[asterisk-commits] russell: tag 1.4.21-rc2 r120861 - /tags/1.4.21-rc2/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jun 6 08:37:56 CDT 2008
Author: russell
Date: Fri Jun 6 08:37:56 2008
New Revision: 120861
URL: http://svn.digium.com/view/asterisk?view=rev&rev=120861
Log:
Importing files for 1.4.21-rc2 release
Added:
tags/1.4.21-rc2/.lastclean (with props)
tags/1.4.21-rc2/.version (with props)
tags/1.4.21-rc2/ChangeLog (with props)
Added: tags/1.4.21-rc2/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.21-rc2/.lastclean?view=auto&rev=120861
==============================================================================
--- tags/1.4.21-rc2/.lastclean (added)
+++ tags/1.4.21-rc2/.lastclean Fri Jun 6 08:37:56 2008
@@ -1,0 +1,1 @@
+32
Propchange: tags/1.4.21-rc2/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.4.21-rc2/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.4.21-rc2/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.4.21-rc2/.version
URL: http://svn.digium.com/view/asterisk/tags/1.4.21-rc2/.version?view=auto&rev=120861
==============================================================================
--- tags/1.4.21-rc2/.version (added)
+++ tags/1.4.21-rc2/.version Fri Jun 6 08:37:56 2008
@@ -1,0 +1,1 @@
+1.4.21-rc2
Propchange: tags/1.4.21-rc2/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.4.21-rc2/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.4.21-rc2/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.4.21-rc2/ChangeLog
URL: http://svn.digium.com/view/asterisk/tags/1.4.21-rc2/ChangeLog?view=auto&rev=120861
==============================================================================
--- tags/1.4.21-rc2/ChangeLog (added)
+++ tags/1.4.21-rc2/ChangeLog Fri Jun 6 08:37:56 2008
@@ -1,0 +1,18086 @@
+2008-06-06 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.21-rc2 released.
+
+2008-06-05 18:03 +0000 [r120731-120735] Russell Bryant <russell at digium.com>
+
+ * UPGRADE-1.2.txt: fix filename
+
+ * UPGRADE-1.2.txt (added), UPGRADE.txt: Add the UPGRADE.txt file
+ from Asterisk 1.2, for handy reference.
+
+2008-06-05 16:56 +0000 [r120675] Philippe Sultan <philippe.sultan at gmail.com>
+
+ * res/res_jabber.c: Ignore appended resource when comparing JIDs.
+
+2008-06-05 16:38 +0000 [r120671] Russell Bryant <russell at digium.com>
+
+ * doc/smdi.txt, res/res_smdi.c: It turns out that searching on the
+ forwarding station isn't very useful for most people, so pull in
+ the changes that allow searching for SMDI messages based on other
+ components of the SMDI message. Also, update the SMDI
+ documentation.
+
+2008-06-04 22:05 +0000 [r120513] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Make sure that the string we set will survive
+ the unref of the queue member. Thanks to Russell, who pointed
+ this out.
+
+2008-06-04 18:35 +0000 [r120425] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_zap.c: If we fail to setup the PRI request channel,
+ don't continue, exit with an error. (closes issue #11989)
+ Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2008-06-04 16:26 +0000 [r120371] Russell Bryant <russell at digium.com>
+
+ * pbx/pbx_config.c: Make the "dialplan remove include" CLI command
+ actually work. Also, tweak some formatting, and make the success
+ message a little bit more clear. (closes AST-52)
+
+2008-06-04 14:11 +0000 [r120285] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Tab completion when removing a member should
+ give the member's interface, not the name, since the interface is
+ what is expected for the command. (closes issue #12783) Reported
+ by: davevg
+
+2008-06-04 13:31 +0000 [r120282] Joshua Colp <jcolp at digium.com>
+
+ * main/pbx.c, pbx/pbx_config.c: Fix a log message and add a message
+ for when the dialplan is done reloading. (closes issue #12716)
+ Reported by: chappell Patches: dialplan_reload_2.diff uploaded by
+ chappell (license 8)
+
+2008-06-03 22:41 +0000 [r120226] Tilghman Lesher <tlesher at digium.com>
+
+ * pbx/pbx_loopback.c: Due to incorrect use of the
+ AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform
+ any translation on the extension number before searching for it
+ in the target context. (closes issue #12473) Reported by:
+ chappell Patches: pbx_loopback.c.diff uploaded by chappell
+ (license 8)
+
+2008-06-03 22:15 +0000 [r120173] Jeff Peeler <jpeeler at digium.com>
+
+ * main/config.c: (closes issue #11594) Reported by: yem Tested by:
+ yem This change decreases the buffer size allocated on the stack
+ substantially in config_text_file_load when LOW_MEMORY is turned
+ on. This change combined with the fix from revision 117462
+ (making mkintf not copy the zt_chan_conf structure) was enough to
+ prevent the crash.
+
+2008-06-03 21:34 +0000 [r120168] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix another place where peer->callno could
+ change at a very bad time, and also fix a place where a peer was
+ used after the reference was released. (inspired by rev 120001)
+
+2008-06-03 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.21-rc1 released.
+
+2008-06-03 18:23 +0000 [r120001-120061] Tilghman Lesher <tlesher at digium.com>
+
+ * main/manager.c: When listing the manager users, managers in
+ users.conf are not shown, even though they are allowed to
+ connect. (closes issue #12594) Reported by: bkruse Patches:
+ 12594-managerusers-2.diff uploaded by qwell (license 4) Tested
+ by: bkruse
+
+ * channels/chan_iax2.c: Save the callno when we're poking, because
+ our peer structure could change during deadlock avoidance (and
+ thus we unlock the wrong callno, causing a cascade failure).
+ (closes issue #12717) Reported by: gewfie Patches:
+ 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: gewfie
+
+2008-06-03 15:26 +0000 [r119929-119966] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
+ pbx/ael/ael-test/ref.ael-vtest13,
+ pbx/ael/ael-test/ref.ael-vtest17,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
+ pbx/ael/ael-test/ref.ael-test15: Updated the regressions on AEL.
+ Hadn't updated this for the changes I made to preserve ${EXTEN}
+ in switches, which affected several tests because it adds extra
+ priorities, and at least one needed to be updated because of the
+ removal of the empty extension warning message.
+
+ * pbx/pbx_ael.c: as per
+ http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
+ which is a message from Philipp Kempgen, requesting that the
+ WARNING that an extension is empty be reduced to a NOTICE or
+ less, as empty extensions are syntactically possible, and no big
+ deal. With which I agree, and have removed that WARNING message
+ entirely. I think it is not necessary to see this message. It
+ didn't state that a NoOp() was inserted automatically on your
+ behalf, and really, as users, who cares? Why freak out dialplan
+ writers with unnecessary warnings? The details of the
+ machinations a compiler goes thru to produce working assembly
+ code is of little interest to most programmers-- we will follow
+ the unix principal of doing our work silently.
+
+2008-06-03 14:46 +0000 [r119926] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Treat ECONNREFUSED as an error that will
+ stop further retransmissions. (issue #AST-58, patch from
+ Switchvox)
+
+2008-06-02 20:08 +0000 [r119742-119838] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Revert a change made for issue #12479. This
+ change caused a regression such that a dial string such as
+ (IAX2/foo) did not automatically fall back to dialing the 's'
+ extension anymore. (closes issue #12770) Reported by: dagmoller
+
+ * main/manager.c: Improve CLI command blacklist checking for the
+ command manager action. Previously, it did not handle case or
+ whitespace properly. This made it possible for blacklisted
+ commands to get executed anyway. (closes issue #12765)
+
+2008-06-02 14:32 +0000 [r119740] Philippe Sultan <philippe.sultan at gmail.com>
+
+ * channels/chan_gtalk.c, res/res_jabber.c: Do not link the guest
+ account with any configured XMPP client (in jabber.conf). The
+ actual connection is made when a call comes in Asterisk. Fix the
+ ast_aji_get_client function that was not able to retrieve an XMPP
+ client from its JID. (closes issue #12085) Reported by: junky
+ Tested by: phsultan
+
+2008-06-02 12:30 +0000 [r119687] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Even of the first PING or LAGRQ doesn't get
+ sent because it comes up too soon, make sure to reschedule so it
+ gets sent later.
+
+2008-06-02 09:29 +0000 [r119585-119636] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/isdn_lib.c: fixed compile issue when dev-mode is
+ enabled
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Added
+ counter for unhandled_bmsg Print, this prevents the logs to be
+ flooded to fast and save CPU in this error scenario. Added
+ 'last_used' element to bc structure, when a bchannel changes from
+ used to free this exact time will be marked in last_used. When a
+ new channel is requested the find_free_chan function will check
+ if the new empty channel was used within the last second, if yes
+ it will search for the next channel, if no it will return this
+ channel. This simple mechanism has prooven to prevent race
+ conditions where the NT and TE tried to allocate the exact same
+ channel at the same time (RELEASE cause: 44).
+
+2008-06-02 01:06 +0000 [r119530-119533] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Change a debug message to an actual debug
+ message
+
+ * apps/app_dial.c: Fix another typo in documentation
+
+2008-06-01 20:47 +0000 [r119478] Michiel van Baak <michiel at vanbaak.info>
+
+ * apps/app_dial.c: small typo fix 'retires' => 'retries'
+
+2008-05-30 21:17 +0000 [r119404] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: When joinempty=strict, it only failed on join
+ if there were busy members. If all members were logged out OR
+ paused, then it (incorrectly) let callers join the queue. (closes
+ issue #12451) Reported by: davidw
+
+2008-05-30 19:46 +0000 [r119354] Joshua Colp <jcolp at digium.com>
+
+ * main/autoservice.c: Fix a bug I found while testing for another
+ issue.
+
+2008-05-30 16:44 +0000 [r119301] Michiel van Baak <michiel at vanbaak.info>
+
+ * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
+ contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.mandrake.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk: dont use a bashism way to
+ check the $VERSION variable. The rc/init.d scripts, and
+ safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2
+ (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski)
+ (closes issue #12687) Reported by: loloski Patches:
+ 20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by
+ mvanbaak (license 7) Tested by: loloski, mvanbaak
+
+2008-05-30 12:55 +0000 [r119076-119238] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 119237 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30
+ May 2008) | 7 lines - Instead of only enforcing destination call
+ number checking on an ACK, check all full frames except for PING
+ and LAGRQ, which may be sent by older versions too quickly to
+ contain the destination call number. (As suggested by Tim Panton
+ on the asterisk-dev list) - Merge changes from
+ team/russell/iax2-frame-race, which prevents PING and LAGRQ from
+ being sent before the destination call number is known. ........
+
+ * main/autoservice.c: Fix a race condition in channel autoservice.
+ There was still a small window of opportunity for a DTMF frame,
+ or some other deferred frame type, to come in and get dropped.
+ (closes issue #12656) (closes issue #12656) Reported by: dimas
+ Patches: v3-12656.patch uploaded by dimas (license 88) -- with
+ some modifications by me
+
+ * include/asterisk/audiohook.h: Oddly enough, all of the contents
+ of audiohook.h were in there twice. I have removed the second
+ copy.
+
+2008-05-29 20:24 +0000 [r119071] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_zap.c: Call waiting tone occurs too often, because
+ it's getting serviced by both subchannels. (closes issue #11354)
+ Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded
+ by Corydon76 (license 14)
+
+2008-05-29 19:04 +0000 [r118956-119012] Russell Bryant <russell at digium.com>
+
+ * apps/app_milliwatt.c: - Fix a typo in the argument to Playtones -
+ use ast_safe_sleep() instead of calling the wait application
+ (thanks to tilghman for pointing these out!)
+
+ * /, channels/chan_iax2.c: Merged revisions 119008 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29
+ May 2008) | 7 lines Merge changes from
+ team/russell/iax2-another-fix-to-the-fix As described in the
+ following post to the asterisk-dev mailing list, only enforce
+ destination call numbers when processing an ACK.
+ http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
+ (closes issue #12631) ........
+
+ * apps/app_milliwatt.c: - Mark app_milliwatt dependent on
+ res_indications (thanks to jsmith) - fix a typo in a log message
+ (thanks to qwell)
+
+ * apps/app_milliwatt.c: Change milliwatt to use the proper tone by
+ default (1004 Hz) instead of 1000 Hz. An option is there to use
+ 1000 Hz for anyone that might want it.
+
+2008-05-29 17:33 +0000 [r118953-118954] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/lock.h: Define also when not DEBUG_THREADS
+
+ * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
+ channels/chan_agent.c, channels/chan_alsa.c, main/utils.c,
+ include/asterisk/lock.h, channels/chan_iax2.c: Add some debugging
+ code that ensures that when we do deadlock avoidance, we don't
+ lose the information about how a lock was originally acquired.
+
+2008-05-29 00:25 +0000 [r118858] Steve Murphy <murf at digium.com>
+
+ * main/cdr.c, apps/app_forkcdr.c: (closes issue #10668) (closes
+ issue #11721) (closes issue #12726) Reported by: arkadia Tested
+ by: murf These changes: 1. revert the changes made via bug 10668;
+ I should have known that such changes, even tho they made sense
+ at the time, seemed like an omission, etc, were actually integral
+ to the CDR system via forkCDR. It makes sense to me now that
+ forkCDR didn't natively end any CDR's, but rather depended on
+ natively closing them all at hangup time via traversing and
+ closing them all, whether locked or not. I still don't completely
+ understand the benefits of setvar and answer operating on locked
+ cdrs, but I've seen enough to revert those changes also, and stop
+ messing up users who depended on that behavior. bug 12726 found
+ reverting the changes fixed his changes, and after a long review
+ and working on forkCDR, I can see why. 2. Apply the suggested
+ enhancements proposed in 10668, but in a completely compatible
+ way. ForkCDR will behave exactly as before, but now has new
+ options that will allow some actions to be taken that will
+ slightly modify the outcome and side-effects of forkCDR. Based on
+ conversations I've had with various people, these small tweaks
+ will allow some users to get the behavior they need. For
+ instance, users executing forkCDR in an AGI script will find the
+ answer time set, and DISPOSITION set, a situation not covered
+ when the routines were first written. 3. A small problem in the
+ cdr serializer would output answer and end times even when they
+ were not set. This is now fixed.
+
+2008-05-28 16:10 +0000 [r118716] Brett Bryant <bbryant at digium.com>
+
+ * channels/chan_iax2.c: merge revision 118702 from trunk to 1.4 --
+ Fixes a bug in chan_iax that uses send_command to poke a peer
+ while a channel is unlocked in some cases, and because it can
+ cause seemingly random failures could be related to some bugs in
+ the tracker...
+
+2008-05-28 14:23 +0000 [r118558-118646] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add an
+ option to use the source IP address of RTP as the destination IP
+ address of UDPTL when a specific option is enabled. If the remote
+ side is properly configured (ports forwarded) then UDPTL will
+ flow. (closes issue #10417) Reported by: cstadlmann
+
+ * channels/chan_sip.c: Fix an issue where codec preferences were
+ not set on dialogs that were not authenticated via a user or peer
+ and allow framing to work without rtpmap in the SDP. (closes
+ issue #12501) Reported by: slimey
+
+2008-05-27 19:15 +0000 [r118551] Tilghman Lesher <tlesher at digium.com>
+
+ * main/cli.c: When showing an error message for a command, don't
+ shorten the command output, as it tends to confuse the user (it's
+ fine for suggesting other commands, however). Reported by:
+ seanbright (on #asterisk-dev) Fixed by: me
+
+2008-05-27 19:07 +0000 [r118509] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_chanspy.c: Russell noted to me that in the case that
+ separate threads use their own addressing system, the fix I made
+ for issue 12376 does not guarantee uniqueness to the datastores'
+ uids. Though I know of no system that works this way, I am going
+ to change this right now to prevent trying to track down some
+ future bug that may occur and cause untold hours of debugging
+ time to track down. The change involves using a global counter
+ which increases with each new chanspy_ds which is created. This
+ guarantees uniqueness.
+
+2008-05-27 18:58 +0000 [r118465] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: NULL character should terminate only commands
+ back to the core, not log messages to the console. (closes issue
+ #12731) Reported by: seanbright Patches:
+ 20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: seanbright
+
+2008-05-27 17:17 +0000 [r118416] Michiel van Baak <michiel at vanbaak.info>
+
+ * apps/app_voicemail.c: small update to the g() option of
+ app_voicemail to note that gain changes only work on zap channels
+ right now. issue #12578 shows it's not clear right now.
+
+2008-05-27 16:38 +0000 [r118365] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_chanspy.c: Add a unique id to the datastore allocated in
+ app_chanspy since it is possible that multiple spies may be
+ listening to the same channel. (closes issue #12376) Reported by:
+ DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut
+ (license 60) Tested by: destiny6628 (closes issue #12243)
+ Reported by: atis
+
+2008-05-27 15:45 +0000 [r118358] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/queues.conf.sample: Add a note that pbx_config.so is
+ needed for Local channels. (Closes issue #12671)
+
+2008-05-25 16:02 +0000 [r118251] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Realtime flag affects construction in
+ multiple ways, so consulting whether rtcachefriends was set was
+ done too soon (needed to be done inside build_peer, not just as a
+ flag to build_peer). Also, fullcontact needed to be
+ reconstructed, because realtime separates the embedded ';' into
+ multiple fields. (closes issue #12722) Reported by: barthpbx
+ Patches: 20080525__bug12722.diff.txt uploaded by Corydon76
+ (license 14) Tested by: barthpbx (Much of the discussion happened
+ on #asterisk-dev for diagnosing this issue)
+
+2008-05-23 21:21 +0000 [r118163] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_zap.c: Fix a few things I missed to ensure
+ zt_chan_conf structure is not modified in mkintf
+
+2008-05-23 13:18 +0000 [r118052-118055] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/utils.h: Add format type checking for recently
+ de-inlined function
+
+ * doc/cli.txt (added), doc/00README.1st: Add information on using
+ the Asterisk console, including tab command line completion.
+ (Closes issue #12681)
+
+2008-05-23 12:30 +0000 [r118048] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/utils.h, main/utils.c: Don't declare a function
+ that takes variable arguments as inline, because it's not valid,
+ and on some compilers, will emit a warning.
+ http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes
+ issue #12289) Reported by: francesco_r Patches by Tilghman, final
+ patch by me
+
+2008-05-22 18:53 +0000 [r117809-117899] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Also remove preamble from asynchronous events
+ (reported by jsmith on #asterisk-dev)
+
+ * funcs/func_realtime.c: Take into account the length of delimiters
+ when calculating result string length. (closes issue #12696)
+ Reported by: adomjan Patches: func_realtime.c-longdelimiter.patch
+ uploaded by adomjan (license 487)
+
+2008-05-21 20:11 +0000 [r117582] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_zap.c: Ensure that passed in zt_chan_conf structure
+ is not modified in mkintf.
+
+2008-05-21 19:38 +0000 [r117574] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Apply the autoframing setting to dialogs
+ that do not get matched against a user or peer.
+
+2008-05-21 18:44 +0000 [r117519-117523] Tilghman Lesher <tlesher at digium.com>
+
+ * pbx/pbx_spool.c: Revert accidental commit of the last change
+
+ * main/asterisk.c, pbx/pbx_spool.c: Strip the preamble from the
+ output also when -rx is not being used (Related to issue #12702)
+
+2008-05-21 18:28 +0000 [r117479-117514] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c: Don't filter the magic character in the network
+ verboser. It gets filtered once it reaches the client. (related
+ to issue #12702, pointed out by tilghman)
+
+ * main/asterisk.c, pbx/pbx_gtkconsole.c: 1) Don't print the verbose
+ marker in front of every message from ast_verbose() being sent to
+ remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose
+ marker. (related to issue #12702)
+
+ * main/asterisk.c: Don't display the verbose marker for calls to
+ ast_verbose() that do not include a VERBOSE_PREFIX in front of
+ the message. (closes issue #12702) Reported by: johnlange Patched
+ by me
+
+2008-05-21 16:58 +0000 [r117462] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_zap.c: Pass a pointer for the conf parameter to the
+ function mkintf rather than the whole zt_chan_conf structure.
+
+2008-05-20 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.20 released.
+
+2008-05-14 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.20-rc3 released.
+
+2008-05-14 12:51 +0000 [r116230] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Accept text messages even with Content-Type:
+ text/plain;charset=Södermanländska
+
+2008-05-13 23:47 +0000 [r116088] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c, include/asterisk/lock.h: A change to the way
+ channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
+ After debugging a deadlock, it was noticed that when
+ DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin
+ of channel locks is obscured by the fact that all channel locks
+ appear to happen in the function ast_channel_lock(). This code
+ change redefines ast_channel_lock to be a macro which maps to
+ __ast_channel_lock(), which then relays the proper file name,
+ line number, and function name information to the core lock
+ functions so that this information will be displayed in the case
+ that there is some sort of locking error or core show locks is
+ issued.
+
+2008-05-13 21:17 +0000 [r115990-116038] Russell Bryant <russell at digium.com>
+
+ * channels/chan_local.c: Fix a deadlock involving channel
+ autoservice and chan_local that was debugged and fixed by
+ mmichelson and me. We observed a system that had a bunch of
+ threads stuck in ast_autoservice_stop(). The reason these threads
+ were waiting around is because this function waits to ensure that
+ the channel list in the autoservice thread gets rebuilt before
+ the stop() function returns. However, the autoservice thread was
+ also locked, so the autoservice channel list was never getting
+ rebuilt. The autoservice thread was stuck waiting for the channel
+ lock on a local channel. However, the local channel was locked by
+ a thread that was stuck in the autoservice stop function. It
+ turned out that the issue came down to the local_queue_frame()
+ function in chan_local. This function assumed that one of the
+ channels passed in as an argument was locked when called.
+ However, that was not always the case. There were multiple cases
+ in which this channel was not locked when the function was
+ called. We fixed up chan_local to indicate to this function
+ whether this channel was locked or not. The previous assumption
+ had caused local_queue_frame() to improperly return with the
+ channel locked, where it would then never get unlocked. (closes
+ issue #12584) (related to issue #12603)
+
+ * main/autoservice.c: Fix an issue that I noticed in autoservice
+ while mmichelson and I were debugging a different problem. I
+ noticed that it was theoretically possible for two threads to
+ attempt to start the autoservice thread at the same time. This
+ change makes the process of starting the autoservice thread,
+ thread-safe.
+
+2008-05-13 20:28 +0000 [r115944] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_alsa.c: Use the right flag to open the audio in
+ non-blocking. (closes issue #12616) Reported by:
+ nicklewisdigiumuser
+
+2008-05-13 18:36 +0000 [r115884] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: If the socket dies (read returns 0=EOF), return
+ immediately. (Closes issue #12637)
+
+2008-05-12 17:51 +0000 [r115735] Mark Michelson <mmichelson at digium.com>
+
+ * main/utils.c: If a thread holds no locks, do not print any
+ information on the thread when issuing a core show locks command.
+ This will help to de-clutter output somewhat. Russell said it
+ would be fine to place this improvement in the 1.4 branch, so
+ that's why it's going here too.
+
+2008-05-09 16:34 +0000 [r115579] Joshua Colp <jcolp at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Improve res_ninit and res_ndestroy autoconf logic on the Darwin
+ platform.
+
+2008-05-08 19:19 +0000 [r115545-115568] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Remove debug output.
+
+ * /, channels/chan_iax2.c: Merged revisions 115564 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08
+ May 2008) | 25 lines Fix a race condition that bbryant just found
+ while doing some IAX2 testing. He was running Asterisk trunk
+ running IAX2 calls through a few Asterisk boxes, however, the
+ audio was extremely choppy. We looked at a packet trace and saw a
+ storm of INVAL and VNAK frames being sent from one box to
+ another. It turned out that what had happened was that one box
+ tried to send a CONTROL frame before the 3 way handshake had
+ completed. So, that frame did not include the destination call
+ number, because it didn't have it yet. Part of our recent work
+ for security issues included an additional check to ensure that
+ frames that are supposed to include the destination call number
+ have the correct one. This caused the frame to be rejected with
+ an INVAL. The frame would get retransmitted for forever, rejected
+ every time ... This race condition exists in all versions that
+ got the security changes, in theory. However, it is really only
+ likely that this would cause a problem in Asterisk trunk. There
+ was a control frame being sent (SRCUPDATE) at the _very_
+ beginning of the call, which does not exist in 1.2 or 1.4.
+ However, I am fixing all versions that could potentially be
+ affected by the introduced race condition. These changes are what
+ bbryant and I came up with to fix the issue. Instead of simply
+ dropping control frames that get sent before the handshake is
+ complete, the code attempts to wait a little while, since in most
+ cases, the handshake will complete very quickly. If it doesn't
+ complete after yielding for a little while, then the frame gets
+ dropped. ........
+
+ * channels/chan_sip.c: Don't give up on attempting an outbound
+ registration if we receive a 408 Timeout. (closes issue #12323)
+
+ * contrib/scripts/postgres_cdr.sql (removed): remove
+ postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as
+ well (closes issue #9676)
+
+ * contrib/init.d/rc.debian.asterisk: Don't exit the script if
+ Asterisk is not running. (closes issue #12611)
+
+ * main/pbx.c: Don't use a channel before checking for channel
+ allocation failure. (closes issue #12609) Reported by: edantie
+
+ * contrib/init.d/rc.debian.asterisk: Use the same method for
+ executing Asterisk as the rest of the script. (closes issue
+ #12611) Reported by: b_plessis
+
+2008-05-07 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.20-rc2 released.
+
+2008-05-07 18:17 +0000 [r115512-115517] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Track peer references when stored in the
+ sip_pvt struct as the peer related to a qualify ping or a
+ subscription. This fixes some realtime related crashes. (closes
+ issue #12588) (closes issue #12555)
+
+2008-05-06 19:55 +0000 [r115418-115422] Jason Parker <jparker at digium.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
+ 7 lines read requires an argument on some non-bash shells (closes
+ issue #12593) Reported by: bkruse Patches:
+ getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
+ ........
+
+ * res/res_musiconhold.c: Switch to using ast_random() rather than
+ just rand(). This does not fix the bug reported, but I believe it
+ is correct. (from issue #12446) Patches: bug_12446.diff uploaded
+ by snuffy (license 35)
+
+2008-05-06 19:31 +0000 [r115415] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Don't print the terminating NUL. (Closes issue
+ #12589)
+
+2008-05-06 13:54 +0000 [r115341] Joshua Colp <jcolp at digium.com>
+
+ * configure, configure.ac: Add in missing argument.
+
+2008-05-05 22:50 +0000 [r115333] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, main/logger.c: Separate verbose output from CLI
+ output, by using a preamble. (closes issue #12402) Reported by:
+ Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt
+ uploaded by Corydon76 (license 14)
+ 20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by
+ Corydon76 (license 14)
+
+2008-05-05 22:10 +0000 [r115327] Joshua Colp <jcolp at digium.com>
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
+ configure.ac: Make sure that either the main speex library
+ contains preprocess functions or that speexdsp does. If both fail
+ then speex stuff can not be built.
+
+2008-05-05 21:41 +0000 [r115320] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Don't consider a caller "handled" until the
+ caller is bridged with a queue member. There was too much of an
+ opportunity for the member to hang up (either during a delay,
+ announcement, or overly long agi) between the time that he
+ answered the phone and the time when he actually was bridged with
+ the caller. The consequence of this was that if the member hung
+ up in that interval, then proper abandonment details would not be
+ noted in the queue log if the caller were to hang up at any point
+ after the member hangup. (closes issue #12561) Reported by:
+ ablackthorn
+
+2008-05-05 20:17 +0000 [r115308-115312] Tilghman Lesher <tlesher at digium.com>
+
+ * Makefile: Reverse order, such that user configs override default
+ selections
+
+ * include/asterisk/res_odbc.h: Err, the documentation on the return
+ value of ast_odbc_backslash_is_escape is exactly backwards.
+
+2008-05-05 19:49 +0000 [r115297-115304] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Avoid putting opaque="" in Digest
+ authentication. This patch came from switchvox. It fixes
+ authentication with Primus in Canada, and has been in use for a
+ very long time without causing problems with any other providers.
+ (closes issue AST-36)
+
+2008-05-05 03:22 +0000 [r115285] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
+ contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.mandrake.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk: When starting Asterisk, bug
+ out if Asterisk is already running. (closes issue #12525)
+ Reported by: explidous Patches: 20080428__bug12525.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: mvanbaak
+
+2008-05-04 02:09 +0000 [r115276-115282] Joshua Colp <jcolp at digium.com>
+
+ * configure, acinclude.m4: Expand the test function for GCC
+ attributes so that more complex attributes are properly
+ recognized.
+
+ * include/asterisk/compiler.h: For my next trick I will make these
+ work with what our autoconf header file gives us.
+
+ * configure, acinclude.m4: Treat warnings as errors when checking
+ if a GCC attribute exists. We have to do this as GCC will just
+ ignore the attribute and pop up a warning, it won't actually fail
+ to compile.
+
+2008-05-02 20:25 +0000 [r115257] Brett Bryant <bbryant at digium.com>
+
+ * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, CHANGES: Add new "pri show version" command to show
+ the libpri version for support reasons.
+
+2008-05-02 14:28 +0000 [r115196] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/sched.h: Clarify a comment that was, well, just
+ wrong. It turns out that ignoring the way that macros expand.
+ Instead, I have clarified in the comment why the macro will work
+ even if the scheduler id for the task to be deleted changes
+ during the execution of the macro.
+
+2008-05-01 23:20 +0000 [r115017-115102] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/res_odbc.h: Change the comment of deprecated to
+ an actual compiler deprecation
+
+ * main/utils.c: '#' is another reserved character for URIs that
+ also needs to be escaped. (closes issue #10543) Reported by:
+ blitzrage Patches: 20080418__bug10543.diff.txt uploaded by
+ Corydon76 (license 14)
+
+2008-05-01 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.20-rc1 released.
+
+2008-04-30 16:30 +0000 [r114891] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
+ Merge changes from team/russell/iax2_find_callno and
+ iax2_find_callno_1.4 These changes address a critical performance
+ issue introduced in the latest release. The fix for the latest
+ security issue included a change that made Asterisk randomly
+ choose call numbers to make them more difficult to guess by
+ attackers. However, due to some inefficient (this is by far, an
+ understatement) code, when Asterisk chose high call numbers,
+ chan_iax2 became unusable after just a small number of calls. On
+ a small embedded platform, it would not be able to handle a
+ single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
+ more than about 16 IAX2 channels. Ouch. These changes address
+ some performance issues of the find_callno() function that have
+ bothered me for a very long time. On every incoming media frame,
+ it iterated through every possible call number trying to find a
+ matching active call. This involved a mutex lock and unlock for
+ each call number checked. So, if the random call number chosen
+ was 20000, then every media frame would cause 20000 locks and
+ unlocks. Previously, this problem was not as obvious since
+ Asterisk always chose the lowest call number it could. A second
+ container for IAX2 pvt structs has been added. It is an astobj2
+ hash table. When we know the remote side's call number, the pvt
+ goes into the hash table with a hash value of the remote side's
+ call number. Then, lookups for incoming media frames are a very
+ fast hash lookup instead of an absolutely insane array traversal.
+ In a quick test, I was able to get more than 3600% more IAX2
+ channels on my machine with these changes.
+
+2008-04-30 16:23 +0000 [r114890] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Don't crash on bad SIP replys. Fix created
+ in Huntsville together with Mark M (putnopvut) (closes issue
+ #12363) Reported by: jvandal Tested by: putnopvut, oej
+
+2008-04-30 14:46 +0000 [r114875-114880] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro
+ for indexing through the iaxs/iaxsl arrays so that the size of
+ the arrays can be adjusted in one place, and change the size of
+ the arrays from 32768 calls to 2048 calls when LOW_MEMORY is
+ defined
+
+ * Makefile.rules: pay attention to *all* header files for
+ dependency tracking, not just the local ones (inspired by r578 of
+ asterisk-addons by tilghman)
+
+2008-04-29 19:40 +0000 [r114848] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel
+ variables instead of the channel's macrocontext and macroexten
+ fields. This is needed because if macros are daisy-chained, the
+ incorrect context and extension are placed on the new channel. I
+ also added locking to the channel prior to accessing these
+ variables as noted in trunk's janitor project file. (closes issue
+ #12549) Reported by: darren1713 Patches:
+ app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
+ (with modifications from me) Tested by: putnopvut
+
+2008-04-29 17:08 +0000 [r114829] Jason Parker <jparker at digium.com>
+
+ * res/res_config_pgsql.c: Change warning message to debug, since
+ there are cases where 0 results is perfectly fine.
+
+2008-04-29 12:53 +0000 [r114823] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
+ 2008) | 2 lines stop script from appending source code if run
+ multiple times ........
+
+2008-04-28 04:47 +0000 [r114708] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, channels/chan_gtalk.c: When modules are
[... 17293 lines stripped ...]
More information about the asterisk-commits
mailing list