [r287197] Jason Parker <jparker at digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Add LSB headers for Debian
+ init script, since Debian will complain if it isn't there.
+ Headers were taken from trunk. (closes issue #17958) Reported by:
+ javyer
+
+2010-09-16 20:04 +0000 [r287114-287118] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c: Don't limit hint processing in
+ ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
+ (closes issue #17928) Reported by: mdu113 Patches:
+ 20100831__issue17928.diff.txt uploaded by tilghman (license 14)
+ Tested by: mdu113
+
+ * main/cdr.c: Don't stop printing cdr variables if we encounter one
+ with a blank name or value. (closes issue #17900) Reported by:
+ under Patches: core-show-channel-cdr-fix1.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson
+
+2010-09-15 20:20 +0000 [r286941-286956] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: whitespace fix
+
+ * apps/app_voicemail.c: Ensure mailbox is not filled to capacity
+ before doing message forwarding. Specifically, before prompting
+ to record a prepended message the capacity is checked first. If
+ the mailbox is full the extension will be reprompted. ABE-2517
+
+2010-09-14 19:26 +0000 [r286679-286756] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Don't clear the username from a realtime
+ database when a registration expires. Non-realtime chan_sip does
+ not clear the username from memory when a registration expiries
+ so realtime probably shouldn't either. (closes issue #17551)
+ Reported by: ricardolandim Patches:
+ reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
+ 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
+ (license 96) reg-expiry-username-1.8-fix1.diff uploaded by
+ mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
+ uploaded by mnicholson (license 96) Tested by: ricardolandim,
+ mnicholson
+
+ * main/channel.c: Only drop duplicate answer frames if the channel
+ is bridged. Back in r3710 ast_read() was modified to drop answer
+ frames on channels that were in the UP state. This modification
+ prevented bridges that were up before the answer from being
+ broken and reestablished by an ANSWER control frame. That change
+ also prevents pickup of channels called from the ast_dial
+ framework from working properly. The ast_dial framework expects
+ to see an ANSWER frame after dialing and the pickup code queues
+ one but ast_read() drops it. This new change only drops ANSWER
+ frames when the channel is bridged, allowing the answer queued by
+ the pickup code to properly pass through ast_read() on to the
+ ast_dial framework. ABE-2473 (related to issue #2342)
+
+2010-09-13 15:12 +0000 [r286381] Jason Parker <jparker at digium.com>
+
+ * tests: Add stuff to svn:ignore for tests/ directory. (closes
+ issue #17983) Reported by: oej
+
+2010-09-11 16:59 +0000 [r286267] Olle Johansson <oej at edvina.net>
+
+ * main/file.c: Handle error response when we can't make file
+ compatible Review: https://reviewboard.asterisk.org/r/911/
+
+2010-09-10 22:54 +0000 [r286222] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c: Return -1 if chan_local doesn't support an
+ option
+
+2010-09-10 20:35 +0000 [r286114] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_iax2.c: Load iax.conf before registering any
+ functions/applications/actions. Review:
+ https://reviewboard.asterisk.org/r/914/
+
+2010-09-10 20:33 +0000 [r286113] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: An outgoing call may not get hung up if a
+ pre-connect incoming ISDN call is disconnected. If the ISDN link
+ a pre-connect incoming call is using fails or is reset, the
+ outgoing leg may not hang up or be delayed in hanging up.
+ (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
+ PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
+ PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
+ incoming call leg hangs up before connecting for any reason. It
+ makes no sense to send a BUSY or CONGESTION control frame to the
+ outgoing call leg under these circumstances.
+
+2010-09-10 20:03 +0000 [r286070] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Fixes sip extension state update DEADLOCK
+ PROBLEM: In chan_sip, and all the other channel drivers, it is
+ common for us to hold the tech_pvt lock while we ask the Asterisk
+ core about an extension and context. Every time we do this the
+ locking order becomes, (1. tech_pvt lock ---> 2. global context
+ lock). In chan_sip when a dialog subscribes to a hint, that
+ locking order is reversed in the extensionstate callback which
+ will occur outside of the channel_driver's monitor loop. So, on
+ an extension state update we have (1. global context lock ---->
+ 2. tech_pvt lock). Typically when we have to do a reversed
+ locking order like this we'd just do some sort of deadlock
+ avoidance to fix the problem... That will not work here. There
+ are more locks involved here than just the context and tech_pvt.
+ Those are the two that are colliding, but it is impossible to
+ give up the context lock because the global hints list lock MUST
+ be held as well and we can not give that lock up during the
+ extensionstate callback traversal... The locking order for the
+ context and hints are (1. global context lock ----> 2. hints list
+ lock). Deadlock avoidance is not an option here. SOLUTION: The
+ solution this patch implements is to queue the extension state
+ updates into a list and send the NOTIFY messages out during the
+ do_monitor pvt traversal. This clears out the problem of having
+ to hold the context lock before the tech_pvt lock entirely.
+ (closes issue #17888) Reported by: zerohalo
+
+2010-09-10 19:25 +0000 [r286059] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c, funcs/func_channel.c,
+ include/asterisk/channel.h, include/asterisk/pbx.h,
+ include/asterisk/frame.h: Inherit CHANNEL() writes to both sides
+ of a Local channel Having Local (/n) channels as queue members
+ and setting the language in the extension with
+ Set(CHANNEL(language)=fr) sets the language on the Local/...,2
+ channel. Hold time report playbacks happen on the Local/...,1
+ channel and therefor do not play in the specified language. This
+ patch modifies func_channel_write to call the setoption callback
+ and pass the CHANNEL() write info to the callback. chan_local
+ uses this information to look up the other side of the channel
+ and apply the same changes to it. (closes issue #17673) Reported
+ by: Guggemand Review: https://reviewboard.asterisk.org/r/903/
+
+2010-09-10 18:22 +0000 [r285889-286023] Tilghman Lesher <tlesher at digium.com>
+
+ * main/test.c: Missing newline
+
+ * include/asterisk/select.h, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ tests/test_poll.c: Fix Mac OS X build. This also fixes a rather
+ grievous calculation error for the offset of ast_fdset, which was
+ masked on Linux and FreeBSD, because these platforms check the
[... 29842 lines stripped ...]
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