[asterisk-commits] lmadsen: tag 1.4.42 r325413 - in /tags/1.4.42: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jun 28 16:49:23 CDT 2011
Author: lmadsen
Date: Tue Jun 28 16:49:19 2011
New Revision: 325413
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=325413
Log:
Importing files for 1.4.42 release.
Added:
tags/1.4.42/.lastclean (with props)
tags/1.4.42/.version (with props)
tags/1.4.42/ChangeLog (with props)
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+2011-06-28 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.42 Released.
+
+2011-06-28 20:03 +0000 [r325275] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Don't leak SIP username information
+
+2011-06-23 18:16 +0000 [r324627] David Vossel <dvossel at digium.com>
+
+ * res/res_features.c, channels/chan_iax2.c: Addresses AST-2011-010,
+ remote crash in IAX2 driver Thanks to twilson for identifying the
+ issue and providing the patches. AST-2011-010
+
+2011-06-15 18:06 +0000 [r323732] Terry Wilson <twilson at digium.com>
+
+ * res/res_features.c: Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were
+ broken by a recent DTMF change. This patch makes sure that
+ dynamic features are also checked when deciding whether or not to
+ pass DTMF through or store it for interpreting. (closes issue
+ ASTERISK-17914) Reported by: vrban
+
+2011-06-15 15:15 +0000 [r323559] Sean Bright <sean at malleable.com>
+
+ * main/manager.c: Resolve a segfault/bus error when we try to map
+ memory that falls on a page boundary. The fix for ASTERISK-15359
+ was incorrect in that it added 1 to the length of the mmap'd
+ region. The problem with this is that reading/writing to that
+ extra byte outside of the bounds of the underlying fd causes a
+ bus error. The real issue is that we are working with both a FILE
+ * and the raw fd underneath it and not synchronizing between
+ them. The code that was removed in ASTERISK-15359 was correct,
+ but we weren't flushing the FILE * before mapping the fd. Looking
+ at the manager code in 1.4 reveals that the FILE * in 'struct
+ mansession' is never used except to create a temporary file that
+ we immediately fdopen. This means we just need to write a 0 byte
+ to the fd and everything will just work. The other branches
+ require a call to fflush() which, while not a guaranteed fix,
+ should reduce the likelihood of a crash. This all makes sense in
+ my head. (closes issue ASTERISK-16460) Reported by:
+ Ravelomanantsoa Hoby (hoby) Patches:
+ issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license
+ #5060)
+
+2011-06-09 15:36 +0000 [r322646-322698] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: unlock pvt when we drop voice frames
+ received in early media when in t.38 mode
+
+ * channels/chan_sip.c: whitespace
+
+ * channels/chan_sip.c: don't drop any voice frames when checking
+ for T.38 during early media (closes issue ASTERISK-17705) Review:
+ https://reviewboard.asterisk.org/r/1186/ patch by oej reported by
+ oej
+
+2011-05-21 05:09 +0000 [r320393] Paul Belanger <pabelanger at digium.com>
+
+ * cdr/cdr_pgsql.c: Solaris compatibility fixes
+
+2011-05-20 20:38 +0000 [r320235] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_meetme.c: The meetme CLI command completion leaves
+ conferences mutex locked. When issuing a meetme kick CLI command
+ and an invalid (non-existent) conference number is specified,
+ pressing Tab leaves the conferences mutex locked and, therefore,
+ all conferences deadlock. Add missing unlock. (closes issue
+ #19336) Reported by: zvision Patches: app_meetme.diff uploaded by
+ zvision (license 798)
+
+2011-05-20 16:38 +0000 [r320055] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Destroy variables on a sip_pvt
+ before copying vars from the sip_peer. (closes issue #19202)
+ Reported by: wdoekes Patches:
+ issue19202_destroy_challenged_invite_chanvars.patch uploaded by
+ wdoekes (license 717)
+
+2011-05-18 23:04 +0000 [r319527-319652] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Make sure everyone gets an unhold when a
+ transfer succeeds Some phones, like the Snom phones, send a hold
+ to the transfer target after before sending the REFER. We need to
+ make sure that we unhold the parties that are being connected
+ after the masquerade. If Local channels with the /nm option are
+ used when dialing the parties, hold music would still be playing
+ on the transfer target, even after being connected with the
+ transferee.
+
+ * apps/app_dial.c: Fix app_dial ring groups Revert part of r315643.
+ We need to remove the datastore here as well. The code in
+ bridging code will catch anything that app_dial might miss.
+ (closes issue #19311) Reported by: mspuhler Patches:
+ issue_19311_no_answer.diff uploaded by elguero (license 37)
+
+2011-05-13 01:09 +0000 [r318734] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c, res/res_features.c,
+ apps/app_directed_pickup.c: Merged revisions 318671 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8 * The
+ applicable fixes for v1.4 are the SIP deadlock and the in
+ progress masquerade check for multiple parties trying to pickup
+ the same call. issue18654_v1.4.patch uploaded by rmudgett
+ (license 664) * Backported to v1.6.2. issue18654_v1.6.2.patch
+ uploaded by rmudgett (license 664) ........ r318671 | alecdavis |
+ 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix
+ directed group pickup feature code *8 with pickupsounds enabled
+ Since 1.6.2, the new pickupsound and pickupfailsound in
+ features.conf cause many issues. 1).
+ chan_sip:handle_request_invite() shouldn't be playing out the
+ fail/success audio, as it has 'netlock' locked. 2). dialplan
+ applications for directed_pickups shouldn't beep. 3). feature
+ code for directed pickup should beep on success/failure if
+ configured. Created a sip_pickup() thread to handle the pickup
+ and playout the audio, spawned from handle_request_invite. Moved
+ app_directed:pickup_do() to features:ast_do_pickup(). Functions
+ below, all now use the new ast_do_pickup() app_directed_pickup.c:
+ pickup_by_channel() pickup_by_exten() pickup_by_mark()
+ pickup_by_part() features.c: ast_pickup_call() (closes issue
+ #18654) Reported by: Docent Patches:
+ ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license
+ 585) Tested by: lmadsen, francesco_r, amilcar, isis242,
+ alecdavis, irroot, rymkus, loloski, rmudgett Review:
+ https://reviewboard.asterisk.org/r/1185/ ........
+
+2011-05-06 17:59 +0000 [r317719] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Regression after r297603 (Improve handling
+ of REGISTER requests with multiple contact headers.)
+ Uninitialized variable. (issue #18640) (closes issue #18785)
+ Reported by: pnlarsson Patches: issue18785_enegaard.patch
+ uploaded by enegaard (license 1197)
+
+2011-05-06 07:55 +0000 [r317574] Terry Wilson <twilson at digium.com>
+
+ * apps/app_queue.c: Re-fix queue round-robin This part of the
+ change for r315596 was incorrect. No bridge occurs when doing a
+ roundrobin dial and no one answers, so this code shouldn't have
+ been removed.
+
+2011-05-05 18:20 +0000 [r317211] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: chan_sip: fix broken realtime peer count,
+ fix memory leak This patch addresses two bugs in chan_sip: 1) The
+ count of realtime peers and users was off. The increment checked
+ the value of the caching option, while the decrement did not. 2)
+ Add a missing regfree() for a regex. (closes issue #19108)
+ Reported by: vrban Patches: missing_regfree.patch uploaded by
+ vrban (license 756) sip_object_counter.patch uploaded by vrban
+ (license 756)
+
+2011-05-05 14:54 +0000 [r317102] Leif Madsen <lmadsen at digium.com>
+
+ * contrib/scripts/safe_asterisk: Disable console colourization
+ inside safe_asterisk checks. (closes issue #19213) Reported by:
+ lefoyer Patches:
+ issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by
+ wdoekes (license 717) Tested by: wdoekes, lefoyer
+
+2011-05-04 16:08 +0000 [r316707] Sean Bright <sean at malleable.com>
+
+ * apps/app_voicemail.c: If sox fails when processing a voicemail,
+ don't delete the original file. (closes issue #18111) Reported
+ by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright
+ (license 71) Tested by: seanbright
+
+2011-05-03 21:27 +0000 [r316328] David Vossel <dvossel at digium.com>
+
+ * channels/chan_local.c: Fixes chan_local crashs in local_fixup()
+ Thanks OEJ for tracking down the issue and submitting the patch.
+ (closes issue #19053) Reported by: oej Tested by: oej Review:
+ https://reviewboard.asterisk.org/r/1158/
+
+2011-05-02 18:25 +0000 [r316089] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * configure, configure.ac: Breakage from slightly before the
+ outage; would have fixed sooner but for the outage.
+
+2011-04-27 21:20 +0000 [r316006] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Backport the use of curl from 1.6.2 to make the 1.4 target work
+ on Bamboo.
+
+2011-04-27 20:54 +0000 [r315989] Sean Bright <sean at malleable.com>
+
+ * channels/chan_sip.c: Partial revert of r315671 which removed a
+ logging statement and not a manager event. Reported by ibercom in
+ #asterisk-bugs. (issue #16033)
+
+2011-04-27 18:57 +0000 [r315891] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Fix our compliance with RFC 3261 section
+ 18.2.2. This change optimizes the free_via() function and removes
+ some redundant null checking. It also fixes compliance with RFC
+ 3261 section 18.2.2 by always using the port specified in the Via
+ header for routing responses (even when maddr is not set). Also
+ the htons() function is now used when setting the port.
+ Additional documentation comments have been added in various
+ places to make the logic in the code clearer. (closes issue
+ #18951) Reported by: jmls Patches:
+ issue18951_set_proper_port_from_via.patch uploaded by wdoekes
+ (license 717) (modified)
+
+2011-04-26 22:47 +0000 [r315596-315671] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Make sure unregistering a peer unlinks it
+ from the peer container Instead of mostly copying the code from
+ expire_register, just use the function that "does the right
+ thing". (closes issue #16033) Reported by: kkm Patches:
+ 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
+ Tested by: kkm, tilghman, twilson
+
+ * apps/app_dial.c, res/res_features.c, apps/app_queue.c: Allow
+ transfer loops without allowing forwarding loops We try to avoid
+ the situation where two phones may be forwarded to each other
+ causing an infinite loop by storing each dialed interface in a
+ channel datastore and checking the list before dialing out. This
+ works, but currently breaks situations like A calls B, A
+ transfers B to C, B transfers C to A, and A transfers C to B.
+ Since human interaction is happening here and not an automated
+ forwarding loop, it should be allowed. This patch removes the
+ dialed_interfaces datastore when a call is bridged (a suggestion
+ from the brilliant mmichelson). If a call is being bridged, it
+ should be safe to assume that we aren't stuck in a loop. Since we
+ are now handling this is the bridge code, the previous attempts
+ at handling it in app_dial and app_queue are removed. Review:
+ https://reviewboard.asterisk.org/r/1195/
+
+2011-04-26 19:18 +0000 [r315501] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * include/asterisk/select.h: Fix the bounds-checking code. The code
+ that set the bit within the select bitfield was correct, but the
+ bounds-checking code was not. The change to that line uses the
+ new _bitsize macro for clarity. Also, FD_ZERO macro did not
+ zero-out anything but the first word of the bitfield, so this
+ could have caused problems with modules using that macro with the
+ expanded bitfield. (closes issue #18773) Reported by: jamicque
+ Patches: 20110423__issue18773.diff.txt uploaded by tilghman
+ (license 14) Tested by: chris-mac
+
+2011-04-25 19:28 +0000 [r315257] Russell Bryant <russell at digium.com>
+
+ * formats/format_wav.c: Be more flexible with unknown chunks in wav
+ files. This patch makes format_wav ignore unknown chunks instead
+ of erroring out on them. (closes issue #18306) Reported by:
+ jhirsch Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch
+ (license 1156)
+
+2011-04-25 16:14 +0000 [r315147] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/manager.c: Reverted part of r314607, as it can introduce a
+ regression. Specifically, the security check for the "system"
+ privilege was removed. If a user had the "call" privilege but not
+ the "system" privilege, they would loose the ability to execute
+ the system app and dialplan functions that run commands in a
+ shell. This branch never used the "system" privilege for that
+ purpose and did not need to be patched. AST-2011-006 (related to
+ issue 0018787) Reported by: kobaz
+
+2011-04-25 07:06 +0000 [r315051] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/chan_local.c: chan_local:check_bridge() misplaced
+ misplaced ast_mutex_unlock if !p->chan->_bridge->_softhangup path
+ isn't followed, brigde remains locked. (closes issue #19176)
+ Reported by: alecdavis Patches: bug19176.diff.txt uploaded by
+ alecdavis (license 585)
+
+2011-04-22 20:01 +0000 [r314908] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_agent.c: Prevent the login thread and the app
+ threads from using the asterisk channel at the same time.
+ ABE-2756
+
+2011-04-22 14:34 +0000 [r314822] Russell Bryant <russell at digium.com>
+
+ * res/res_agi.c: Initialize buffers in getvar and getvarfull.
+ Initialize the buffers used to hold the result from GET VARIABLE
+ or GET VARIABLE FULL. The bug report shows func_read returning
+ garbage in the result. It assumed that the buffer passed in was
+ initialized, like many other functions do. In the more common
+ code path (through the dialplan), it is initialized, so just
+ initialize it here too. (closes issue #19050) Reported by: johnz
+
+2011-04-21 18:19 +0000 [r314607] Matthew Nicholson <mnicholson at digium.com>
+
+ * configs/http.conf.sample, main/manager.c, channels/chan_skinny.c,
+ main/http.c, configs/skinny.conf.sample: Added limits to the
+ number of unauthenticated sessions TCP based protocols are
+ allowed to have open simultaneously. Also added timeouts for
+ unauthenticated sessions where it made sense to do so. Unrelated,
+ the manager interface now properly checks if the user has the
+ "system" privilege before executing shell commands via the
+ Originate action. AST-2011-005 AST-2011-006 (closes issue #18787)
+ Reported by: kobaz (related to issue #18996) Reported by: tzafrir
+
+2011-04-19 18:37 +0000 [r314300] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * apps/app_voicemail.c: app_voicemail: Fix ODBC Storage compile
+ regression caused by me, from mantis bug #19032 / commit r312070
+ (closes issue #19142) Reported by: vrban Patches:
+ app_voicemail_fix_for_312070.patch uploaded by vrban (license
+ 756) Tested by: vrban, alecdavis
+
+2011-04-13 16:21 +0000 [r313545] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, res/res_agi.c: Asterisk does not hangup a channel
+ after endpoint hangs up. If the call that the dialplan started an
+ AGI script for is hungup while the AGI script is in the middle of
+ a command then the AGI script is not notified of the hangup.
+ There are many AGI Exec commands that this can happen with. The
+ reported applications have been: Background, Wait, Read, and
+ Dial. Also the AGI Get Data command. * Don't wait on the Asterisk
+ channel after it has hung up. The channel is likely to never need
+ servicing again. * Restored the AGI script's ability to return
+ the AGI_RESULT_HANGUP value in run_agi(). It previously only
+ could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the
+ DeadAGI and AGI applications were merged. (closes issue #17954)
+ Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by
+ rmudgett (license 664) issue17954_v1.6.2.patch uploaded by
+ rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue
+ #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761
+ (closes issue #18935) Reported by: nvitaly Tested by: astmiv,
+ rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby
+ Tested by: rmudgett JIRA SWP-2727 Review:
+ https://reviewboard.asterisk.org/r/1165/
+
+2011-04-11 19:30 +0000 [r313277] Leif Madsen <lmadsen at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Fix
+ detection of OpenSSL 1.0 (closes issue #19093) Reported by:
+ tzafrir Patches: detect_openssl_10.diff uploaded by tzafrir
+ (license 46)
+
+2011-04-11 15:27 +0000 [r313188] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Stuck channel using FEATD_MF if caller
+ hangs up at the right time. The cause was actually a caller
+ hanging up just at the end of the Feature Group D DTMF tones that
+ setup the call. The reason for this is a "guard timer" that's
+ implemented using ast_safe_sleep(100). If the caller happens to
+ hang up AFTER the final tone of the DTMF string but BEFORE the
+ end of that ast_safe_sleep(), then ast_safe_sleep() will return
+ non-zero. This causes the code to bounce to the end of
+ ss_thread(), but it does NOT tear down the call properly. This
+ should be a rare occurrence because the caller has to hang up at
+ EXACTLY the right time. Nonetheless, it was happening quite
+ regularly on the reporter's system. It's not easily reproducible,
+ unless you purposely increase the guard-time to 2000 or more.
+ Once you do that, you can reproduce it every time by watching the
+ DTMF debug and hanging up just as it ends. Simply add an
+ ast_hangup() before goto quit. (closes issue #15671) Reported by:
+ jcromes Patches: issue15671.patch uploaded by pabelanger (license
+ 224) Tested by: jcromes
+
+2011-04-05 14:10 +0000 [r312761] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/manager.c, configs/manager.conf.sample: Limit the number of
+ unauthenticated manager sessions and also limit the time they
+ have to authenticate. AST-2011-005 (closes issue #18996) Reported
+ by: tzafrir Tested by: mnicholson
+
+2011-04-04 15:49 +0000 [r312573] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Issues with ISDN
+ calls changing B channels during call negotiations. The handling
+ of the PROCEEDING message was not using the correct call
+ structure if the B channel was changed. (The same for PROGRESS.)
+ The call was also not hungup if the new B channel is not
+ provisioned or is busy. * Made all call connection messages
+ (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT,
+ CONNECT_ACKNOWLEDGE) ensure that they are using the correct
+ structure and B channel. If there is any problem with the
+ operations then the call is now hungup with an appropriate cause
+ code. * Made miscellaneous messages (INFORMATION, FACILITY,
+ NOTIFY) find the correct structure by looking for the call and
+ not using the channel ID. NOTIFY is an exception with versions of
+ libpri before v1.4.11 because a call pointer is not available for
+ Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE,
+ RELEASE_COMPLETE) find the correct structure by looking for the
+ call and not using the channel ID. (closes issue #18313) Reported
+ by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue
+ #18231) Reported by: destiny6628 Tested by: rmudgett JIRA
+ SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA
+ SWP-2929 JIRA AST-437 (The issues fixed here are most likely
+ causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck
+ resetting flag likely fixed)
+
+2011-04-01 11:02 +0000 [r312290] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * apps/app_voicemail.c: app_voicemail: leave_vociemail doesn't use
+ last_message_index to store next message trivial change to bring
+ inline with 1.6.2 1.8 and trunk. The symptom was if msg0000 was
+ missing, and the last was msg0004, the next msgnum would be
+ msg0000 when it should have been msg0005 (issue #18998) Reported
+ by: tootai Patches: bug18998.diff2.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2011-04-01 10:36 +0000 [r312285] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * main/asterisk.c, include/asterisk/select.h: Found some leaking
+ file descriptors while looking at ast_FD_SETSIZE dead code.
+ (issue #18969) Reported by: oej Patches:
+ 20110315__issue18969__14.diff.txt uploaded by tilghman (license
+ 14)
+
+2011-04-01 08:29 +0000 [r312070-312174] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * apps/app_voicemail.c: voicemail: get real last_message_index and
+ count_messages, ODBC resequence change last_message_index to read
+ the max msgnum stored in the database change count_messages to
+ actually count the number of messages. last_message_index change:
+ This fixed overwriting of the last message if msgnum=0 was
+ missing. Previously every incoming message would overwrite
+ msgnum=1. count_messages change: allows us to detect when
+ requencing is required in opneA_mailbox. resequence enabled for
+ ODBC storage: Assists with fixing up corrupt databases with gaps,
+ but only when a user actively opens there mailboxes. (closes
+ issue #18692,#18582,#19032) Reported by: elguero Patches: based
+ on odbc_resequence_mailbox2.1.diff uploaded by elguero (license
+ 37) Tested by: elguero, nivek, alecdavis Review:
+ https://reviewboard.asterisk.org/r/1153/
+
+ * apps/app_voicemail.c: app_voicemail:close_mailbox imap_storage
+ doesn't use last_msg_index
+
+ * apps/app_voicemail.c: app_voicemail: close_mailbox needs to
+ respect additional messages while mailbox is open. close_mailbox
+ leave gaps in message sequence if messages are deleted and new
+ messages arrive during this time, this is because the shuffle
+ down to slot 0, only shuffles the number of pre-existing messages
+ when mailbox is opened, ignoring new arrivals. Fix: in
+ close_mailbox re-evaluate number of messages before the shuffle,
+ this then includes new arrivals. Happens on filebased or ODBC
+ storage. (issues #19032,#18582,#18692,#18998) Reported by:
+ alecdavis,tootai,afosorio Review:
+ https://reviewboard.asterisk.org/r/1153/
+
+2011-03-17 19:14 +0000 [r311199-311203] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_agent.c: Don't hold the pvt lock while streaming a
+ file. ABE-2756
+
+ * main/manager.c: Don't dec the usecount of an eventqent then use
+ it. ABE-2756
+
+ * channels/chan_sip.c: Remove the provisional keepalive scheduler
+ entry's reference to the pvt when we remove the scheduler entry.
+ ABE-2756
+
+2011-03-17 10:43 +0000 [r311048] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * configs/indications.conf.sample: Remove extra quote in
+ indications.conf Picking low hanging fruit. (closes issue #18971)
+ Reported by: IgorG Patches: based on indications.conf.sample.diff
+ uploaded by IgorG (license 20) Tested by: IgorG
+
+2011-03-16 16:58 +0000 [r310888] Terry Wilson <twilson at digium.com>
+
+ * res/res_features.c: Don't delay DTMF in core bridge while
+ listening for DTMF features This patch is mostly the work of Olle
+ Johansson. I did some cleanup and added the silence generating
+ code if transmit_silence is set. When a channel listens for DTMF
+ in the core bridge, the outbound DTMF is not sent until we have
+ received DTMF_END. For a long DTMF, this is a disaster. We send 4
+ seconds of DTMF to Asterisk, which sends no audio for those 4
+ seconds. Some products see this delay and the time skew on RTP
+ packets that results and start ignoring the audio that is sent
+ afterward. With this change, the DTMF_BEGIN frame is inspected
+ and checked. If it matches a feature code, we wait for DTMF_END
+ and activate the feature as before. If transmit_silence=yes in
+ asterisk.conf, silence is sent if we paritally match a
+ multi-digit feature. If it doesn't match a feature, the frame is
+ forwarded along with the DTMF_END without delay. By doing it this
+ way, DTMF is not delayed. (closes issue #15642) Reported by:
+ jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded
+ by twilson (license 396) Tested by: globalnetinc, jde (closes
+ issue #16625) Reported by: sharvanek Review:
+ https://reviewboard.asterisk.org/r/1092/ Review:
+ https://reviewboard.asterisk.org/r/1125/
+
+2011-03-15 00:26 +0000 [r310779] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/utils.c: core show locks: display ThreadID in hexadecimal
+ Allow easier cross referencing of thread ID's with GDB backtraces
+ (closes issue #18968) Reported by: alecdavis Patches:
+ bug18968.diff.txt uploaded by alecdavis (license 585)
+
+2011-03-14 16:38 +0000 [r310633] Richard Mudgett <rmudgett at digium.com>
+
+ * main/callerid.c: "Caller*ID failed checksum" on Wildcard TDM2400P
+ and TDM410 The last character in the caller id message is getting
+ a framing error. The checksum is the last character in the
+ message. A framing error in the checksum could be because: 1) The
+ sender did not send a full stop bit. 2) The sender cut off the
+ FSK carrier too soon. 3) The sender opted to send zero of the
+ specified zero to 10 trailing mark bits and round-off errors in
+ the code resulted in the code not being where it thought it was
+ in the demodulated bit stream. Bit 8 of 'b' is set when parity
+ error. Bit 9 of 'b' is set when framing error. Made ignore the
+ framing and parity error bits if the errored character is the
+ checksum. We can tolerate a framing/parity error there. The
+ checksum character validates the message. (closes issue #18474)
+ Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek
+ (license 636) (with modifications) Tested by: nivek
+
+2011-03-12 20:22 +0000 [r310435] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * pbx/pbx_ael.c: Add AELSub, which provides a stable entry point
+ into AEL subroutines. This commit needs some explanation, given
+ that we're adding a new application into an existing release
+ branch. This is generally a violation of our release policy,
+ except in very limited circumstances, and I believe this is one
+ of those circumstances. The problem that this solves is one of
+ the sanity of using multiple dialplan languages to define a
+ dialplan. In the case of the reporter, he or she is using AEL is
+ define subroutines, while using Realtime extensions to invoke
+ those subroutines. While you can do this, it's based upon the
+ reality of AEL using actual dialplan extensions; however, there
+ is no guarantee that the details of _how_ AEL is compiled into
+ extensions will remain stable. In fact, at the time of this
+ commit, it has already changed twice, once in a fundamental way.
+ Now normally, a new application would only be added to trunk.
+ However, this application is explicitly to create a stable
+ user-level API between versions, and adding it to trunk only will
+ not solve the user's problem of switching between 1.6.2 and 1.8,
+ nor will it help anybody switching from 1.8 to 1.10. Therefore,
+ it needs to go into existing release branches. For the sake of
+ consistency, and also because one of the changes was between 1.4
+ and 1.6.x, I am also electing to commit this to 1.4. (closes
+ issue #18910) Reported by: alexandrekeller Patches:
+ 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman
+ (license 14) 20110304__issue18919__1.4.diff.txt uploaded by
+ tilghman (license 14) Tested by: alexandrekeller
+
+2011-03-10 05:38 +0000 [r310140] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * apps/app_voicemail.c, res/res_config_odbc.c: Initialize column
+ size to 0 to deal with a potential UnixODBC bug on 64-bit
+ systems. (closes issue #18295) Reported by: pruiz
+
+2011-03-08 02:42 +0000 [r309947] Terry Wilson <twilson at digium.com>
+
+ * apps/app_externalivr.c: Don't try to free statically allocated
+ memory. Note: compiling after ./configure --enable-dev-mode will
+ keep these kinds of mistakes from being committed.
+
+2011-03-07 22:02 +0000 [r309856] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_mixmonitor.c: Bug fix for MixMonitor involving filenames
+ with '.' not in the extension Closes issue #18391) Reported by:
+ pabelanger Patches: bugfix.patch uploaded by jrose (license 1225)
+ Tested by: jrose
+
+2011-03-04 00:34 +0000 [r309355] David Ruggles <thedavidfactor at gmail.com>
+
+ * apps/app_externalivr.c: fix small memory leak fix small memory
+ leak caused by a string allocation that wasn't freed (closes
+ issue #18907) Reported by: andy11 Patches:
+ asterisk_trunk-app_externalivr-leak.patch uploaded by andy11
+ (license 1224)
+
+2011-02-24 17:42 +0000 [r308813] Terry Wilson <twilson at digium.com>
+
+ * main/manager.c: Don't broadcast FullyBooted to every AMI
+ connection The FullyBooted event should not be sent to every AMI
+ connection every time someone connects via AMI. It should only be
+ sent to the user who just connected. (closes issue #18168)
+ Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre
+ (license 1142) Tested by: FeyFre, twilson
+
+2011-02-24 14:54 +0000 [r308721] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/udptl.c: silence gcc 4.2 compiler warning
+
+2011-04-25 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.41 Released.
+
+ * AST-2011-005, AST-2011-006
+
+ * Reverted part of r314607, as it can introduce a regression.
+ Specifically, the security check for the "system" privilege was
+ removed. If a user had the "call" privilege but not the "system" privilege,
+ they would lose the ability to execute the system app and dialplan functions
+ that run commands in a shell. This branch never used the "system" privilege
+ for that purpose and did not need to be patched.
+ (AST-2011-006)
+
+2011-02-23 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.41-rc1 Released.
+
+2011-02-21 14:57 +0000 [r308413] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/udptl.c: Properly check the bounds of arrays when decoding
+ UDPTL packets. Also, remove broken support for receiving UDPTL
+ packets larger than 16k. That shouldn't ever happen anyway.
+ AST-2011-002 FAX-281
+
+2011-02-15 23:32 +0000 [r308002] Jason Parker <jparker at digium.com>
+
+ * apps/app_queue.c: Fix regression that changed behavior of queues
+ when ringing a queue member. This reverts r298596, which was to
+ fix a highly bizarre and contrived issue with a queue member that
+ called into his own queue being transferred back into his own
+ queue. I couldn't reproduce that issue in any way. I think one of
+ the other recent transfer fixes actually fixed this. (closes
+ issue #18747) Reported by: vrban
+
+2011-02-11 00:29 +0000 [r307623] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Reentrancy problem if outgoing call gets
+ different B channel than requested. The chan_dahdi
+ pri_fixup_principle() routine needs to protect the Asterisk
+ channel with the channel lock when it changes the technology
+ private pointer to a new private structure. * Added lock
+ protection while pri_fixup_principle() moves a call from one
+ private structure to another. * Made some pri_fixup_principle()
+ messages more meaningful. Partial backport from v1.8 -r300714.
+
+2011-02-10 22:33 +0000 [r307534] Jason Parker <jparker at digium.com>
+
+ * main/asterisk.c: Remove color when executing commands via a
+ remote console. Essentially this makes '-x' imply '-n' on
+ rasterisk. This was done in a different and incomplete way
+ previously, which I'll be reverting shortly. (issue #18776)
+ Reported by: alecdavis
+
+2011-02-08 20:05 +0000 [r306972] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Fix comparison for REFER Replaces tags with
+ pedantic=yes
+
+2011-02-08 19:40 +0000 [r306864-306965] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: fix this line again
+
+ * apps/app_voicemail.c: clean this up, sorry my brain is not really
+ working
+
+ * apps/app_voicemail.c: Backup file storing message duration is not
+ used with IMAP_STORAGE, remove code. The message duration is
+ stored in the body of the email when using IMAP_STORAGE, so
+ nothing needs to happen with the backup file. (closes issue
+ #18718) Reported by: kerframil
+
+ * apps/app_voicemail.c: make this safer and fully correct, pointed
+ out by Steve Davis
+
+2011-02-07 22:35 +0000 [r306617-306672] Terry Wilson <twilson at digium.com>
+
+ * res/res_features.c: Don't try to pickup a call in the middle of a
+ masquerade If A calls B which doesn't answer and C & D both try
+ to do a call pickup, it is possible for ast_pickup_call to answer
+ the call, then fail to masquerade one of the calls because the
+ other one is already in the process of masquerading. This patch
+ checks to see if the channel is in the process of masquerading
+ before call before selecting it for a pickup. Review:
+ https://reviewboard.asterisk.org/r/1094/
+
+ * channels/chan_sip.c: Don't allow a REFER w/replaces to replace
+ its own dialog Asterisk currently accepts a REFER with a Refer-To
+ with an embedded Replaces header that matches the dialog of the
+ REFER. This would be a situation like A calls B, A calls C, A
+ transfers B to A, which is just silly. This patch makes the
+ transfer fail instead of making Asterisk freak out and forget to
+ hang other channels up. Review:
+ https://reviewboard.asterisk.org/r/1093/
+
+2011-02-03 20:43 +0000 [r306120] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_features.c: Fix no MOH and frame queueing problem for
+ parked calls. This was a regression introduced when select was
+ changed to poll and was just a conversion error: POLLPRI detects
+ OOB data, not POLLERR. (closes issue #18637) Reported by: jvandal
+
+2011-02-03 20:36 +0000 [r306119] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c: Set hangup cause in local_hangup When a
+ call involves a local channel (like SIP -> Local -> SIP), the
+ hangup cause was not being set. This resulted in SIP channels
+ sometimes getting a 503 error instead of a 486 when the far side
+ sent a busy. In Asterisk 1.8+ this also can cause issues with
+ CCSS that involve a local channel. This patch sets the
+ hangupcause for one side of the local channel to the other in
+ local_hangup for outbound calls.
+
+2011-02-03 00:02 +0000 [r305888] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, channels/chan_sip.c, apps/app_sendtext.c: Minor
+ AST_FRAME_TEXT related issues. * Include the null terminator in
+ the buffer length. When the frame is queued it is copied. If the
+ null terminator is not part of the frame buffer length, the
+ receiver could see garbage appended onto it. * Add channel lock
+ protection with ast_sendtext().
+
+2011-02-01 17:00 +0000 [r305471] Jason Parker <jparker at digium.com>
+
+ * res/res_musiconhold.c: Close file descriptor for timing source
+ when a MOH class gets destroyed. (closes issue #18457) Reported
+ by: mcallist Patches: 18457-closetimer.diff uploaded by qwell
+ (license 4) 18457-closetimer_trunk.diff uploaded by qwell
+ (license 4) Tested by: qwell, loloski
+
+2011-01-31 23:45 +0000 [r305341] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Obtain the pri lock for PRI queue
+ counters. Need to obtain the pri lock when calling
+ pri_dump_info_str() to avoid a reentrancy problem when
+ calculating the Q.921 Q count statistic. JIRA AST-484
+
+2011-01-31 22:56 +0000 [r305129-305252] Jason Parker <jparker at digium.com>
+
+ * apps/app_dial.c, channels/chan_sip.c: Prevent a crash when
+ dialing a technology with no destination (ex: Dial(SIP/))
+ chan_iax2 and other channel drivers already had code to prevent
+ this. The attempt that app_dial was making to prevent it was not
+ correct, so I fixed that. (closes issue #18371) Reported by:
+ gbour Patches: 18371.patch uploaded by gbour (license 1162)
+
+ * res/res_musiconhold.c: Set file descriptors to -1 on creation, so
+ that we don't see weirdness later.
+
+2011-01-31 06:54 +0000 [r304952] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * apps/app_voicemail.c: Fix compilation when ODBC_STORAGE is
+ defined.
+
+2011-01-29 21:48 +0000 [r304820] Sean Bright <sean at malleable.com>
+
+ * apps/app_meetme.c: Backport MeetMe related reference leaks fixes
+ from 1.6.2/1.8/trunk. I had forgotten that MeetMe in 1.4 also
+ used astobj2, so backport the fixes where appropriate.
+
+2011-01-27 16:57 +0000 [r304460-304464] Jason Parker <jparker at digium.com>
+
+ * configure, configure.ac: Fix default prefix=/usr regression on
+ non-Linux systems. This partially reverts a change made in
+ branches/1.4/ r267759, which will cause issue #17013 to be
+ reopened. This issue was pointed out by a user on #asterisk, who
+ helpfully discovered that paths were being set incorrectly. To
+ truly understand what was wrong, one should run: svn diff --force
+ -c<this revision> configure
+
+ * configure, include/asterisk/autoconfig.h.in: Rerun bootstrap.sh
+ with no changes, so that it is more obvious what my next commit
+ changes.
+
+2011-01-26 21:00 +0000 [r304247] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Convert from network to host byte ordering
+ before checking if an IP is a multicast address.
+
+2011-01-26 20:38 +0000 [r304242] Mark Michelson <mmichelson at digium.com>
+
[... 31448 lines stripped ...]
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