[asterisk-commits] bebuild: tag 1.6.2.23 r359764 - in /tags/1.6.2.23: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Mar 15 15:00:42 CDT 2012
Author: bebuild
Date: Thu Mar 15 15:00:38 2012
New Revision: 359764
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=359764
Log:
Importing files for 1.6.2.23 release.
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tags/1.6.2.23/.version (with props)
tags/1.6.2.23/ChangeLog (with props)
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+2012-03-15 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.6.2.23 Released.
+
+ * AST-2012-002
+
+2012-03-15 18:32 +0000 [r359645] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_milliwatt.c, /: Fix remotely exploitable stack overrun
+ in Milliwatt Milliwatt is vulnerable to a remotely exploitable
+ stack overrun when using the 'o' option. This occurs due to the
+ milliwatt_generate function not accounting for
+ AST_FRIENDLY_OFFSET when calculating the maximum number of
+ samples it can put in the output buffer. This patch resolves this
+ issue by taking into account AST_FRIENDLY_OFFSET when determining
+ the maximum number of samples allowed. Note that at no point is
+ remote code execution possible. The data that is written into the
+ buffer is the pre-defined Milliwatt data, and not custom data.
+ (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested
+ by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by
+ Russell Bryant (license 6283) Note that this patch was written by
+ Russell, even though Matt uploaded it
+
+2011-12-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.6.2.22 Released
+
+2011-12-18 18:25 +0000 [r348515] Kevin P. Fleming <kpfleming at digium.com>
+
+ * configs/sip.conf.sample: Correct two flaws in sip.conf.sample
+ related to AST-2011-013.
+
+ * The sample file listed *two* values
+ for the 'nat' option as being the default. Only 'yes' is the
+ default.
+
+ * The warning about having differing 'nat' settings
+ confusingly referred to both peers and users.
+
+2011-12-08 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.6.2.21 Released.
+
+ * AST-2011-013, AST-2011-014
+
+2011-12-08 21:03 +0000 [r347659] Leif Madsen <lmadsen at digium.com>
+
+ * /: Update svn:externals to use menuselect from 1.6.2.20 and not
+ later. This change is required because when making security
+ releases, if you pull from menuselect/trunk you'll get changes
+ meant for later versions of Asterisk.
+
+2011-12-08 16:17 +0000 [r347530] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Don't crash on INFO automon request with no
+ channel AST-2011-014. When automon was enabled in features.conf,
+ it was possible to crash Asterisk by sending an INFO request if
+ no channel had been created yet. (closes issue ASTERISK-18805)
+
+2011-11-21 20:33 +0000 [r345800-345827] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Don't set the nat default twice. Cleaning up
+ a small merge issue ASTERISK-18862
+
+ * configs/sip.conf.sample, CHANGES, /, channels/chan_sip.c: Default
+ to nat=yes; warn when nat in general and peer differ It is
+ possible to enumerate SIP usernames when the general and
+ user/peer nat settings differ in whether to respond to the port a
+ request is sent from or the port listed for responses in the Via
+ header. In 1.4 and 1.6.2, this would mean if one setting was
+ nat=yes or nat=route and the other was either nat=no or
+ nat=never. In 1.8 and 10, this would mean when one was
+ nat=force_rport and the other was nat=no. In order to address
+ this problem, it was decided to switch the default behavior to
+ nat=yes/force_rport as it is the most commonly used option and to
+ strongly discourage setting nat per-peer/user when at all
+ possible. For more discussion of the issue, please see:
+ http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
+ (closes issue ASTERISK-18862) Review:
+ https://reviewboard.asterisk.org/r/1591/ ........ Merged
+ revisions 345776 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.4
+
+2011-08-05 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.20 Released.
+
+2011-08-01 21:19 +0000 [r330490-330505] Jonathan Rose <jrose at digium.com>
+
+ * main/features.c: fixes reference leak pointed out by rmudgett in
+ https://reviewboard.asterisk.org/r/1337/
+
+ * main/features.c: Asterisk 18103 - Fix reload crash caused by
+ destroying default parking lot Default parking lot was being
+ destroyed in reload and was not being rebuilt properly. This
+ patch keeps features.c reload from destroying the default parking
+ lot in 1.6.2. Bug was caused by a hasty backport which didn't
+ test reload enough times to catch the problem. (closes issue
+ ASTERISK-18103) Reported by: 808blogger Review:
+ https://reviewboard.asterisk.org/r/1337/
+
+2011-07-08 22:26 +0000 [r327255] Jason Parker <jparker at digium.com>
+
+ * cdr, formats, codecs/gsm/src, funcs, bridges, codecs/lpc10,
+ main/db1-ast/btree, codecs/g722, main, main/db1-ast/recno, res,
+ pbx, res/ael, channels, main/stdtime, codecs, agi, utils,
+ main/db1-ast/hash, apps, main/db1-ast/db, main/db1-ast/mpool: Add
+ .o files to svn:ignore property, since it's only ignored if
+ locally configured to do so.
+
+2011-06-28 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.19 Released.
+
+2011-06-28 20:06 +0000 [r325277] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 325275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r325275 | twilson | 2011-06-28 15:03:19 -0500 (Tue, 28 Jun 2011)
+ | 2 lines Don't leak SIP username information ........
+
+2011-06-23 18:21 +0000 [r324643] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Addresses AST-2011-008, memory corruption
+ and remote crash in SIP driver. AST-2011-008
+
+2011-06-23 18:18 +0000 [r324634] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c, /, main/features.c: Merged revisions 324627
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011)
+ | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver
+ Thanks to twilson for identifying the issue and providing the
+ patches. AST-2011-010 ........
+
+2011-06-21 16:10 +0000 [r324306] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_confbridge.c: ConfBridge does not handle hangup properly
+ When playing back a prompt to a channel, confbridge neglects to
+ check for hangup events causing lockup condititions for hangups
+ that occur before actually joining the conference. This change
+ ensures that the user is removed from the conference in the event
+ of a premature hangup. Review:
+ https://reviewboard.asterisk.org/r/1277/
+
+2011-06-15 18:13 +0000 [r323733] Terry Wilson <twilson at digium.com>
+
+ * /, main/features.c: Merged revisions 323732 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011)
+ | 9 lines 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 15:22 +0000 [r323579] Sean Bright <sean at malleable.com>
+
+ * main/manager.c, /: Merged revisions 323559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun
+ 2011) | 25 lines 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) ........
+
+2011-06-10 19:15 +0000 [r323039] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Unlock the sip channel during fax detection
+ like chan_dahdi does to prevent a deadlock with
+ ast_autoservice_stop. (closes issue ASTERISK-17798) tested by
+ mnicholson
+
+2011-06-09 15:37 +0000 [r322668-322699] 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: fix for previous commit
+
+ * /, channels/chan_sip.c: Merged revisions 322646 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r322646 | mnicholson | 2011-06-09 10:10:30 -0500 (Thu, 09 Jun
+ 2011) | 5 lines 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 ........
+
+2011-05-27 08:24 +0000 [r321210] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/features.c: Fix *8 directed pickup locks system during
+ pickupsound play out move playout from sip_pickup_thread to
+ bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2
+ threads trying to write audio to same channel. In addition fixes
+ choppy audio beep in issue 19177. (issue #18654) (issue #19177)
+ Reported by: Docent Patches: review1232-1.6.2.diff.txt uploaded
+ by alecdavis (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1232/
+
+2011-05-23 16:15 +0000 [r320506-320562] David Vossel <dvossel at digium.com>
+
+ * main/tcptls.c: Adds missing part to the ast_tcptls_server_start
+ fails second attempt to bind patch. (closes issue #19289)
+ Reported by: wdoekes Patches:
+ issue19289_delay_old_address_setting_tcptls_2.patch uploaded by
+ wdoekes (license 717)
+
+ * apps/app_chanspy.c: Fixes chanspy enforced mode lacking a
+ channel_unlock. (closes issue #19348) Reported by: wdoekes
+ Patches: issue19348_chanspy_missing_channel_unlock.patch uploaded
+ by wdoekes (license 717)
+
+2011-05-22 23:25 +0000 [r320444] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * res/res_odbc.c: Don't crash when the connection fails. (closes
+ issue #19250) Reported by: seadweller Patches:
+ 20110514__issue19250.diff.txt uploaded by tilghman (license 14)
+ Tested by: seadweller, sum
+
+2011-05-20 21:24 +0000 [r320271] David Vossel <dvossel at digium.com>
+
+ * main/tcptls.c: Fixes issue with ast_tcptls_server_start failing
+ on second attempt to bind. (closes issue #19289) Reported by:
+ wdoekes Patches:
+ issue19289_delay_old_address_setting_tcptls.patch uploaded by
+ wdoekes (license 717)
+
+2011-05-20 20:44 +0000 [r320236] Richard Mudgett <rmudgett at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 320235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011)
+ | 13 lines 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 18:45 +0000 [r320179] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: This commit modifies the way polling is done
+ on TLS sockets. Because of the buffering the TLS layer does,
+ polling is unreliable. If poll is called while there is data
+ waiting to be read in the TLS layer but not at the network layer,
+ the messaging processing engine will not proceed until something
+ else writes data to the socket, which may not occur. This change
+ modifies the logic around TLS sockets to only poll after a failed
+ read on a non-blocking socket. This way we know that there is no
+ data waiting to be read from the buffering layer. (closes issue
+ #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by
+ mnicholson (license 96) Tested by: mnicholson
+
+2011-05-18 23:11 +0000 [r319528-319653] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 319652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011)
+ | 8 lines 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, /: Merged revisions 319527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011)
+ | 10 lines 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) ........
+
+2011-05-16 18:00 +0000 [r319202] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Unlink a peer from peers_by_ip when expiring
+ a registration Review: https://reviewboard.asterisk.org/r/1218/
+
+2011-05-16 15:56 +0000 [r319144] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Fixes issue with peer ref-counting during
+ handle_request_subscribe. (closes issue #19293) Reported by:
+ irroot
+
+2011-05-16 15:51 +0000 [r319141] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Make sure tcptls_session exists before
+ dereferencing it. (closes issue #19192) Reported by: stknob
+ Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by
+ Chainsaw (license 723) Tested by: vois, Chainsaw
+
+2011-05-13 01:14 +0000 [r318636-318735] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/features.h, /, channels/chan_sip.c,
+ apps/app_directed_pickup.c, main/features.c: Merged revisions
+ 318734 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r318734 | rmudgett | 2011-05-12 20:09:40 -0500
+ (Thu, 12 May 2011) | 43 lines 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/ ........
+ ................
+
+ * channels/chan_sip.c: Merged revision 222981 from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 Similar
+ deadlock possible when running the Pickup application internally.
+ ------------------------------------------------------------------------
+ r222981 | dvossel | 2009-10-08 17:04:41 -0500 (Thu, 08 Oct 2009)
+ | 13 lines Deadlock between ast_cel_report_event and
+ ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner
+ channel while only the pvt lock is held. Since pbx_exec calls
+ ast_cel_report_event which attempts to lock the channel, invalid
+ locking order occurs. Channels should be locked before pvt's.
+ (closes issue #15512) Reported by: lmsteffan Patches:
+ ast_cel_deadlock_15512.diff uploaded by dvossel (license 671)
+
+2011-05-11 17:15 +0000 [r318548] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Clean up several chan_sip reference leaks
+ Several situations in the code could lead to peers or sip_pvt
+ references being leaked. This would cause RTP ports to never be
+ destroyed (leading to exhaustion of all available RTP ports) and
+ memory leaks. The original patch for this issue from rgagnon was
+ the result of an obscene amount of testing and hard work, for
+ which I am very grateful. I did some cleanup and added a few
+ additional refcount fixes that I found. (closes issue #17255)
+ Reported by: kvveltho Patches:
+ tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by
+ rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes,
+ loloski Review: https://reviewboard.asterisk.org/r/1101/ Review:
+ https://reviewboard.asterisk.org/r/1207/
+
+2011-05-09 20:04 +0000 [r318331] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Don't offer video to directmedia callee
+ unless caller offered it as well Make sure that when directmedia
+ is enabled, that video is not offered to the callee even if it
+ supports it. p->vrtp will not exist since the caller didn't offer
+ video. (closes issue #19195) Reported by: one47 Patches:
+ sip_cant_add_video_rtp uploaded by one47 (license 23)
+
+2011-05-09 16:51 +0000 [r318230] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Fixes cases where sip_set_rtp_peer can
+ return too early during media path reset. (closes issue #19225)
+ Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by
+ one47 (license 23)
+
+2011-05-06 19:34 +0000 [r317859] Matthew Nicholson <mnicholson at digium.com>
+
+ * pbx/pbx_lua.c: pbx_lua autoservice fixes Don't start an
+ autoservice in pbx_lua if pbx_lua already started one and don't
+ stop one if we didn't start one. Also start and stop the
+ autoservice when transferring control from and to the pbx.
+
+2011-05-06 18:03 +0000 [r317720] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 317719 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317719 | rmudgett | 2011-05-06 12:59:05 -0500 (Fri, 06 May 2011)
+ | 11 lines 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 15:18 +0000 [r317666] Matthew Nicholson <mnicholson at digium.com>
+
+ * pbx/pbx_lua.c: Add a datastore fixup to fix a pbx_lua crash.
+ (closes issue #19055) Reported by: jamhed Patches:
+ lua_datastore_fixup1.diff uploaded by mnicholson (license 96)
+ Tested by: mnicholson, jamhed
+
+2011-05-06 08:04 +0000 [r317575] Terry Wilson <twilson at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 317574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011)
+ | 6 lines 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 18:29 +0000 [r317255] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 317211 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011)
+ | 15 lines 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 17:59 +0000 [r317195] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Set SO_KEEPALIVE on SIP TCP sockets so that
+ they eventually go away when a peer abruptly disappears. This
+ mostly occurs after a successful registration. (closes issue
+ #17544) Reported by: marcelloceschia Patches: (modified)
+ tcptls.patch uploaded by st (license 907)
+
+2011-05-05 14:56 +0000 [r317103] Leif Madsen <lmadsen at digium.com>
+
+ * contrib/scripts/safe_asterisk, /: Merged revisions 317102 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011)
+ | 8 lines 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 16:10 +0000 [r316708] Sean Bright <sean at malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 316707 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed,
+ 04 May 2011) | 8 lines 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 ........
+
+2011-05-04 14:23 +0000 [r316616-316644] David Vossel <dvossel at digium.com>
+
+ * apps/app_chanspy.c: Fixes one-way-audio when chanspy activated
+ with the 'o' option (closes issue #18382) Reported by: jkister
+ Patches:
+ 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt
+ uploaded by malin (license ) Tested by: firstsip, Greenlightcrm,
+ malin, wdoekes, boroda, dvossel
+
+ * channels/chan_sip.c: Fixes session-timers=refuse not being
+ enforced for *caller* During handle_request_invite, the session
+ timer mode was retrieved from a cached variable. This patch
+ forces a peer lookup of the session timer mode in the case of an
+ incoming invite. (closes issue #18804) Reported by: wdoekes
+ Patches: issue18804_session_timer_refuse_caller.patch uploaded by
+ wdoekes (license 717) issue_18804_v2.diff uploaded by dvossel
+ (license 671)
+
+2011-05-04 02:23 +0000 [r316475] Sean Bright <sean at malleable.com>
+
+ * apps/app_meetme.c: Honor the C option to MeetMe when L is passed.
+ This fixes a case that r304773 and friends missed. (closes issue
+ #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff
+ uploaded by var (license 1227) Tested by: seanbright
+
+2011-05-03 21:29 +0000 [r316329] David Vossel <dvossel at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 316328 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03
+ May 2011) | 10 lines 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 19:04 +0000 [r316093] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * funcs/func_curl.c: More possible crashes based upon invalid
+ inputs. (closes issue #18161) Reported by: wdoekes Patches:
+ 20110301__issue18161.diff.txt uploaded by tilghman (license 14)
+ Tested by: wdoekes
+
+2011-04-27 19:03 +0000 [r315893] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 315891 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr
+ 2011) | 14 lines 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 22:52 +0000 [r315643-315672] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 315671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011)
+ | 11 lines 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_queue.c, apps/app_dial.c, /, main/features.c: Merged
+ revisions 315596 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011)
+ | 18 lines 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 19:22 +0000 [r315502] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * include/asterisk/select.h, /: Merged revisions 315501 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011)
+ | 14 lines 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 ........
+
+2011-04-26 02:17 +0000 [r315393] Paul Belanger <pabelanger at digium.com>
+
+ * pbx/pbx_config.c: Add back CLI command 'dialplan save' (closes
+ issue #19140) Reported by: lmadsen Patches:
+ __20110419_dialplan_save.patch.txt uploaded by lmadsen (license
+ 10)
+
+2011-04-25 19:31 +0000 [r315212-315258] Russell Bryant <russell at digium.com>
+
+ * /, formats/format_wav.c: Merged revisions 315257 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25
+ Apr 2011) | 10 lines 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) ........
+
+ * channels/chan_sip.c: Don't link non-cached realtime peers into
+ the peers_by_ip container. (closes issue #18924) Reported by:
+ wdoekes Patches:
+ issue18924_uncached_realtime_peers_leak-1.6.2.17.patch uploaded
+ by wdoekes (license 717)
+
+2011-04-25 07:11 +0000 [r315052] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/chan_local.c, /: Merged revisions 315051 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25
+ Apr 2011) | 11 lines 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 20:49 +0000 [r314958] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 311203,314908 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar
+ 2011) | 4 lines Don't hold the pvt lock while streaming a file.
+ ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48
+ -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and
+ the app threads from using the asterisk channel at the same time.
+ ABE-2756 ........
+
+2011-04-22 14:35 +0000 [r314776-314823] Russell Bryant <russell at digium.com>
+
+ * /: Recorded merge of revisions 314822 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r314822 | russell | 2011-04-22 09:34:23 -0500 (Fri, 22 Apr 2011)
+ | 11 lines 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
+ ........
+
+ * 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
+
+ * main/features.c: Fix handling of some call parking config
+ options. This patch adjusts the handling of some call parking
+ config options to fix some issues that have already been
+ addressed in 1.8 and trunk. (closes issue #19167) Reported by:
+ bluecrow76 Patches:
+ asterisk-1.6.2.17.2-fix-build-parkinglot-parked-AST_FEATURE_FLAGS.diff
+ uploaded by bluecrow76 (license 270)
+
+2011-04-21 18:22 +0000 [r314620] Matthew Nicholson <mnicholson at digium.com>
+
+ * configs/sip.conf.sample, configs/skinny.conf.sample,
+ configs/http.conf.sample, main/manager.c, /, channels/chan_sip.c,
+ channels/chan_skinny.c, main/http.c: Merged revisions 314607 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr
+ 2011) | 14 lines 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 ........
+
+2011-04-21 00:17 +0000 [r314549] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Don't allocate more space than necessary for
+ a sip_pkt This extra allocation is a hold-over from when
+ pkt->data was a character array. Now that it is an allocated
+ string, just allocate enough for the sip_pkt.
+
+2011-04-19 14:27 +0000 [r314202-314205] Leif Madsen <lmadsen at digium.com>
+
+ * funcs/func_channel.c: Remove duplicate documentation from
+ func_channel.c (closes issue #18970) Reported by: IgorG Patches:
+ func_channel.c.doc.diff uploaded by IgorG (license 20)
+
+ * apps/app_dial.c: Update seconds to milliseconds in ast_verb
+ output. (closes issue #19084) Reported by: smurfix Patches:
+ app_dial.patch uploaded by smurfix (license 547) Tested by:
+ lmadsen, smurfix
+
+2011-04-15 14:58 +0000 [r313859] Jonathan Rose <jrose at digium.com>
+
+ * main/cli.c: Fix a Tab Completion bug that occurs due to multiple
+ matches on a substring. Makes word_match function in cli.c repeat
+ a search for a command string until a proper match is found or
+ the string is searched to the last point. (closes issue #17494)
+ Reported by: ffossard Review:
+ https://reviewboard.asterisk.org/r/1180/
+
+2011-04-13 16:29 +0000 [r313579] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, /, res/res_agi.c: Merged revisions 313545 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011)
+ | 41 lines 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/ ........
+
+2011-04-12 18:44 +0000 [r313432-313435] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_dahdi.c: fixing stupid mistake with putting code
+ before variable declaration ........ Merged revisions 313433 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) |
+ 14 lines reload Chan_dahdi memory leak caused by variables
+ chan_dahdi reloading with variables set via setvar in
+ chan_dahdi.conf would stay in the dahdi_pvt structs for
+ individual channels (causing them to just continue adding the new
+ ones to the list) and also there was a memory leak causes by the
+ conf objects. This patch resolves both of these by using
+ ast_variables_destroy during the loading process. (closes issue
+ #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
+ jrose (license 1225) Tested by: tilghman, jrose Review:
+ https://reviewboard.asterisk.org/r/1170/ ........ ........
+
+ * channels/chan_dahdi.c: white space change ........ reload
+ Chan_dahdi memory leak caused by variables chan_dahdi reloading
+ with variables set via setvar in chan_dahdi.conf would stay in
+ the dahdi_pvt structs for individual channels (causing them to
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