[asterisk-commits] russell: tag 1.4.5 r69600 - in /tags/1.4.5: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jun 15 16:03:21 CDT 2007
Author: russell
Date: Fri Jun 15 16:03:20 2007
New Revision: 69600
URL: http://svn.digium.com/view/asterisk?view=rev&rev=69600
Log:
importing files for 1.4.5 release
Added:
tags/1.4.5/.lastclean (with props)
tags/1.4.5/.version (with props)
tags/1.4.5/ChangeLog (with props)
Added: tags/1.4.5/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.5/.lastclean?view=auto&rev=69600
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--- tags/1.4.5/ChangeLog (added)
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+2007-06-15 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.5 released.
+
+2007-06-15 20:18 +0000 [r69579] Russell Bryant <russell at digium.com>
+
+ * res/res_features.c: Fix a silly deadlock in res_features that I
+ found while debugging on one of blitzrage's test machines. It was
+ one of the situations where he was seeing hung channels, and may
+ be the cause of some of the reports from other people. (related
+ to issue #9235)
+
+2007-06-15 19:23 +0000 [r69558] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_speech_utils.c: Add support for setting the maximum
+ length of acceptable DTMF in SpeechBackground.
+
+2007-06-15 15:27 +0000 [r69518] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: The SLATRUNK_STATUS variable indicated
+ "SUCCESS" for both an answer of the incoming call on the trunk,
+ or if the trunk reached its ring timeout. This patch changes the
+ variable to say "RINGTIMEOUT" in that case. (issue #9973,
+ reported by n00dle, patch by me)
+
+2007-06-14 23:22 +0000 [r69434-69470] Jason Parker <jparker at digium.com>
+
+ * main/config.c, /: Merged revisions 69469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4
+ lines Fix an issue where the line number in an unterminated
+ comment block error message would show the wrong line number.
+ "Reported" to me on #asterisk (somebody posted an error message,
+ and I happened to catch it) ........
+
+ * sounds/Makefile: Update to latest versions of sound files.
+
+2007-06-14 21:50 +0000 [r69392] Kevin P. Fleming <kpfleming at digium.com>
+
+ * cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c,
+ cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c,
+ main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c,
+ apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c,
+ main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
+ channels/chan_iax2.c: use ast_localtime() in every place
+ localtime_r() was being used
+
+2007-06-14 21:08 +0000 [r69358] Russell Bryant <russell at digium.com>
+
+ * main/say.c: Fix some problems with saying dates and times for the
+ "tw" langauge (issue #9964, ljmid)
+
+2007-06-14 15:21 +0000 [r69259] Jason Parker <jparker at digium.com>
+
+ * funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun
+ 2007) | 4 lines Change a quite broken while loop to a for loop,
+ so "continue;" works as expected instead of eating 99% CPU...
+ Issue 9966, patch by me. ........
+
+2007-06-13 21:19 +0000 [r69184-69222] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: Whoops...
+
+ * channels/chan_iax2.c: Let's make chan_iax2 media only native
+ transfers actually work. (issue #9376 reported by simone
+ cittadini)
+
+ * channels/iax2-parser.c: Add TXMEDIA to list so that it is
+ properly displayed during iax2 packet output.
+
+2007-06-13 19:57 +0000 [r69183] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Move the logic for destroying a call when no
+ response is received to a BYE outside of the block that checks
+ for FLAG_FATAL to be set. This flag is only set when the packet
+ is transmitted with the reliability set to XMIT_CRITICAL when the
+ original packet is transmitted. A BYE is always sent with it set
+ to XMIT_RELIABLE, meaning this code could never be encountered.
+ This resulted in seeing some SIP channels that would never go
+ away with the last packet sent being a BYE. (part of issue #9235,
+ patch from jcmoore)
+
+2007-06-13 19:41 +0000 [r69181] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Contains a patch for fixing an encoding
+ problem when using Outlook to view voicemail emails and
+ attachments. This fix has also been tested on Thunderbird,
+ Evolution, Pine, and Mutt. (Issue 9336, reported by marwick,
+ patched by mutterc)
+
+2007-06-13 19:08 +0000 [r69128-69144] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_meetme.c: Really ignore NULL frames and check whether
+ the channel hungup or not. (issue #9912 reported by junky)
+
+ * /, main/app.c: Merged revisions 69127 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2
+ lines Return group counting to previous behavior where you could
+ only have one group per category. (issue #9711 reported by
+ irroot) ........
+
+2007-06-13 16:56 +0000 [r69016-69071] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Clarify a bit of logic. This doesn't change
+ behavior in any way, but it is helpful when following the logic
+ to debug problems like 9235.
+
+ * channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct
+ was accessed without the lock held. This issue was reported to me
+ via email by Dmitry Mishchenko. Thanks!
+
+ * cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois
+ in #asterisk-bugs. PQclear() was not called on the result
+ structure after doing a PQexec(). Also, fix up some formatting in
+ passing.
+
+2007-06-12 19:36 +0000 [r69012-69014] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: Change the full frame dropping log message
+ to debug to avoid future bug reports.
+
+ * channels/chan_iax2.c: Schedule the sending of a PING packet a
+ second later than previously so that it does not collide with the
+ LAGRQ.
+
+2007-06-12 19:13 +0000 [r69010] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: In ast_channel_make_compatible(), just return if
+ the channels' read and write formats already match up. There are
+ code paths that call this function on a pair of channels multiple
+ times. This made calls fail that were using g729 in some cases.
+ The reason is that codec_g729a will unregister itself from the
+ list of available translators will all licenses are in use. So,
+ the first time the function got called, the right translation
+ path was allocated. However, the second time it got called, the
+ code would not find a translation path to/from g729 and make the
+ call fail, even if the channel actually already had a g729
+ translation path allocated. (SPD-32)
+
+2007-06-12 14:23 +0000 [r68922] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c, /: Merged revisions 68921 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2
+ lines Bring RTP back to Asterisk at the end of a native bridge no
+ matter what. ........
+
+2007-06-11 21:20 +0000 [r68814] Jason Parker <jparker at digium.com>
+
+ * include/asterisk/time.h: Solaris 10 sometimes (?) needs this
+ include in order to have NULL defined.
+
+2007-06-11 20:45 +0000 [r68781] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly
+ used
+
+2007-06-11 16:57 +0000 [r68733] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
+ Merged revisions 68732 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) |
+ 1 line added check for NULL Pointer when calling misdn_new.
+ Asterisk does not allow us to create channels anymore when stop
+ gracefully is used :). also modified the restart_indicator to 0
+ ........
+
+2007-06-11 14:33 +0000 [r68683] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c, /: Merged revisions 68682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2
+ lines Improve deadlock handling of the channel list. (issue #8376
+ reported by one47) ........
+
+2007-06-11 10:29 +0000 [r68644] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, /, channels/misdn/ie.c,
+ channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) |
+ 1 line fixed problem that the dummybc chanels had no lock,
+ checking for the lock now. Also fixed the channel restart stuff,
+ we can now specify and restart particular channels too. ........
+
+2007-06-11 04:21 +0000 [r68595] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * pbx/pbx_config.c: "dialplan save" produced garbage in the config
+ file
+
+2007-06-08 22:23 +0000 [r68527] Russell Bryant <russell at digium.com>
+
+ * /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) |
+ 4 lines Don't automatically hang up after running Dictate so that
+ callers can exit cleanly using '#' (closes issue #9577, patch
+ from Thomas Andrews) ........
+
+2007-06-08 15:52 +0000 [r68450] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_iax2.c: actually remember the type/subclass of full
+ frames that are in process
+
+2007-06-08 00:17 +0000 [r68370-68401] Joshua Colp <jcolp at digium.com>
+
+ * /, main/say.c: Merged revisions 68397 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2
+ lines Don't call ast_waitstream_full when the control file
+ descriptor and audio file descriptor are not set, simply call
+ ast_waitstream! (issue #8530 reported by rickead2000) ........
+
+ * main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2
+ lines Do a DNS lookup immediately upon calling the dnsmgr
+ function, don't wait until a refresh happens. (issue #9097
+ reported by plack) ........
+
+2007-06-07 23:14 +0000 [r68354] Russell Bryant <russell at digium.com>
+
+ * /, main/say.c: Merged revisions 68351 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) |
+ 3 lines Fix a problem where saying a character wouldn't properly
+ break out when the caller pressed '#' (issue #8113, reported by
+ patbaker82, patch from jamesgolovich (hey, long time no see!) and
+ patbaker82) ........
+
+2007-06-07 23:00 +0000 [r68326] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.c: Fix incorrect French syntax of "old
+ messages". Request for feedback was sent to asterisk-dev mailing
+ list, with little response. Issue 9118, patch by junky.
+
+2007-06-07 22:14 +0000 [r68313] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_iax2.c: some improvements to the IAX2 full frame
+ dropping logic recently added: - use inaddrcmp(), since we have
+ it - output the type of frame and subclass being dropped, and the
+ type/subclass that is already being processed (which caused the
+ drop)
+
+2007-06-07 21:16 +0000 [r68280] Russell Bryant <russell at digium.com>
+
+ * channels/chan_agent.c, apps/app_queue.c: Fix loading persistent
+ queue members when using realtime configuration for queues. Also,
+ remove an unneeded leading slash for the astdb family. (issue
+ #9911, patch by atis)
+
+2007-06-07 20:25 +0000 [r68211-68249] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Fix an issue with newer phones which
+ require packets be padded out to the correct length. Issue 9887,
+ patch by DEA.
+
+ * apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4
+ lines Don't try to save voicemail greetings unless the user
+ presses '1' to accept/save. Issue 9904, patch by me. ........
+
+2007-06-07 19:47 +0000 [r68198] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a
+ check to make sure that greetings get stored properly. (Issue
+ 8016, reported by edhorton, patched by alamantia with
+ modification by me. Thanks to Jason Parker for the advice on
+ this).
+
+2007-06-07 19:46 +0000 [r68196] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_features.c: Disable chan_features by default in
+ menuselect
+
+2007-06-07 19:30 +0000 [r68192] Russell Bryant <russell at digium.com>
+
+ * main/strcompat.c: Include stdarg.h for build issues on Solaris
+ (issue #9381)
+
+2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c: Fix logic when doing a name based channel search
+ for a structure when you want to start from a specific point in
+ the channel list. (issue #9324 reported by slavon)
+
+ * apps/app_dial.c, /: Merged revisions 68070 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2
+ lines Allow the 'g' option to work if used with the 'S' option.
+ (issue #9888 reported by gasparz) ........
+
+2007-06-07 10:00 +0000 [r67993-68030] Olle Johansson <oej at edvina.net>
+
+ * res/res_jabber.c: Adding a few Todo's to res_jabber so we don't
+ forget.
+
+ * res/res_jabber.c: Ok, we found out that this is not about if you
+ have any *active* clients using TLS, but if you have initialized
+ TLS at all during the lifetime of the module. So if you reload to
+ disable TLS, it won't help.
+
+ * res/res_jabber.c: If you have a jabber client that uses TLS,
+ refuse unload. Bad fix, but will prevent crashes while we are
+ trying to find a workaround. Iksemel development seems to have
+ stalled and we might have to stop using the TCP/TLS connections
+ in that library and use our own, which would scale better from a
+ poll/select perspective I guess. It would also make it easier to
+ migrate to OpenSSL and stop Asterisk from depending on both
+ OpenSSL and GnuTLS.
+
+ * include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make
+ sure we can unload res_jabber. Patch by phsultan - thanks! Due to
+ a bug in the iksemel library, this will not work if you are using
+ GTLS in the connection. That's being investigated. If you figure
+ out a way to handle that without us having to patch iksemel, let
+ us know in the bug report. Thanks.
+
+2007-06-07 00:10 +0000 [r67924-67941] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2
+ lines Only notify the devicestate system of a peer state change
+ when the peer is built from the config file. (issue #9900
+ reported by arkadia) ........
+
+ * main/file.c: Properly handle cases where a stream can't be
+ written to. (issue #9757 reported by junky)
+
+2007-06-06 22:08 +0000 [r67862-67872] Russell Bryant <russell at digium.com>
+
+ * res/res_snmp.c: Disable reload functionality in res_snmp. It is
+ not possible to initialize the snmp library more than once
+ without completely unloading the module and loading it again.
+ (issue #9571, reported by hristo, additional helpful debug
+ information from festr, patch from me)
+
+ * channels/chan_sip.c: Fix a crash when doing call pickups with SIP
+ phones. The code unlocked the channel when it should not have.
+ (issue #9652, reported by corruptor, fixed by me)
+
+2007-06-06 19:26 +0000 [r67804] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Fix for Issue 9810. There was a segfault
+ under a specific set of circumstances: 1. VoiceMailMain was
+ configured in the dialplan with an extension as its argument 2. A
+ message was left for this mailbox 3. Tried to call VoiceMailMain
+ but hung up before entering password. This was fixed by checking
+ that a pointer was non-null prior to trying to dereference it.
+ (Issue 9810, reported by xmarksthespot, patched by Corydon76 with
+ modifications by me).
+
+2007-06-06 16:55 +0000 [r67716] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, /, include/asterisk/linkedlists.h: Merged
+ revisions 67715 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) |
+ 5 lines We have some bug reports showing crashes due to a double
+ free of a channel. Add a sanity check to ast_channel_free() to
+ make sure we don't go on trying to free a channel that wasn't
+ found in the channel list. (issue #8850, and others...) ........
+
+2007-06-06 13:30 +0000 [r67594-67650] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c, /: Merged revisions 67649 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2
+ lines Reinvite the RTP back to the Asterisk machine when the
+ timeout happens. (issue #9888 reported by gasparz) ........
+
+ * main/translate.c: Fix plc_samples warning when registering a
+ translator. (issue #9897 reported by xylome)
+
+ * apps/app_directed_pickup.c: Include macroexten while searching
+ for a channel to pick up in case they are in a macro. (issue
+ #9491 reported by jamesb63)
+
+ * res/res_agi.c: Make the new "agi debug off" CLI command work.
+ (issue #9890 reported by eliel)
+
+ * /, main/devicestate.c: Merged revisions 67593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2
+ lines Revert channel name splitting fix for Zap. The moral of the
+ story is don't use - in your user/peer names. (issue #9668
+ reported by stevedavies) ........
+
+2007-06-05 23:01 +0000 [r67558] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: Fix some crashes related to the use of the
+ "meetme" CLI command. The code for this command was not locking
+ the conference list at all. (issue #9351, reported by and patch
+ submitted by Junk-Y, committed patch is different and by me)
+
+2007-06-05 21:30 +0000 [r67526] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug
+ 9883, wherein macros were not allowing the includes construct.
+ fixed and tested, looks OK. Now includes can serve as an adjunct
+ to catch.
+
+2007-06-05 20:53 +0000 [r67457-67492] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/linkedlists.h: This bug has been hanging over my
+ head ever since I wrote this SLA code. Every time I tried to go
+ debug it by adding some debug output, the behavior would change.
+ It turns out I wasn't crazy. I had the following piece of code:
+ if (remove) AST_LIST_REMOVE_CURRENT(...); Well,
+ AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my
+ conditional statement didn't do much good at all. It always ran
+ at least all of the macro minus the first statement, so I was
+ seeing list entries magically disappear when they weren't
+ supposed to. After many hours of debugging, I have come to this
+ extremely irritating fix. :) (issues #9581, #9497)
+
+ * channels/chan_zap.c: Suppress a bunch of debug output unless
+ option_debug is on
+
+2007-06-05 18:32 +0000 [r67424] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails
+ saved to IMAP storage using extensions other than gsm were unable
+ to be played over the phone. (Issue 9786, reporter:
+ xmarksthespot, Patched by xmarksthe spot with revisions by me,
+ reviewed by Russell Bryant).
+
+2007-06-05 18:18 +0000 [r67421] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Correctly update date/time on devices
+ throughout the life of the device, instead of just at
+ registration. Issue 9152, yet another patch by DEA.
+
+2007-06-05 18:17 +0000 [r67420] Steve Murphy <murf at digium.com>
+
+ * pbx/pbx_ael.c: Added code to automatically add a default case to
+ switches that don't have one. In some cases, rather than fall
+ thru, it results in a goto with -1 result, which terminates the
+ extension; a sort of dialplan seqfault, sort of. This was
+ required to fix bug reported in 9881
+
+2007-06-05 17:07 +0000 [r67360-67372] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Handle a failure in malloc() in
+ ast_safe_string_alloc()
+
+ * main/channel.c: Fix a problem that showed itself by causing Zap
+ channel names to be completely bogus on my machine.
+ ast_safe_string_alloc() was broken. It called vsnprintf() on a
+ va_args list twice without re-initializing it. After the first
+ usage, va_end() and va_start() must be called again.
+
+2007-06-05 16:14 +0000 [r67329-67334] Christian Richter <christian.richter at beronet.com>
+
+ * /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) |
+ 1 line briding is a bool, fixed copy and paste issue. ........
+
+ * channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05
+ Jun 2007) | 1 line simplified the EVENT_SETUP handling in the
+ cb_events function a lot. Commented the different possibilities a
+ bit and made functions of shared code. When the dialed extension
+ does not exist in the extensions.conf we'll jump into the 'i'
+ extension if this does exist, else we disconnect the call with
+ the cause:1 = No Route to Destination. ........
+
+2007-06-05 15:51 +0000 [r67308] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c, main/loader.c, include/asterisk/module.h: When
+ shutting down "gracefully", go through and run the unload()
+ callbacks for all of the modules. "stop now" is considered a
+ non-graceful shutdown and will not go through this process.
+ (issue #9804, reported by chrisost, patch by me)
+
+2007-06-05 15:22 +0000 [r67304] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: Only muck with the thread structure if an
+ idle one was found/created.
+
+2007-06-05 14:35 +0000 [r67270] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_iax2.c: ensure that a burst of full frames
+ (AST_FRAME_DTMF being the prime example) will not be processed
+ out of order... this is a brute force fix, but seems to be the
+ safest fix for now (thanks to the Digium PQ department for
+ finding this bug)
+
+2007-06-05 10:25 +0000 [r67210] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c, /,
+ channels/misdn/chan_misdn_config.h: Merged revisions 67209 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) |
+ 1 line added possibility to deactivate bridging per port ........
+
+2007-06-04 23:43 +0000 [r67162] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * /, funcs/func_math.c: Merged revisions 67161 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007)
+ | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops.
+ ........
+
+2007-06-04 23:31 +0000 [r67158] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt
+ structure can disappear and the code did not account for it and
+ crashes. (issues #9642, #9569, #9666, probably others ... based
+ on the work by stevedavies and mihai, with additional changes
+ from me)
+
+2007-06-04 23:26 +0000 [r67121-67156] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Fix for skinny keepalives. If there is no
+ traffic from the phone for (keep_alive * 1100) ms (arbitrarily
+ adding 10% for network issues, etc), unregister the device. Issue
+ 8394, patch by DEA.
+
+ * channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar
+ to the recent fix for chan_skinny) Issue 9855, patch by DEA.
+
+2007-06-04 22:28 +0000 [r67119] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Add comments for two functions that get
+ called with the appropriate call locked, but perform operations
+ that could result in the pvt structure getting destroyed before
+ returning again, causing numerous seg faults all over the module.
+ (inspired by issues #9642, #9569, and #9666, and the work done by
+ stevedavies and mihai)
+
+2007-06-04 21:59 +0000 [r67073] Steve Murphy <murf at digium.com>
+
+ * main/cdr.c: This typo has been here since 1.4 forked. It has been
+ the source of heartburn to many a dialplan/CDR programmer.
+
+2007-06-04 21:47 +0000 [r67071] Russell Bryant <russell at digium.com>
+
+ * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC)
+
+2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Better handle SIP devices that say they have
+ SDP content... but really don't. (issue #9398 reported by
+ mthomasslo)
+
+ * apps/app_dial.c: Initialize cidname variable to nothing since it
+ may be used without having been touched. (issue #9661 reported by
+ dimas)
+
+ * res/res_features.c: Returning a value that indicates the parking
+ of a call was a success when it really wasn't (because the
+ parking slot selected was in use) is the wrong thing to do.
+ (issue #9723 reported by mdu113)
+
+2007-06-04 17:11 +0000 [r67061] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * 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.mandrake.zaptel,
+ contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007)
+ | 2 lines Add revision Id tags (by request of tzafrir) ........
+
+2007-06-04 16:02 +0000 [r67026] Russell Bryant <russell at digium.com>
+
+ * configure, configure.ac: Change the configure script to build a
+ test program against libcurl to make sure the results from
+ curl-config can be used to compile successfully. This is intended
+ to help prevent a situation where you are cross compiling, and
+ the configure script finds the curl library installed on the
+ host. (issue #9865, reported and patched by zandbelt)
+
+2007-06-04 15:50 +0000 [r67021] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash
+
+2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When
+ handling an implicit ACK to a frame that was marked as the final
+ transmission for a call, don't call iax2_destroy() for that call
+ while the global frame queue is still locked. There is a very
+ nice explanation of the deadlock in the report. (issue #9663,
+ thorough report and patch from stevedavies, additional positive
+ test reports from mihai and joff_oconnell)
+
+ * include/asterisk/stringfields.h: Fix some compiler warnings in
+ C++ modules. (issue #9866, reported by osk, patch by Corydon76)
+
+2007-06-01 21:45 +0000 [r66919] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * funcs/func_odbc.c: On some drivers, deallocating the statement
+ handle isn't enough. We also have to clear the cursor (nice,
+ Oracle)
+
+2007-06-01 21:31 +0000 [r66897-66917] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Removing extraneous debugging lines from
+ revision 66897. Sorry :)
+
+ * apps/app_voicemail.c: Submitting a fix for voicemail with IMAP
+ storage. Attachments with format specified as gsm were duplicated
+ (i.e. two attachments) were left. Thank you very much to
+ xmarksthespot for submitting the patch that fixed this. (Issues
+ 9787 and 8873, Reported by xmarksthespot and jerjer, patched by
+ xmarksthespot)
+
+2007-06-01 19:41 +0000 [r66879-66881] Russell Bryant <russell at digium.com>
+
+ * channels/chan_skinny.c: Changes to the way DTMF is handled in the
+ core broke dialing in chan_skinny. This patch makes chan_skinny
+ usable again. I did not end up testing this, but there are
+ multiple positive test reports listed in the bug report. (issue
+ #9596, reported by pj, testing by pj and mvanbaak, and the fix
+ was written by DEA)
+
+ * apps/app_page.c: List app_meetme as a module that app_page
+ depends on.
+
+2007-05-31 23:03 +0000 [r66821] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * doc/asterisk.8: Issue 9850 - update preferred command line syntax
+
+2007-05-31 18:41 +0000 [r66775] Russell Bryant <russell at digium.com>
+
+ * res/res_speech.c, include/asterisk/app.h,
+ include/asterisk/speech.h: Change a couple of header files to not
+ use "new", which is a reserved keyword in C++. (issue #9830,
+ reported by osk)
+
+2007-05-31 17:15 +0000 [r66770] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * /, apps/app_macro.c: Merged revisions 66744 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007)
+ | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime.
+ Issue 8329 will remain unfixed for pbx_realtime, but only because
+ we lack core API to do it. ........
+
+2007-05-31 16:14 +0000 [r66768] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2
+ lines It is now possible for this path of execution to have the
+ frame pointer be NULL, therefore we need to check for it before
+ trying to access it. (issue #9836 reported by barthpbx) ........
+
+2007-05-30 23:26 +0000 [r66671] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Fixed seg-faults when recording greetings
+ in voicemail with IMAP enabled. (Issue No. 9735, reported by
+ xmarksthespot, patched by me)
+
+2007-05-30 17:28 +0000 [r66602-66639] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Silly me for having out of date source! Oh
+ well... I'm still leaving my comment.
+
+ * channels/chan_sip.c: When calling some peer/host that may not
+ exist/reply back... don't keep the dialog in memory for all of
+ eternity.
+
+ * channels/chan_zap.c, channels/chan_features.c: Change how channel
+ names are generated a bit. (issue #9825 reported by eldadran)
+
+2007-05-29 21:56 +0000 [r66538] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007)
+ | 2 lines If the value of a variable passed to FIELDQTY is blank,
+ then FIELDQTY should return 0, not 1. ........
+
+2007-05-29 19:32 +0000 [r66474-66503] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Properly handle 408 request timeout -
+ according to the RFC, the dialog dies if a request in a dialog
+ gets this response.
+
+ * channels/chan_sip.c: Don't issue hangup on hangup on hangup on
+ hangup (for jcmoore)
+
+2007-05-29 16:44 +0000 [r66437] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Handle cases where a frame may have no data. (issue
+ #9519 reported by dmb)
+
+2007-05-29 16:07 +0000 [r66404-66414] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Don't reset hangupcause if we already have
+ one
+
+ * channels/chan_sip.c: Tracking down hanging channels, killing them
+ one by one. Issue #9235 and related
+
+2007-05-29 15:43 +0000 [r66398] Joshua Colp <jcolp at digium.com>
+
+ * doc/datastores.txt: Update datastores documentation. (issue #9801
+ reported by mnicholson)
+
+2007-05-29 09:41 +0000 [r66363] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2
+ lines Issue #9802 - Change inuse counter on CANCEL ........
+
+2007-05-28 23:16 +0000 [r66312] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_zap.c: Make the usedistinctiveringdetection option
+ work again. (issue #9823 reported by premeau)
+
+2007-05-27 04:12 +0000 [r66244] Jason Parker <jparker at digium.com>
+
+ * channels/chan_zap.c: I don't know what this was trying to do, but
+ it's clearly incorrect. Issues 9808 and 9809.
+
+2007-05-25 14:43 +0000 [r66160] Kevin P. Fleming <kpfleming at digium.com>
+
+ * configure, configure.ac: have to check for OSP toolkit _after_
+ checking for OpenSSL
+
+2007-05-25 14:41 +0000 [r66159] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * /, main/say.c: Merged revisions 66127 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007)
+ | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch
+ ........
+
+2007-05-25 14:28 +0000 [r66157] Kevin P. Fleming <kpfleming at digium.com>
+
+ * configure, configure.ac, channels/chan_gtalk.c, makeopts.in,
+ res/res_jabber.c: handle the GNUTLS library properly in the
+ configure script and build system don't build in OSP support
+ unless we have found and are allowed to use SSL support
+
+2007-05-24 22:23 +0000 [r66076] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: if the string field init fails, clean up the
+ stuff that was allocated already
+
+2007-05-24 22:16 +0000 [r66074] Joshua Colp <jcolp at digium.com>
+
+ * main/slinfactory.c: Fix slinfactory logic when dealing with
+ frames coming in that may already be in the signed linear format.
+
+2007-05-24 22:07 +0000 [r66068-66070] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Check the result of ast_string_field_init() in
+ ast_channel_alloc()
+
+ * main/rtp.c: Make 1.4 build on my machine, too..
+
+2007-05-24 20:54 +0000 [r66029-66030] Jason Parker <jparker at digium.com>
+
+ * configure: Rebuild configure script for previous ar fix.
+
+ * configure.ac: Following moving strip to AC_PATH_TOOL, we need to
+ do something similar for ar.
+
+2007-05-24 20:42 +0000 [r65978-66026] Russell Bryant <russell at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Checking for the strip application needs to be done with
+ AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross
+ compilation environments.
+
+ * Makefile: Clear CFLAGS before running make for menuselect. (issue
+ #9784, reported by ovi, patch by me)
+
+2007-05-24 18:28 +0000 [r65965-65967] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_gtalk.c: oops, use #ifdef instead of #if
+
+ * channels/chan_gtalk.c: don't reference GnuTLS headers and
+ functions unless the configure script found it
+
+ * main/rtp.c: don't use uninitialized variables
+
+2007-05-24 15:27 +0000 [r65902] Joshua Colp <jcolp at digium.com>
+
+ * main/manager.c: Add the ability to blacklist certain commands
+ from being executed using the Command AMI action. (issue #9240
+ reported by junky)
+
+2007-05-24 15:26 +0000 [r65892-65901] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk
+ core dump since the GnuTLS interface did not support
+ multithreading correctly.
+
+ * channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN.
+ Patch by phsultan. Thanks!
+
+2007-05-24 15:16 +0000 [r65877-65883] Jason Parker <jparker at digium.com>
+
+ * .cleancount: Update cleancount for that last commit - just for
+ good measure.
+
+ * include/asterisk/translate.h, codecs/codec_speex.c,
+ main/translate.c, codecs/codec_ilbc.c: Fix handling of
+ zero-length frames when a codec is capable of native PLC. Issue
+ 9183, patch by Mihai.
+
+2007-05-24 15:08 +0000 [r65866] Dwayne M. Hubbard <dhubbard at digium.com>
+
+ * funcs/func_math.c: merged qwell's func_math patch for issue 9507
+
+2007-05-24 15:08 +0000 [r65863] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: I like it when the RTP stack compiles myself...
+
+2007-05-24 15:05 +0000 [r65857] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues
+ when calling from gtalk to SIP over nat.
+
+2007-05-24 15:04 +0000 [r65842-65853] Russell Bryant <russell at digium.com>
+
+ * apps/app_festival.c: Ensure that frames are fully initialized.
+ This will probably fix getting weird timestamp log messages in
+ logs when using the Festival app. (issue #9781, patch by me)
+
+ * main/rtp.c: Fix the calculation of the RTT for RTCP. The previous
+ code would result in oscillating and incorrect data.
+ Additionally, the RTT would sometimes report negative values due
+ to incorrect calculations. (issue #9601, patch from davetroy)
+
+2007-05-24 14:48 +0000 [r65841] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for
+ jingle
+
[... 7483 lines stripped ...]
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