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git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.4@62251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
6703 lines
286 KiB
Plaintext
6703 lines
286 KiB
Plaintext
2007-04-27 Russell Bryant <russell@digium.com>
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* Asterisk 1.4.4 released.
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2007-04-27 21:10 +0000 [r62218] Russell Bryant <russell@digium.com>
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* channels/chan_agent.c: Fix a weird problem where when a caller
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talking to someone sitting behind an agent channel sent a digit,
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the digit would be played to the agent for forever. This is
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because chan_agent always returned -1 from its send_digit_begin
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and _end callbacks. This non-zero return value indicates to the
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Asterisk core that it would like an inband DTMF generator put on
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the channel. However, this is the wrong thing to do. It should
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*always* return 0, instead. When the digit begin and end
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functions are called on the proxied channel, the underlying
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channel will indicate whether inband DTMF is needed or not, and
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the generator will be put on that one, and not the Agent channel.
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(issue #9615, #9616, reported by jiddings and BigJimmy, and fixed
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by me)
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2007-04-27 16:17 +0000 [r62174] Jason Parker <jparker@digium.com>
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* /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3
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lines This transcoder message needn't be a NOTICE. I've seen it
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cause confusion more than a few times. ........
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2007-04-27 16:14 +0000 [r62171] Russell Bryant <russell@digium.com>
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* main/pbx.c: If no variables were passed into
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pbx_substitute_variables_helper_full(), then don't even bother
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creating a temporary bogus channel, since that is only for
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allowing certain functions to operate on the variables as if they
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were on a channel. Most importantly, this fixes a crash. (issue
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#9613, reported by callguy, fixed by me)
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2007-04-27 14:04 +0000 [r62095-62137] Olle Johansson <oej@edvina.net>
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* /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4
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lines Issue #7351 - SIP Cancel fails due to the wrong contact
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uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka
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- THANKS!!!! THis was a hard one to catch. ........
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* channels/chan_zap.c, main/manager.c: Issue #9608 - fix some
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annoying DEBUG messages not controlled by option_debug (DEA).
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Thanks!
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2007-04-26 16:33 +0000 [r61959-62038] Joshua Colp <jcolp@digium.com>
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* /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2
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lines Revert previous fix for when the IAX2 channel goes funky
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(that's the technical term). This is causing legit calls to be
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prematurely hung up. (issue #9600 reported by justdave) ........
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* main/channel.c: Missed an ast_app_group_discard during merge.
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Thanks blitzrage!
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* res/res_monitor.c: Don't always say that the channel is being
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paused if it is actually being unpaused in the Manager ack
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message. (reported by jsmith in #asterisk-bugs)
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* main/config.c, /: Merged revisions 61958 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2
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lines Don't count failed include attempts against the
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configuration include level. (issue #9593 reported by mostyn)
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........
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2007-04-25 22:29 +0000 [r61914] Kevin P. Fleming <kpfleming@digium.com>
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* channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007)
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| 2 lines handle a very bizarre race condition with channels
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being redirected before a simple switch can be started on them
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(issue #9286) ........
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2007-04-25 21:59 +0000 [r61863-61870] Russell Bryant <russell@digium.com>
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* /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) |
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2 lines If the callerid= option is specified, but empty, clear
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any previous data. ........
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* /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) |
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2 lines Ensure that callerid settings are reset on a reload.
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........
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2007-04-25 19:21 +0000 [r61805] Joshua Colp <jcolp@digium.com>
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* main/cli.c, main/channel.c, include/asterisk/app.h,
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funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804
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via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2
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lines Merge rewritten group counting support. No more storing
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data on the variable list of the channels. That was bad, mmmk?
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(issue #7497 reported by sabbathbh) ........
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2007-04-25 16:22 +0000 [r61799] Russell Bryant <russell@digium.com>
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* channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) |
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3 lines Fix a typo where cid_num got copied instead of cid_ani.
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(issue #9587, reported and patched by xrg) ........
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2007-04-24 Russell Bryant <russell@digium.com>
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* Asterisk 1.4.3 released.
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2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant <russell@digium.com>
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* main/manager.c, /: Merged revisions 61786 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) |
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4 lines Don't crash if a manager connection provides a username
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that exists in manager.conf but does not have a password, and
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also requests MD5 authentication. (ASA-2007-012) ........
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* main/channel.c, include/asterisk/channel.h: Improve DTMF handling
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in ast_read() even more in response to a discussion on the
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asterisk-dev mailing list. I changed the enforced minimum length
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of a digit from 100ms to 80ms. Furthermore, I made it now enforce
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a gap of 45ms in between digits. These values are not
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configurable in a configuration file right now, but they can be
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easily changed near the top of main/channel.c.
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2007-04-24 18:43 +0000 [r61779] Dwayne M. Hubbard <dhubbard@digium.com>
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* channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007)
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| 1 line removed #if 0 block from chan_phone, chan_zap, and
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chan_modem restart_monitor() ........
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2007-04-24 16:16 +0000 [r61774] Russell Bryant <russell@digium.com>
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* main/dial.c: Add a few more state changes in
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handle_frame_ownerless() so that the SLA code will get notified
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of these changes even when an owner channel is not provided. This
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isn't from a specific bug report, it's just something I noticed
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while poking around.
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2007-04-24 16:07 +0000 [r61772] Joshua Colp <jcolp@digium.com>
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* /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2
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lines Allow RFC2833 to be sent in the response SDP when an INVITE
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comes in without SDP. (issue #9546 reported by mcrawford)
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........
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2007-04-23 18:17 +0000 [r61763-61765] Russell Bryant <russell@digium.com>
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* main/pbx.c: Some dialplan functions, such as CUT(), expect to
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operate on variables on a channel. So, this little hack lets them
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work in places where a channel doesn't exist, such as within
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DUNDi configuration. (issue #9465, reported and patched by
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Corydon76, testing by blitzrage)
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* main/channel.c: Ensure that digits passing through Asterisk have
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a reasonable minimum length. It is currently 100 ms. If someone
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thinks this should be different, feel free to speak up. (related
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to issues #8944, #9250, and #9348)
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2007-04-20 21:35 +0000 [r61705-61707] Jason Parker <jparker@digium.com>
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* main/rtp.c: Avoid invalid seqno cycling detection. Per comment
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from Dave Troy: This adds back in some simple typecasting I had
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in an earlier version which I realize now may be breaking things.
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Issue #9554.
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* main/loader.c, /: Merged revisions 61704 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4
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lines Fix an issue that I noticed while looking over issue 9571.
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The reload timestamp was getting set after reloading the built-in
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stuff, and before the modules. ........
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2007-04-20 20:42 +0000 [r61697] Russell Bryant <russell@digium.com>
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* main/rtp.c: Remove a stray debug message introduced by a recent
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commit.
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2007-04-20 19:51 +0000 [r61694] Jason Parker <jparker@digium.com>
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* /, apps/app_queue.c: Merged revisions 61692 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5
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lines If the '* to hangup' option is not enabled, we don't need
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to disable * as a valid exit key. If it was enabled, this
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statement would've never been checked in the first place. Issue
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#9552 ........
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2007-04-20 18:19 +0000 [r61690] Russell Bryant <russell@digium.com>
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* main/config.c, apps/app_voicemail.c, main/manager.c,
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include/asterisk/config.h: Fix the UpdateConfig manager action to
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properly treat "variables" and "objects" differently (a=b versus
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a=>b). (issue #9568, reported by pari, patch by me)
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2007-04-19 08:37 +0000 [r61686] Olle Johansson <oej@edvina.net>
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* /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3
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lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by
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Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........
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2007-04-19 04:36 +0000 [r61681-61683] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
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* main/manager.c: Bug 9557 - simple reason why reading a function
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always returned NULL
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* funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c,
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funcs/func_groupcount.c, /, funcs/func_timeout.c,
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funcs/func_cdr.c: Merged revisions 61680 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007)
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| 5 lines Bug 9557 - Specifying the GetVar AMI action without a
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Channel parameter can cause Asterisk to crash. The reason this
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needs to be fixed in the functions instead of in AMI is because
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Channel can legitimately be NULL, such as when retrieving global
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variables. ........
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2007-04-18 22:10 +0000 [r61678] Kevin P. Fleming <kpfleming@digium.com>
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* sounds/Makefile: allow external build systems to extract the
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required sound file versions
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2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson <oej@edvina.net>
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* main/rtp.c: Clean upp formatting, add some doxygen stuff while
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we're in cleaning mode... Thanks Kevin!
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* main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy)
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2007-04-16 14:47 +0000 [r61664-61666] Olle Johansson <oej@edvina.net>
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* channels/chan_sip.c: #9483, half of patch by twilson to solve 302
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redirect issues
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* /: Blocking AstHoloPatch from 1.2
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2007-04-13 21:17 +0000 [r61658] Steve Murphy <murf@digium.com>
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* main/cdr.c: This is a fix to the way CDR merge handles the data
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that results from ForkCDR.
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2007-04-13 19:17 +0000 [r61648-61656] Joshua Colp <jcolp@digium.com>
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* apps/app_dial.c, /: Merged revisions 61655 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2
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lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves
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the same as OUTBOUND_GROUP except it will get unset after use so
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it won't get accidentally inherited. (issue #BE-140) ........
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* apps/app_speech_utils.c: Do not bother looking for a result if
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none are present.
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* channels/chan_sip.c: For those very verbose SIP implementations
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that attach tons of info to the Contact header... let's increase
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our variable sizes. (issue #9535 reported by jeffg)
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2007-04-13 17:10 +0000 [r61645] Russell Bryant <russell@digium.com>
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* apps/app_voicemail.c: Eliminate a compiler warning with
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ODBC_STORAGE enabled so that it will build under dev-mode.
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2007-04-13 17:01 +0000 [r61644] Steve Murphy <murf@digium.com>
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* channels/chan_oss.c: A fix for chan_oss that resulted from the
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CDR changes; it helps to use the right info.
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2007-04-13 16:32 +0000 [r61641] Joshua Colp <jcolp@digium.com>
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* channels/chan_sip.c: Don't assume the callid of a dialog will be
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set, as in some circumstances it may not. (issue #9534 reported
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by tecnoxarxa)
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2007-04-11 16:05 +0000 [r61477] Russell Bryant <russell@digium.com>
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* /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) |
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5 lines If someone sets the "useragent" option in sip.conf to be
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empty, then don't add the User-Agent header at all. It is an
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optional header, anyway. Also, the bug report says that some of
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Japan's SIP providers don't allow it for some weird reason.
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(issue #9488, reported by makoto, fixed by me) ........
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2007-04-11 15:39 +0000 [r61443] Nadi Sarrar <ns@beronet.com>
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* channels/chan_misdn.c: Don't export AOCD variables on
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misdn_hangup anymore, this was mainly a fix for trunk..
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2007-04-11 15:09 +0000 [r61377-61427] Russell Bryant <russell@digium.com>
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* /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) |
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6 lines Fix a bug with switching between host=dynamic and using
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specific hosts for peers. The code would only reset the peer's
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address when it is dynamic if it was a new peer structure. Now,
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it will also reset the address if it was already in the peer
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list, but before the reload, it was not dynamic. (issue #9515,
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reported by caio1982, fixed by me) ........
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* main/http.c: Add "svgz" to the mimetypes table. (issue #9510,
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bkruse) In passing, constify the elements of the mimetypes table.
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* /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) |
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5 lines Remove the attempt at reporting configuration errors in
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sip.conf. This can cause a bunch of improper messages when using
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realtime. I give up. As oej tried to convince me when I put this
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in, there is just no easy way to do it. (inspired by a message on
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the -dev list) ........
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2007-04-11 13:40 +0000 [r61342-61373] Nadi Sarrar <ns@beronet.com>
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* channels/chan_misdn.c: Export AOCD variables on misdn_hangup.
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* channels/chan_misdn.c: Ignore facility messages in case we don't
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have a corresponding channel object.
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* channels/chan_misdn.c: AOCD's are now exported to asterisk
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channel variables.
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2007-04-10 16:05 +0000 [r61220] Russell Bryant <russell@digium.com>
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* main/Makefile, main/http.c, main/minimime (removed): File upload
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support was added to solve some needs for the Asterisk GUI.
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However, after much discussion, it has been decided that adding
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this to 1.4 is not in the best interests of the project. It has
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been removed here, but will remain in trunk.
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2007-04-10 12:43 +0000 [r61183] Nadi Sarrar <ns@beronet.com>
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* channels/misdn_config.c, /: Merged revisions 61170 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/branches/1.2
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........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr
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2007) | 2 lines msns config parameter defaults to '*' ........
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2007-04-10 05:18 +0000 [r61136] Steve Murphy <murf@digium.com>
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* apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a
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previous fix to overcome a compiler warning; the app NoCDR() has
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been updated to mark the channel CDR as POST_DISABLED instead of
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destroying the CDR; this way its flags are propagated thru a
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bridge and the CDR is actually dropped. The cases where only one
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channel in a bridge has a CDR was cleaned up.
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2007-04-09 19:58 +0000 [r61072] Olle Johansson <oej@edvina.net>
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* /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3
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lines - Don't send ActionID before Response: header. - Don't use
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a blank in an AMI header ........
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2007-04-09 19:55 +0000 [r61062-61070] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/minimime/mm_envelope.c, res/res_features.c: fix up some
|
|
warnings found using --enable-dev-mode
|
|
|
|
* main/minimime/Doxyfile (removed),
|
|
main/minimime/tests/messages/CVS (removed),
|
|
main/minimime/tests/CVS (removed): remove some more stuff we
|
|
don't need
|
|
|
|
2007-04-09 19:41 +0000 [r61042-61044] Russell Bryant <russell@digium.com>
|
|
|
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* main/minimime/test (removed): Remove another directory that
|
|
should no longer be there
|
|
|
|
* main/minimime/Make.conf (removed), main/minimime/mytest_files
|
|
(removed), main/minimime/.cvsignore (removed), main/minimime/sys
|
|
(removed), main/minimime/mm-docs (removed): Remove various files
|
|
that I thought I already removed.
|
|
|
|
2007-04-09 19:05 +0000 [r61022] Jason Parker <jparker@digium.com>
|
|
|
|
* apps/app_queue.c: Use the appropriate interface name with
|
|
COMPLETECALLER. Issue 9395.
|
|
|
|
2007-04-09 18:32 +0000 [r60989] Steve Murphy <murf@digium.com>
|
|
|
|
* channels/chan_oss.c, main/channel.c, main/cdr.c,
|
|
channels/chan_phone.c, channels/chan_misdn.c,
|
|
channels/chan_skinny.c, channels/chan_features.c,
|
|
channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c,
|
|
channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
|
|
channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c,
|
|
channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
|
|
include/asterisk/channel.h, channels/chan_gtalk.c,
|
|
channels/chan_iax2.c: This is a big improvement over the current
|
|
CDR fixes. It may still need refinement, but this won't have as
|
|
many folks bothered.
|
|
|
|
2007-04-09 18:02 +0000 [r60984] Olle Johansson <oej@edvina.net>
|
|
|
|
* res/res_jabber.c: Add final new line after JabberEvent
|
|
|
|
2007-04-09 17:22 +0000 [r60936] Jason Parker <jparker@digium.com>
|
|
|
|
* /, apps/app_directory.c: Merged revisions 60935 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5
|
|
lines Allow matching on names shorter than 3 chars. This also
|
|
fixes the case where somebody wants to match on less then 3
|
|
chars. Issue 9071 ........
|
|
|
|
2007-04-09 03:01 +0000 [r60847-60850] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/asterisk.c, include/asterisk.h, /: Merged revisions 60849
|
|
via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007)
|
|
| 2 lines Don't check for error when lowering priority (according
|
|
to the manpage, it should never happen anyway). It might could
|
|
happen, though, if another thread messed with the priority, so
|
|
safeguard against that (reported via -dev list). ........
|
|
|
|
* channels/chan_local.c, /: Merged revisions 60846 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08
|
|
Apr 2007) | 2 lines Bug 9505 - If the return value for
|
|
local_queue_frame is set, then p->lock is no longer valid.
|
|
........
|
|
|
|
2007-04-09 01:03 +0000 [r60762-60798] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_dial.c, /: Merged revisions 60797 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2
|
|
lines When calling a device that then forwards us elsewhere... we
|
|
have to make our channels compatible if it is the only channel
|
|
being dialed. (issue #9445 reported by marcelbarbulescu) ........
|
|
|
|
* apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if
|
|
MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy)
|
|
|
|
2007-04-08 14:14 +0000 [r60661-60713] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /, apps/app_macro.c: Merged revisions 60711 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007)
|
|
| 2 lines Gosub called within a Macro resets the arguments
|
|
improperly and causes general weirdness. (Issue 8329) ........
|
|
|
|
* main/http.c: Fix --enable-dev-mode
|
|
|
|
* channels/chan_oss.c: Off by one error, resulting in a crash
|
|
(Issue 9500)
|
|
|
|
* /, main/file.c: Merged revisions 60660 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007)
|
|
| 2 lines Bug 9486 - memory leak when opening a filestream
|
|
........
|
|
|
|
2007-04-06 20:58 +0000 [r60603] Russell Bryant <russell@digium.com>
|
|
|
|
* main/minimime/sys/mm_queue.h, main/minimime/Doxyfile,
|
|
main/minimime/mimeparser.yy.c, main/minimime/minimime.c,
|
|
main/manager.c, main/minimime/mm_mimepart.c,
|
|
main/minimime/test.sh, configure, include/asterisk/compat.h,
|
|
main/strcompat.c, main/minimime/mm_internal.h, main/http.c,
|
|
main/minimime/tests/parse.c, main/minimime/mm_base64.c,
|
|
main/minimime/mm_mimeutil.c, main/minimime/mm.h,
|
|
main/minimime/tests, main/minimime/mm_header.c,
|
|
main/minimime/mm_error.c, main/Makefile,
|
|
main/minimime/mm_codecs.c, main/minimime/mm_param.c,
|
|
configure.ac, main/minimime/Makefile, main/minimime/mm_init.c,
|
|
include/asterisk/manager.h, main/minimime/strlcpy.c,
|
|
configs/http.conf.sample, main/minimime/mm_parse.c,
|
|
main/minimime/tests/create.c, main/minimime/mm_contenttype.c,
|
|
main/minimime/mm_util.c, main/minimime/mm_envelope.c,
|
|
main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c,
|
|
main/minimime/tests/messages/test2.txt,
|
|
main/minimime/tests/messages/test3.txt,
|
|
main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c,
|
|
main/minimime/tests/messages/test4.txt,
|
|
main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h,
|
|
main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c,
|
|
main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt,
|
|
main/minimime/mimeparser.l, main/minimime/mm_context.c,
|
|
main/minimime/mimeparser.tab.h, main/minimime (added),
|
|
main/minimime/mm_warnings.c, main/minimime/mm_queue.h,
|
|
main/minimime/tests/messages, include/asterisk/autoconfig.h.in,
|
|
main/minimime/mimeparser.y, Makefile.moddir_rules,
|
|
main/minimime/sys, main/minimime/tests/Makefile: To be able to
|
|
achieve the things that we would like to achieve with the
|
|
Asterisk GUI project, we need a fully functional HTTP interface
|
|
with access to the Asterisk manager interface. One of the things
|
|
that was intended to be a part of this system, but was never
|
|
actually implemented, was the ability for the GUI to be able to
|
|
upload files to Asterisk. So, this commit adds this in the most
|
|
minimally invasive way that we could come up with. A lot of work
|
|
on minimime was done by Steve Murphy. He fixed a lot of bugs in
|
|
the parser, and updated it to be thread-safe. The ability to
|
|
check permissions of active manager sessions was added by Dwayne
|
|
Hubbard. Then, hacking this all together and do doing the
|
|
modifications necessary to the HTTP interface was done by me.
|
|
|
|
2007-04-06 20:32 +0000 [r60568-60572] Dwayne M. Hubbard <dhubbard@digium.com>
|
|
|
|
* UPGRADE.txt: clarified a sentence in the format_wav section
|
|
|
|
* UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and
|
|
plan to remove GAIN code from trunk
|
|
|
|
2007-04-06 19:50 +0000 [r60521-60565] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_meetme.c: When a station picks up a trunk that was on
|
|
hold, make the hints reflect that nobody has the trunk on hold
|
|
anymore.
|
|
|
|
* apps/app_meetme.c: Fix a few problems with SLA. (issue #9459,
|
|
reported by francesco_r, fixed by me) * The original behavior was
|
|
that if one station put a call on hold, another one picked it up,
|
|
and then hung up, the code would still consider the call on hold
|
|
by the first station, so the trunk would not be hung up. However,
|
|
to better comply with what most people seem to expect it to
|
|
behave, it will now hang up the trunk. * Fix a problem with
|
|
"barge=no". This was only intended to prevent people from joining
|
|
calls that are in progress. However, it also prevented other
|
|
people from picking up a call that was on hold. This has been
|
|
fixed. * When there are no active stations on a trunk and it is
|
|
on hold, the code now indicates the HOLD and UNHOLD conditions to
|
|
the trunk channel. This allows music on hold to be played to the
|
|
trunk when it is on hold.
|
|
|
|
2007-04-06 18:21 +0000 [r60459-60485] Matt Frederickson <creslin@digium.com>
|
|
|
|
* channels/chan_zap.c: Make sure we check the faxdetect option
|
|
before doing fax processing
|
|
|
|
* channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2
|
|
lines There should only be one code path for doing DTMF
|
|
conditionals on channels. This fixes it. ........
|
|
|
|
2007-04-06 14:49 +0000 [r60399] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007)
|
|
| 2 lines remove undocumented 'cardsmode' parameter and stop
|
|
searching for transcoders during reload() ........
|
|
|
|
2007-04-06 01:14 +0000 [r60361] Joshua Colp <jcolp@digium.com>
|
|
|
|
* res/res_speech.c, apps/app_speech_utils.c,
|
|
include/asterisk/speech.h: Add support for returning different
|
|
types of results (ie: NBest).
|
|
|
|
2007-04-05 22:58 +0000 [r60325] Dwayne M. Hubbard <dhubbard@digium.com>
|
|
|
|
* formats/format_wav.c: modified default GAIN for issue 5823,
|
|
thanks jrwalliker
|
|
|
|
2007-04-05 22:35 +0000 [r60323] Steve Murphy <murf@digium.com>
|
|
|
|
* configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added
|
|
some clarification to the example configs for CDRs, on how to
|
|
select a backend. Also, made cdr-csv the default if you 'make
|
|
samples', and no other changes.
|
|
|
|
2007-04-05 16:10 +0000 [r60268] Jason Parker <jparker@digium.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5
|
|
lines Just because we can't find the voicemail configuration
|
|
file, doesn't mean that the module failed to load. The user could
|
|
be using realtime. Issue #9473 ........
|
|
|
|
2007-04-05 15:47 +0000 [r60265] Russell Bryant <russell@digium.com>
|
|
|
|
* main/http.c: Add the MIME type for gif by request from Pari
|
|
|
|
2007-04-05 12:55 +0000 [r60214] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2
|
|
lines Only unlock our pvt and net locks if we are actually going
|
|
to try to lock the owner again. (issue #9472 reported by zoa)
|
|
........
|
|
|
|
2007-04-04 17:40 +0000 [r60013-60137] Russell Bryant <russell@digium.com>
|
|
|
|
* main/manager.c, /: Merged revisions 60134 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) |
|
|
6 lines It is valid to redirect channels via the manager
|
|
interface that are not in the UP state. Instead of checking for
|
|
that to prevent to ensure a dead channel doesn't get redirected,
|
|
just use the ast_check_hangup() API call. (issue #9457, reported
|
|
by Callmewind, patch by me) (related to issue #8977) ........
|
|
|
|
* channels/chan_sip.c: Add a Content-Length of 0 to the response
|
|
built by transmit_response_with_unsupported(). (issue #9454,
|
|
reported by makoto, fixed by me)
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) |
|
|
4 lines Fix the return value of handle_common_options() so that
|
|
it always properly indicates whether it handled the option or
|
|
not. (issue #9455, reported by Netview, fixed by me) ........
|
|
|
|
* apps/app_meetme.c: Fix a problem where if a trunk was hung up
|
|
while it was on hold, all of the hints would reflect the line
|
|
still on hold, even though it should reflect that it is back to
|
|
not in use. (issue #9459, reported by francesco_r, fixed by me)
|
|
|
|
* /: Blocked revisions 60016 via svnmerge ........ r60016 | russell
|
|
| 2007-04-03 18:23:23 -0500 (Tue, 03 Apr 2007) | 3 lines Add a
|
|
missing "\r\n" in the body of the NOTIFY that is sent to indicate
|
|
the status of a transfer. (issue #9388, reported by rarritt)
|
|
........
|
|
|
|
* /: Blocked revisions 60014 via svnmerge ........ r60014 | russell
|
|
| 2007-04-03 18:00:10 -0500 (Tue, 03 Apr 2007) | 3 lines Use the
|
|
more generic check for "sed -r" support that was already present
|
|
in 1.4. (related to issue #9399) ........
|
|
|
|
* /: Blocked revisions 60012 via svnmerge ........ r60012 | russell
|
|
| 2007-04-03 17:54:49 -0500 (Tue, 03 Apr 2007) | 3 lines On
|
|
Darwin, the -r argument to sed is not valid. It has to be -E.
|
|
(issue #9399, reported by jcovert) ........
|
|
|
|
2007-04-03 19:40 +0000 [r59963] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_speech_utils.c: Don't clash when a person both speaks
|
|
and uses DTMF.
|
|
|
|
2007-04-03 19:16 +0000 [r59853-59939] Russell Bryant <russell@digium.com>
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) |
|
|
4 lines Don't attempt to report configuration errors in
|
|
build_user(). oej pointed out that for a "friend" entry, this
|
|
won't work, because all user options are valid for peers, but not
|
|
the other way around. ........
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) |
|
|
3 lines Make chan_sip report when it encounters an unknown
|
|
option. (issue #9440, reported by nightcrawler) ........
|
|
|
|
* /, main/app.c: Merged revisions 59886 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) |
|
|
5 lines When doing a built-in blind or attended transfer, restore
|
|
the ability to use '#' to terminate the number and immediately do
|
|
the transfer instead of having to dial the number and just wait
|
|
for the feature digit timeout. (issue #8366, xueliangliang)
|
|
........
|
|
|
|
* Makefile: Ensure that menuselect gets executed in dependency
|
|
check mode every time you run make.
|
|
|
|
2007-04-03 11:02 +0000 [r59804] Nadi Sarrar <ns@beronet.com>
|
|
|
|
* channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h:
|
|
Merged revisions 59788,59803 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2
|
|
lines Use the new sysfs way of mISDN 1.2 to check if a port is NT
|
|
or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di,
|
|
03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........
|
|
|
|
2007-04-03 07:20 +0000 [r59774] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
|
|
channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h:
|
|
Merged revisions 59623-59624,59639 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) |
|
|
1 line we can now make 30 channels on a PRI (before we forgot
|
|
chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200
|
|
(Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........
|
|
r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) |
|
|
1 line added option which allows us to accept incoming SETUP
|
|
Messages without automatically sending Proceeding or Setup
|
|
Acknowledge, this is useful with some broken switches and if you
|
|
want to Release incoming calls without previously having
|
|
acknowledged them. The new option is
|
|
noautorespond_on_setup=yes|no default is no, so we don't break
|
|
the existing behaviour ........
|
|
|
|
2007-04-02 18:58 +0000 [r59724] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2
|
|
lines Increase the maximum size for a string of mailboxes to
|
|
1024. (issue #9270 reported by rtucker) ........
|
|
|
|
2007-04-02 17:31 +0000 [r59688] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/pbx_ael.c: continue in for-loop should go to the incrementer,
|
|
not the test. As per 9435, thanks to marcelbarbulescu
|
|
|
|
2007-04-02 15:39 +0000 [r59654] Russell Bryant <russell@digium.com>
|
|
|
|
* main/netsock.c, /: Merged revisions 59608 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) |
|
|
6 lines Add the SO_REUSEADDR flag to sockets handled by netsock.
|
|
This is needed by the patch that went in for issue 7874.
|
|
chan_iax2 needs to be able to create socket that is lisetning on
|
|
INADDR_ANY, but also be able to bind sockets to specific
|
|
addresses. (Thanks to Stevenson on the asterisk-dev mailing list
|
|
for explaining why this flag was needed.) ........
|
|
|
|
2007-03-30 22:50 +0000 [r59573] Jason Parker <jparker@digium.com>
|
|
|
|
* configure, main/Makefile, acinclude.m4: Add linux-uclibc host
|
|
arch..."thingy". Sorry, I don't know what it's called...
|
|
|
|
2007-03-30 17:51 +0000 [r59452-59522] Steve Murphy <murf@digium.com>
|
|
|
|
* main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
|
|
include/asterisk/cdr.h: several changes via kpflemings review
|
|
|
|
* main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
|
|
include/asterisk/cdr.h: These mods fix CDR issues from 8221,
|
|
8593, 8680, 8743, and perhaps others. Mainly with CDRs generated
|
|
from transfer situations.
|
|
|
|
* configs/extensions.conf.sample: A small clarification to keep
|
|
bugs from being filed, and confusion from rising, if
|
|
clearglobalvars is set, and globals are set in the AEL file.
|
|
(9419)
|
|
|
|
2007-03-29 17:43 +0000 [r59363] Russell Bryant <russell@digium.com>
|
|
|
|
* res/res_jabber.c: When building a response to a subscription, the
|
|
"from" must be the full Jabber ID. This fixes some problems where
|
|
jabber users are not able to add their Asterisk account to their
|
|
user list, since they are unable to get Asterisk to approve their
|
|
subscription. (issue #8210, reported by caspy, and verified by
|
|
bradtem)
|
|
|
|
2007-03-29 17:38 +0000 [r59361] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2
|
|
lines Keep a global array of variables indicating whether certain
|
|
conference rooms are in use. This ensures that two people going
|
|
into a new dynamic conference when the 'e' option is set don't go
|
|
into the same conference room. (issue #8835 reported by eliel)
|
|
........
|
|
|
|
2007-03-29 17:17 +0000 [r59304-59358] Russell Bryant <russell@digium.com>
|
|
|
|
* main/rtp.c, /: Merged revisions 59357 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) |
|
|
5 lines If an error occurs when reading from an RTP socket, and
|
|
the error code does not indicate that we should try again, then
|
|
return NULL instead of a "null frame". This will prevent Asterisk
|
|
from trying over and over again, and eventually causing the
|
|
system to crash. (issue #8285, john) ........
|
|
|
|
* /: Blocked revisions 59355 via svnmerge ........ r59355 | russell
|
|
| 2007-03-29 12:10:28 -0500 (Thu, 29 Mar 2007) | 3 lines Backport
|
|
the change to chan_iax2 to return NULL instead of a "null frame"
|
|
from its read callback. See revision 59341 to the 1.4 branch for
|
|
more info. ........
|
|
|
|
* channels/chan_iax2.c: When the IAX2 read callback gets called,
|
|
return NULL instead of a "null frame". This will cause Asterisk
|
|
to hangup the call instead of keep trying whatever it was doing.
|
|
Under normal conditions, this function would *never* be called.
|
|
However, the author of this patch says an error will occur that
|
|
will cause it to get called every 100 thousand calls or so. When
|
|
this does happen, it puts the channel in a loop that eventually
|
|
brings down the system. So, hangup up the call is certainly a
|
|
better alternative. (issue #8286, john)
|
|
|
|
* Makefile: Export the GTK2 library and include information to sub
|
|
Makefiles.
|
|
|
|
2007-03-29 16:07 +0000 [r59300-59302] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007)
|
|
| 3 lines Issue 9415 - No point to getting a diagnostic field if
|
|
we aren't doing anything with the information. (Plus, it tends to
|
|
crash the Postgres ODBC driver.) ........
|
|
|
|
* /: Blocked revisions 59299 via svnmerge ........ r59299 |
|
|
tilghman | 2007-03-29 10:33:10 -0500 (Thu, 29 Mar 2007) | 2 lines
|
|
Change ENV section to use setenv, instead of putenv (Alexandru
|
|
Pirvulescu <sigxcpu@gmail.com>, reported via -dev list) ........
|
|
|
|
2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* res/res_odbc.c: Another crash that I thought we had fixed already
|
|
- Issue 9396
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007)
|
|
| 2 lines Oops ........
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007)
|
|
| 2 lines Fix a few remaining bad mmap(2) return values ........
|
|
|
|
2007-03-27 23:20 +0000 [r59262-59278] Russell Bryant <russell@digium.com>
|
|
|
|
* /, apps/app_directory.c: Merged revisions 59277 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) |
|
|
3 lines Fix the check of the return value from mmap(). Thanks to
|
|
Corydon for catching this one. ........
|
|
|
|
* apps/app_directory.c: Fix app_directory to actually compile with
|
|
ODBC_STORAGE, and update the code to the latest res_odbc API.
|
|
|
|
* apps/Makefile: Fix app_directory when ODBC_STORAGE is being used.
|
|
The Makefile did not properly ensure that this information got
|
|
copied from what was selected for app_voicemail. (issue #9224)
|
|
|
|
* channels/chan_sip.c: Fix the check that ensures that the CHANNEL
|
|
function's first argument is "rtpqos". Thanks, Corydon. :)
|
|
|
|
2007-03-27 18:16 +0000 [r59261] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes
|
|
asterisk), kpfleming pointed on asterisk-dev, that DECLINE in
|
|
this case the proper thing to do. This change now has it doing
|
|
the proper thing.
|
|
|
|
2007-03-27 18:05 +0000 [r59256-59259] Russell Bryant <russell@digium.com>
|
|
|
|
* /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) |
|
|
4 lines Fix the use of the "sourceaddress" option when "bindaddr"
|
|
is set to 0.0.0.0 instead of having each interface explicitly
|
|
listed. (issue #7874, patch by stevens) ........
|
|
|
|
* channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS
|
|
function to just be additional parameter of the CHANNEL function.
|
|
This way, it will be possible for other RTP based channel drivers
|
|
to expose this information in the future.
|
|
|
|
2007-03-27 15:00 +0000 [r59254] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27
|
|
Mär 2007) | 1 line fixed #9355 ........
|
|
|
|
2007-03-26 21:45 +0000 [r59230] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* channels/chan_sip.c: Oops, this should be case insensitive
|
|
|
|
2007-03-26 21:41 +0000 [r59228] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes
|
|
asterisk). I turned a duplicate context from a WARNING to an
|
|
ERROR. Now you get a module load failure, and asterisk just
|
|
exits. That's better than a crash, right\?
|
|
|
|
2007-03-26 21:37 +0000 [r59227] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* channels/chan_sip.c: Change this to a single dp function to make
|
|
oej happy.
|
|
|
|
2007-03-26 20:06 +0000 [r59225] Steve Murphy <murf@digium.com>
|
|
|
|
* main/config.c: Fix for 9257; by eliminating the globals in
|
|
main/config.c, we make it thread-safe, which is a minimum
|
|
requirement.
|
|
|
|
2007-03-26 19:34 +0000 [r59223] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_speech_utils.c: Add ability to specify no timeout. This
|
|
means as soon as the prompt is done playing it moves on to the
|
|
next priority.
|
|
|
|
2007-03-26 18:33 +0000 [r59215-59217] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_voicemail.c: Somehow the code for building the email for
|
|
voicemail got out of sync. This change makes a few tweaks to get
|
|
1.4 in sync with trunk. (issue #9301)
|
|
|
|
* apps/app_meetme.c: Fix some codec negotiation problems when
|
|
CallerID support is not enabled in SLA. (issue #9308, reported by
|
|
twilson)
|
|
|
|
2007-03-26 18:13 +0000 [r59213] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_speech_utils.c: Make SpeechBackground obey the digit
|
|
timeout value.
|
|
|
|
2007-03-26 17:53 +0000 [r59207-59209] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_sip.c: Rename the new dialplan functions to match
|
|
the variable name
|
|
|
|
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The
|
|
AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in
|
|
some because they get set in sip_hangup. So, there are common
|
|
situations where the variables will not be available in the
|
|
dialplan at all. So, this patch provides an alternate method for
|
|
getting to this information by introducing AUDIORTPQOS and
|
|
VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76,
|
|
with some testing by blitzrage)
|
|
|
|
2007-03-26 17:38 +0000 [r59206] Steve Murphy <murf@digium.com>
|
|
|
|
* main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
|
|
pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE,
|
|
and STANDALONE_AEL
|
|
|
|
2007-03-26 15:25 +0000 [r59202] Nadi Sarrar <ns@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
|
|
channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure,
|
|
include/asterisk/autoconfig.h.in, channels/misdn/Makefile,
|
|
channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2
|
|
provides a dsp pipeline for i.e. echo cancellation modules, make
|
|
chan_misdn use it. * add a check for linux/mISDNdsp.h to
|
|
configure.ac and update the autogenerated files: 'configure',
|
|
'autoconfig.h.in' (the 'configure' script was not in sync with
|
|
the latest configure.ac, so the diff is a bit bigger than
|
|
expected).
|
|
|
|
2007-03-26 15:16 +0000 [r59200] Joshua Colp <jcolp@digium.com>
|
|
|
|
* pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the
|
|
aelparse binary! DONT_OPTIMIZE should now work once again.
|
|
|
|
2007-03-24 01:39 +0000 [r59195] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2
|
|
lines Only try to handle a response if it has a response code.
|
|
(ASA-2007-011) ........
|
|
|
|
2007-03-23 16:11 +0000 [r59188-59189] Steve Murphy <murf@digium.com>
|
|
|
|
* /: blocking out the fix in 59187... already incorporated here
|
|
|
|
* /, apps/app_macro.c: Merged revisions 59186 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1
|
|
line Added a few words in the Macro doc strings about the
|
|
behavior of macros with hangups (et al.), as per 9337 ........
|
|
|
|
2007-03-22 23:40 +0000 [r59180-59182] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_sip.c: don't allow string input to overrun the
|
|
buffer to hold it (ASA-2007-010)
|
|
|
|
* channels/chan_misdn.c: remove variables that are no longer used
|
|
(--enable-dev-mode is good, developers should be using it)
|
|
|
|
2007-03-22 14:40 +0000 [r59145] Steve Murphy <murf@digium.com>
|
|
|
|
* utils/Makefile: The stuff in utils was compiling with -O6 even if
|
|
DONT_OPTIMIZE is set in menuconfig. Added the include to fix that
|
|
|
|
2007-03-21 18:08 +0000 [r59081-59089] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/http.c: Add svg mimetype for pari.
|
|
|
|
* res/res_monitor.c, /: Merged revisions 59086 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2
|
|
lines Indicate the filename changed when it is changed. (issue
|
|
#9311 reported by jsmith) ........
|
|
|
|
* channels/chan_sip.c: Until we can do media level parsing for
|
|
sendrecv/etc just use the first value found. This crept up when a
|
|
phone was offered audio+video and returned an inactive video
|
|
stream. chan_sip thought the phone said to put the person on hold
|
|
but that was totally wrong. (issue #9319 reported by benbrown)
|
|
|
|
2007-03-20 21:04 +0000 [r59078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/logger.c: Fix defines for inline stack backtraces (only used
|
|
by developers anyway)
|
|
|
|
2007-03-20 20:42 +0000 [r59076] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/iax2-parser.c: Copy len variable as well, should fix
|
|
remaining IAX2 DTMF issues.
|
|
|
|
2007-03-20 17:48 +0000 [r59069-59070] Steve Murphy <murf@digium.com>
|
|
|
|
* apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should
|
|
return it to its previous, untouched, state.
|
|
|
|
* apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h:
|
|
The fix for the AEL <<security hole>> (bug 9316) is here...
|
|
|
|
2007-03-20 13:16 +0000 [r59064] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
|
|
channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
|
|
channels/misdn/chan_misdn_config.h: Merged revisions
|
|
58849-58850,59062-59063 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) |
|
|
1 line added method standard_dec for dialing out on groups, to
|
|
avoid conflicts, which caused issues with some ISDN providers
|
|
........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13
|
|
Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 |
|
|
crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
|
|
avoid sending a disconnect when we already received one. ........
|
|
r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) |
|
|
1 line modified a loglevel ........
|
|
|
|
2007-03-19 Jason Parker <jparker@digium.com>
|
|
|
|
* Asterisk 1.4.2 released.
|
|
|
|
2007-03-19 22:29 +0000 [r59049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* funcs/func_strings.c: Oops, this should have been a %d all along
|
|
|
|
2007-03-19 15:52 +0000 [r59042] Joshua Colp <jcolp@digium.com>
|
|
|
|
* funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295
|
|
reported by ajohnson)
|
|
|
|
2007-03-19 15:42 +0000 [r59040] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* configs/sip_notify.conf.sample: Fix unescaped semicolon (reported
|
|
via -dev list)
|
|
|
|
2007-03-18 20:37 +0000 [r59037] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return
|
|
code 0 (reported by qwerty1979)
|
|
|
|
2007-03-18 16:36 +0000 [r59035] BJ Weschke <bweschke@btwtech.com>
|
|
|
|
* apps/app_followme.c: Don't return a non-zero return code if the
|
|
profile doesn't exist, to match what the documentation says it
|
|
already does. (#9307 Reported by kkiely)
|
|
|
|
2007-03-16 16:12 +0000 [r58992] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_page.c: Wait for the async thread to exit when hanging
|
|
up all of the paged phones under all circumstances. (issue #9181
|
|
reported by PhilSmith)
|
|
|
|
2007-03-16 01:42 +0000 [r58947-58957] Russell Bryant <russell@digium.com>
|
|
|
|
* configs/sla.conf.sample: fix a couple SLA documentation
|
|
references
|
|
|
|
* doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex
|
|
(removed), doc/freetds.txt (added), doc/odbcstorage.txt (added),
|
|
doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added),
|
|
doc/channelvariables.txt (added), doc/ael.txt (added),
|
|
doc/billing.tex (removed), build_tools/prep_tarball,
|
|
doc/callingpres.txt (added), doc/enum.txt (added),
|
|
doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added),
|
|
doc/cdrdriver.tex (removed), build_tools/make_buildopts_h,
|
|
doc/security.txt (added), doc/imapstorage.txt (added),
|
|
doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed),
|
|
doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac,
|
|
doc/iax.txt (added), doc/ael.tex (removed),
|
|
doc/channelvariables.tex (removed), doc/enum.tex (removed),
|
|
doc/security.tex (removed), doc/math.txt (added), Makefile,
|
|
doc/imapstorage.tex (removed), doc/privacy.tex (removed),
|
|
doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt
|
|
(added), apps/app_voicemail.c, doc/cliprompt.txt (added),
|
|
doc/chaniax.txt (added), doc/app-sms.txt (added),
|
|
doc/ast_appdocs.tex (removed), doc/realtime.tex (removed),
|
|
doc/ices.txt (added), doc/dundi.tex (removed),
|
|
doc/linkedlists.txt (added), doc/queuelog.txt (added),
|
|
doc/extconfig.txt (added), doc/radius.txt (added),
|
|
doc/cliprompt.tex (removed), doc/chaniax.tex (removed),
|
|
doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex
|
|
(removed), doc/ices.tex (removed), doc/asterisk.tex (removed),
|
|
doc/queuelog.tex (removed), doc/configuration.txt (added),
|
|
doc/asterisk-conf.txt (added), doc/sla.pdf (added),
|
|
doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt
|
|
(added), doc/mp3.tex (removed), doc/configuration.tex (removed),
|
|
doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added),
|
|
doc/channels.txt (added), doc/ip-tos.tex (removed),
|
|
doc/extensions.txt (added), doc/queues-with-callback-members.txt
|
|
(added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added),
|
|
doc/misdn.txt (added), doc/manager.txt (added),
|
|
doc/jitterbuffer.tex (removed), doc/extensions.tex (removed),
|
|
doc/billing.txt (added), doc/localchannel.txt (added),
|
|
doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt
|
|
(added), doc/00README.1st (added): Making these documentation
|
|
changes in the 1.4 branch upset various people, so these chanes
|
|
will only be done in the trunk.
|
|
|
|
* build_tools/prep_tarball: Add the --pdf option to the usage of
|
|
rubber in prep_tarball
|
|
|
|
* Makefile, build_tools/menuselect-deps.in, configure,
|
|
include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
|
|
configure script checking for GTK2 and some additional Makefile
|
|
targets to support gmenuselect
|
|
|
|
2007-03-15 23:52 +0000 [r58946] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match
|
|
common syntax and update the resulting appdocs TeX file
|
|
|
|
2007-03-15 23:24 +0000 [r58941] Russell Bryant <russell@digium.com>
|
|
|
|
* doc/asterisk.tex: add a link to the rubber homepage
|
|
|
|
2007-03-15 23:11 +0000 [r58939] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_setcdruserfield.c, main/pbx.c,
|
|
apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c:
|
|
Expand deprecation warnings from simply warning on use to the
|
|
builtin documentation.
|
|
|
|
2007-03-15 22:51 +0000 [r58935-58937] Russell Bryant <russell@digium.com>
|
|
|
|
* doc/asterisk.tex, Makefile: Add Asterisk version information to
|
|
the generated PDF
|
|
|
|
* build_tools/prep_tarball: have prep_tarball attempt to build
|
|
asterisk.pdf
|
|
|
|
2007-03-15 22:32 +0000 [r58933] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* funcs/func_realtime.c: Function works fine, but the documentation
|
|
is backwards.
|
|
|
|
2007-03-15 22:25 +0000 [r58931] Russell Bryant <russell@digium.com>
|
|
|
|
* doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex
|
|
(added), doc/freetds.txt (removed), doc/odbcstorage.txt
|
|
(removed), configure, doc/sla.tex, doc/cygwin.txt (removed),
|
|
doc/model.txt (removed), doc/channelvariables.txt (removed),
|
|
doc/ael.txt (removed), doc/billing.tex (added),
|
|
doc/callingpres.txt (removed), doc/enum.txt (removed),
|
|
doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed),
|
|
doc/cdrdriver.tex (added), build_tools/make_buildopts_h,
|
|
doc/security.txt (removed), doc/imapstorage.txt (removed),
|
|
doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added),
|
|
doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac,
|
|
doc/iax.txt (removed), doc/ael.tex (added),
|
|
doc/channelvariables.tex (added), doc/enum.tex (added),
|
|
doc/security.tex (added), doc/math.txt (removed), Makefile,
|
|
doc/imapstorage.tex (added), doc/privacy.tex (added),
|
|
doc/realtime.txt (removed), doc/dundi.txt (removed),
|
|
doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt
|
|
(removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed),
|
|
doc/ast_appdocs.tex (added), doc/realtime.tex (added),
|
|
doc/ices.txt (removed), doc/dundi.tex (added),
|
|
doc/linkedlists.txt (removed), doc/queuelog.txt (removed),
|
|
doc/extconfig.txt (removed), doc/radius.txt (removed),
|
|
doc/cliprompt.tex (added), doc/chaniax.tex (added),
|
|
doc/hardware.txt (removed), doc/mp3.txt (removed),
|
|
doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex
|
|
(added), doc/queuelog.tex (added), doc/configuration.txt
|
|
(removed), doc/asterisk-conf.txt (removed), doc/sla.pdf
|
|
(removed), doc/ip-tos.txt (removed), doc/hardware.tex (added),
|
|
doc/h323.txt (removed), doc/mp3.tex (added),
|
|
doc/configuration.tex (added), doc/asterisk-conf.tex (added),
|
|
doc/jitterbuffer.txt (removed), doc/channels.txt (removed),
|
|
doc/ip-tos.tex (added), doc/extensions.txt (removed),
|
|
doc/queues-with-callback-members.txt (removed), doc/apps.txt
|
|
(removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt
|
|
(removed), doc/manager.txt (removed), doc/jitterbuffer.tex
|
|
(added), doc/extensions.tex (added), doc/billing.txt (removed),
|
|
doc/localchannel.txt (removed),
|
|
doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt
|
|
(removed), doc/00README.1st (removed): Merge changes from
|
|
svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc
|
|
directory into a single LaTeX formatted document so that we can
|
|
generate a PDF, HTML, or other formats from this information. *
|
|
Add a CLI command to dump the application documentation into
|
|
LaTeX format which will only be include if the configure script
|
|
is run with --enable-dev-mode. * The PDF turned out to be close
|
|
to 1 MB, so it is not included. However, you can simply run "make
|
|
asterisk.pdf" to generate it yourself. We may include it in
|
|
release tarballs or have automatically generated ones on the web
|
|
site, but that has yet to be decided.
|
|
|
|
2007-03-15 18:13 +0000 [r58923] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: Don't assume that the pvt structure will
|
|
still exist after calling schedule_delivery as it may not. (issue
|
|
#9278 reported by fmachado)
|
|
|
|
2007-03-14 19:18 +0000 [r58894-58906] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_sip.c: Some people like to put "limitonpeer"
|
|
instead of "limitonpeers" in their configuration. While we're at
|
|
it, support "limitonpeerz" and "limitonpeerssssss". (inspired by
|
|
issue #9172)
|
|
|
|
* doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the
|
|
examples section
|
|
|
|
* doc/security.txt, /: Merged revisions 58896 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) |
|
|
3 lines Add a note to the security file that the Asterisk CLI and
|
|
log files may contain sensitive information, and that people
|
|
should keep this in mind. ........
|
|
|
|
* configs/sla.conf.sample, apps/app_meetme.c: By default, don't
|
|
attempt to do any CallerID handling at all with SLA because it is
|
|
known to not work properly in some situations. However, add an
|
|
option to enable it for those that would like to use it anyway.
|
|
The short story behind this is that to properly handle CallerID
|
|
with SLA, we need the ability to change the CallerID on an
|
|
existing call, and we are not ready to handle that.
|
|
|
|
2007-03-14 01:47 +0000 [r58880] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* funcs/func_strings.c: Issue 9162 -
|
|
pbx_substitute_variables_helper assumes the buffer is initialized
|
|
to all zeroes. This fixes a case where it wasn't.
|
|
|
|
2007-03-13 23:19 +0000 [r58870-58872] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_meetme.c: Ensure that the blinky lights show that the
|
|
trunk stopped ringing when the trunk hangs up before a station
|
|
has answered it. (issue #9234, reported by francesco_r)
|
|
|
|
* configs/sla.conf.sample: fix the reference to the SLA
|
|
documentation
|
|
|
|
2007-03-13 11:49 +0000 [r58843-58848] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2
|
|
lines Issue #9229 - No port in request URI on register to non
|
|
default SIP ports (neelakantan) ........
|
|
|
|
* channels/chan_sip.c: Don't hangup the call on OK or errors on
|
|
MESSAGE and INFO inside of a dialog (like video update requests).
|
|
|
|
* channels/chan_sip.c: Issue #9251 - Clear From URI from user
|
|
attributes (tgrman)
|
|
|
|
2007-03-12 16:52 +0000 [r58833] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /: Blocked revisions 58832 via svnmerge ........ r58832 | file |
|
|
2007-03-12 12:49:49 -0400 (Mon, 12 Mar 2007) | 2 lines We can't
|
|
use the assembler version of fetchadd_int under Intel Macs.
|
|
(issue #9254 reported by darrell budic) ........
|
|
|
|
2007-03-12 13:08 +0000 [r58825-58826] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
|
|
revisions 57034,57523,57753,58558 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) |
|
|
1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com
|
|
bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02
|
|
19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........
|
|
r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) |
|
|
1 line fixed another place where the out_cause was hardcoded to
|
|
16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09
|
|
Mar 2007) | 1 line we can free channel 31 as well, since we can
|
|
occupy it ........
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
|
|
channels/chan_misdn.c, channels/misdn/ie.c,
|
|
channels/misdn/isdn_msg_parser.c: added UU transceiving and
|
|
corect handling for rdnis
|
|
|
|
2007-03-12 01:21 +0000 [r58779-58783] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Allow RFC2833 compensation to compensate for even
|
|
stupider implementations by queueing up the end frame at the
|
|
start, not the actual end. (issue #8963 reported by AndrewZ)
|
|
|
|
* channels/chan_sip.c, configs/sip.conf.sample: Add
|
|
matchexterniplocally setting which only substitutes your
|
|
externip/externhost setting if it matches the localnet setting. I
|
|
know of at least two people who need opposite settings, so I made
|
|
it an option! (issue #8821 reported by kokoskarokoska)
|
|
|
|
2007-03-10 18:11 +0000 [r58638-58705] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_iax2.c: Fix a few more places in chan_iax2 where
|
|
the ast_frame used for receiving a frame was not properly
|
|
initialized. - Interpolating a frame when the jitterbuffer is in
|
|
use - decrypting a frame when IAX2 encryption is on - frames in
|
|
an IAX2 trunk
|
|
|
|
* apps/app_meetme.c: Make the compiler happy and initialize a
|
|
variable.
|
|
|
|
* doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added):
|
|
Merge some updates to the SLA documentation. I plan to keep
|
|
working on this to explain all of the expected behavior with call
|
|
handling, configuration details for specific phones, and other
|
|
things. However, I got tired of doing it in plain text, so I
|
|
switched to using LaTeX. I have included the PDF version. I
|
|
haven't been able to get a nice looking plain text version out of
|
|
it yet, but I'm not terribly concerned since this is supposed to
|
|
be more of the manual, while the plain text sample configuration
|
|
file is the reference.
|
|
|
|
2007-03-09 21:08 +0000 [r58584-58604] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c: Fix spelling of unavailable in voicemail
|
|
documentation. (issue #9248 reported by tensai)
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2
|
|
lines If we are unable to lookup the host in a c line we have to
|
|
abort, otherwise the previous data is gone and we will
|
|
(potentially) have no data when all is said and done. ........
|
|
|
|
2007-03-08 22:15 +0000 [r58510-58512] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_meetme.c: Hang up the channel that put the call on hold
|
|
in the event processing thread to avoid a race condition. Also,
|
|
if the station originated the call that it is putting on hold,
|
|
don't hang up the trunk if it was the only station on the call
|
|
and it is hanging up due to hold and not a normal hangup.
|
|
|
|
* channels/chan_zap.c: Add a missing break statement so that
|
|
handling the above event does not incorrectly destroy the
|
|
channel. (issue #9242, andrew)
|
|
|
|
2007-03-08 21:33 +0000 [r58479] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* res/res_odbc.c: Fix segfault (Issue 9236)
|
|
|
|
2007-03-08 20:54 +0000 [r58474] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_meetme.c: Refactor hold handling a bit so that it does
|
|
not require keeping the call up when a call is put on hold.
|
|
|
|
2007-03-08 18:01 +0000 [r58389-58436] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Make early SDP seeding even smarter! We have to check
|
|
codecs in the make_compatible function too. (issue #9221 reported
|
|
by marcelbarbulescu)
|
|
|
|
* main/dsp.c, /: Merged revisions 58388 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2
|
|
lines Only print out debug message if the definition that makes
|
|
the variables shows up was actually defined. (issue #9233
|
|
reported by serginuez) ........
|
|
|
|
2007-03-08 13:23 +0000 [r58351-58354] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/http.c: this change was not needed; fclose() handles closing
|
|
the file descriptor already
|
|
|
|
* apps/app_meetme.c: fix a compiler warning, and overwriting 'res'
|
|
value
|
|
|
|
* main/http.c: fix two cases where HTTP session file descriptors
|
|
would not be closed
|
|
|
|
2007-03-08 01:01 +0000 [r58243-58320] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_zap.c, configure, configure.ac: If we receive
|
|
ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256,
|
|
tzafrir) Also, update the configure script to make sure that we
|
|
don't try to build chan_zap if the installed version of zaptel
|
|
does not include ZT_EVENT_REMOVED.
|
|
|
|
* /, channels/chan_iax2.c: (This bug was reported to me by Kinsey
|
|
Moore) Merged revisions 58242 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) |
|
|
7 lines Fix a problem where the Asterisk channel name could be
|
|
that of the wrong IAX2 user for a call. This is because the first
|
|
step of choosing this name is to look for an IAX2 peer that
|
|
happens to have the same IP/port number that this call is coming
|
|
from and assuming that is it. However, this is not always
|
|
correct. So, I have made it change this name after authentication
|
|
happens since at that point, we have an exact match. ........
|
|
|
|
2007-03-07 17:52 +0000 [r58240] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c, channels/chan_sip.c: Ensure we have (or should have)
|
|
at least one matching codec before attempting early bridge SDP
|
|
seeding. (issue #9221 reported by marcelbarbulescu)
|
|
|
|
2007-03-07 00:27 +0000 [r58165-58168] Russell Bryant <russell@digium.com>
|
|
|
|
* /: Blocked revisions 58167 via svnmerge ........ r58167 | russell
|
|
| 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines Fix a
|
|
misplaced block of code in the 1.2 version of the patch to fix
|
|
issue #8977 ........
|
|
|
|
* main/manager.c, /: Merged revisions 58164 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) |
|
|
4 lines If the channels acquired using the manager Redirect
|
|
action are not up, then don't attempt to do anything with them.
|
|
It could lead to weird behavior, including crashes. (issue #8977)
|
|
........
|
|
|
|
2007-03-06 23:10 +0000 [r58121] Steve Murphy <murf@digium.com>
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1
|
|
line Fix for 9220: Eyebeam cannot renew subscriptions for
|
|
presence info. Reason: re-SUBSCRIBE requests don't include Accept
|
|
headers, which the rfc says are optional (to put it tersely), (it
|
|
uses MAY), and luckily, the sip_pvt struct has the format info
|
|
stored, so we simply leave it if the format is set, and the
|
|
accept header null. ........
|
|
|
|
2007-03-06 23:00 +0000 [r58119] Russell Bryant <russell@digium.com>
|
|
|
|
* configs/voicemail.conf.sample: Clarify the documentation of the
|
|
dialout and sendvoicemail options. (issue #9000, caio1982 and
|
|
serge-v)
|
|
|
|
2007-03-06 20:37 +0000 [r58053] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2
|
|
lines Change error message to proper message ........
|
|
|
|
2007-03-06 18:01 +0000 [r58023] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_skinny.c: Return an error of transmit_response is
|
|
called without a session. (issue #9002)
|
|
|
|
2007-03-05 19:19 +0000 [r57870-57914] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: Since chan_iax2 does not support reception
|
|
of DTMF with duration ensure that it is set to 0 on the frame.
|
|
(issue #8521 reported by gdhgdh)
|
|
|
|
* apps/app_meetme.c: Don't create a listen channel and record the
|
|
conference unless the option is turned on. (issue #9204 reported
|
|
by francesco_r)
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2
|
|
lines Make create_dirpath use our standard for return values. -1
|
|
is failure, 0 is success. (issue #9205 reported by ballares)
|
|
........
|
|
|
|
2007-03-05 15:20 +0000 [r57826] Steve Murphy <murf@digium.com>
|
|
|
|
* main/pbx.c, /: Merged revisions 57825 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1
|
|
line Fixed a typo introduced via 9156 (either the gotos or their
|
|
doc strings are wrong) ........
|
|
|
|
2007-03-05 04:19 +0000 [r57768-57798] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/slinfactory.c: Don't allow a NULL pointer to reach
|
|
ast_frdup. (issue #9155 reported by cmaj)
|
|
|
|
* res/res_jabber.c: Don't reference a potentially NULL pointer.
|
|
(issue #9199 reported by klolik)
|
|
|
|
* main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198
|
|
reported by edgreenberg)
|
|
|
|
2007-03-03 15:31 +0000 [r57707] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2,
|
|
pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7:
|
|
Updated the regression tests
|
|
|
|
2007-03-03 06:45 +0000 [r57649] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007)
|
|
| 2 lines Memory leak of a list, if call recording was abandoned
|
|
........
|
|
|
|
2007-03-03 00:59 +0000 [r57620] Dwayne M. Hubbard <dhubbard@digium.com>
|
|
|
|
* main/say.c: submitted patch for Georgian language, issue 9010,
|
|
submitted by Alexander Shaduri
|
|
|
|
2007-03-03 00:02 +0000 [r57591] Russell Bryant <russell@digium.com>
|
|
|
|
* configs/sla.conf.sample: add missing configuration template.
|
|
Thanks to Lacy Moore on asterisk-users for pointing this out\!
|
|
|
|
2007-03-02 Russell Bryant <russell@digium.com>
|
|
|
|
* Asterisk 1.4.1 released.
|
|
|
|
2007-03-02 23:03 +0000 [r57556] Russell Bryant <russell@digium.com>
|
|
|
|
* configure, configure.ac: Update the check that is used to
|
|
determine whether zaptel transcoder support is present. The
|
|
interface has changed.
|
|
|
|
2007-03-02 17:06 +0000 [r57477] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2
|
|
lines If a SIP message comes in and goes to a method handler that
|
|
requires additional values that may not be present then send back
|
|
an error. ........
|
|
|
|
2007-03-02 16:55 +0000 [r57426-57473] Steve Murphy <murf@digium.com>
|
|
|
|
* main/pbx.c, /: Merged revisions 57458 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1
|
|
line further refinement in wording of goto documentation, as per
|
|
9156, goto not proceeding to next instruction ........
|
|
|
|
* pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes
|
|
right, but 9184 points out the problem-- the escape is removed by
|
|
pbx_config, and pbx_ael should also, before sending it down into
|
|
the pbx engine. Also, you have to insert it back in, if you are
|
|
generating extensions.conf code from the AEL.
|
|
|
|
2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant <russell@digium.com>
|
|
|
|
* main/file.c: Return the correct digit that interrupted the
|
|
stream. This fixes exiting the Background application when using
|
|
the m option. (issue #9176, mjagdis)
|
|
|
|
* configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt,
|
|
include/asterisk/channel.h: Merge changes from
|
|
svn/asterisk/team/russell/sla_updates * Originally, I put in the
|
|
documentation that only Zap interfaces would be supported on the
|
|
trunk side. However, after a discussion with Qwell, we came up
|
|
with a way to make IP trunks work as well, using some things
|
|
already in Asterisk. So, here it is, this now officially supports
|
|
IP trunks. * Update the SLA documentation to reflect how to setup
|
|
IP trunks. * Add a section in sla.txt that describes how to set
|
|
up an SLA system with voicemail. * Simplify the way DTMF
|
|
passthrough is handled in MeetMe. * Fix a bug that exposed itself
|
|
when using a Local channel on the trunk side in SLA. The
|
|
station's channel needs to be passed to the dial API when dialing
|
|
the trunk. * Change a WARNING message to DEBUG in channel.h. This
|
|
message is of no use to users.
|
|
|
|
2007-03-01 22:21 +0000 [r57318] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_local.c, /: Merged revisions 57317 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar
|
|
2007) | 2 lines Don't even attempt to optimize things when a
|
|
proxy channel is involved. It will just explode in weird and
|
|
unexplaineable ways. (issue #9175 reported by
|
|
clegall_proformatique) ........
|
|
|
|
2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development <support@transnexus.com>
|
|
|
|
* doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick.
|
|
|
|
2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant <russell@digium.com>
|
|
|
|
* configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla
|
|
docs
|
|
|
|
* configs/sla.conf.sample, apps/app_meetme.c: Merge more changes
|
|
from svn/asterisk/team/russell/sla_updates * Add support for
|
|
private hold. By setting "hold=private" for a trunk, only the
|
|
station that put the call on hold will be able to retrieve it
|
|
from hold. Also, by setting "hold=private" for a station, any
|
|
call that station puts on hold can only be retrieved by that
|
|
station.
|
|
|
|
* apps/app_meetme.c: Minor formatting change
|
|
|
|
* configs/sla.conf.sample, apps/app_meetme.c: Merge changes from
|
|
svn/asterisk/team/russell/sla_updates * Add support for the
|
|
"barge=no" option for trunks. If this option is set, then
|
|
stations will not be able to join in on a call that is on
|
|
progress on this trunk.
|
|
|
|
2007-02-28 19:23 +0000 [r57139] Steve Murphy <murf@digium.com>
|
|
|
|
* main/pbx.c, /: Merged revisions 57118 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1
|
|
line a small documentation update, to reflect reality in the goto
|
|
doc strings, as per 9156, Goto does not proceed to next prio if
|
|
jump fails ........
|
|
|
|
2007-02-28 18:57 +0000 [r57093] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, channels/chan_agent.c: Merged revisions 57092 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb
|
|
2007) | 2 lines Fix a few more issues with the agent logoff CLI
|
|
command. (issue #9123 reported by arbrandes) ........
|
|
|
|
2007-02-28 18:20 +0000 [r57089] Russell Bryant <russell@digium.com>
|
|
|
|
* configs/sla.conf.sample, apps/app_meetme.c: Merge current set of
|
|
changes from svn/asterisk/team/russell/sla_updates * Add support
|
|
for station ring delays. Ring delays can be set globally for a
|
|
station or for specific trunks on the station. * Fix a few bugs
|
|
in existing code. * Restructure and Reorganize code to improve
|
|
readability and maintainability. * Improve formatting of the "sla
|
|
show (trunks|stations)" CLI commands.
|
|
|
|
2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_meetme.c: Picky compiler...
|
|
|
|
* apps/app_speech_utils.c: Better handle timeouts when the
|
|
individual speaks after everything has been played but before the
|
|
timeout ends.
|
|
|
|
2007-02-28 17:15 +0000 [r57049] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/pbx_ael.c: I was surprised that I had not yet downgraded
|
|
missing goto targets and macro call defs to a warning, in case
|
|
they are in extensions.conf; I rectified this problem. Also, A
|
|
goto in a macro to a target in a catch block was not being found;
|
|
I fixed this too; the cause was that I needed to treat catch
|
|
statements like an extension in the find_match code.
|
|
|
|
2007-02-27 17:36 +0000 [r56975] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_voicemail.c: Fix voicemail email attachments. I missed
|
|
the conversion of one of the line endings and there was an extra
|
|
one where it should not have been. (issue #9128)
|
|
|
|
2007-02-26 22:01 +0000 [r56922] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky,
|
|
picky... show deprecation warning in application help, too
|
|
(reported via list)
|
|
|
|
2007-02-26 20:42 +0000 [r56888] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where
|
|
if a device was not specified in alsa.conf, then we just use the
|
|
system default, instead of creating our own default of hw:0,0.
|
|
(issue #9139)
|
|
|
|
2007-02-26 20:07 +0000 [r56856] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2
|
|
lines Obey the clearglobalvars option in extensions reload (or
|
|
dialplan reload depending on your version). (issue #9146 reported
|
|
by ramonpeek) ........
|
|
|
|
2007-02-26 20:04 +0000 [r56847] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_iax2.c: Fix a crash in my last change to
|
|
iax2_indicate(). (issue #9150)
|
|
|
|
2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_record.c: Update app_record documentation to use new CLI
|
|
command, core show file formats. (issue #9151 reported by junky)
|
|
|
|
* main/pbx.c: Use ast_strlen_zero to see if the language and/or
|
|
context argument is not present for Background instead of just
|
|
checking if it is NULL. (issue #9141 reported by mjagdis)
|
|
|
|
2007-02-26 16:51 +0000 [r56785] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_iax2.c: Do more complete locking of the
|
|
chan_iax2_pvt struct in the indicate callback. (Problem brought
|
|
up by Ben Smithurst on the asterisk-dev list)
|
|
|
|
2007-02-26 16:36 +0000 [r56783] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/asterisk.c: Allow both of the show version files and core
|
|
show file versions CLI commands to work. (issue #9135 reported by
|
|
mvanbaak)
|
|
|
|
2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_meetme.c: Move a comment to be in the correct struct.
|
|
|
|
* /: Blocked revisions 56729 via svnmerge ........ r56729 | russell
|
|
| 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines Ensure
|
|
that lock.h is included in utils.c with AST_API_MODULE defined so
|
|
that the implementations will be properly included when the
|
|
AST_INLINE_API functions are not going to be inlined. (issue
|
|
#9124, festr) ........
|
|
|
|
2007-02-25 14:46 +0000 [r56685] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/channel.c, /: Merged revisions 56684 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007)
|
|
| 3 lines Issue 9130 - If prev is the last item on the channel
|
|
list, then evaluating additional conditions (e.g. name prefix)
|
|
will cause a NULL dereference. ........
|
|
|
|
2007-02-24 02:02 +0000 [r56569] Jason Parker <jparker@digium.com>
|
|
|
|
* channels/chan_skinny.c: Make sure to set a speeddials parent on
|
|
creation. Don't crash if hold is pressed when no call is active.
|
|
Don't return in places that we shouldn't..
|
|
|
|
2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* codecs/codec_zap.c: update to match zaptel 1.4 API change that
|
|
was committed a few minutes ago
|
|
|
|
2007-02-23 23:24 +0000 [r56505] Russell Bryant <russell@digium.com>
|
|
|
|
* main/asterisk.c, /: Merged revisions 56504 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) |
|
|
8 lines Fix up a couple more signal handlers to not do bad things
|
|
that could cause various undesirable results. The other day, I
|
|
made Asterisk deadlock by hitting Control-C because of a bad
|
|
signal handler. Now, signal handlers just set a flag and write to
|
|
an alert pipe for the flag to be handled. Then, there is another
|
|
thread that is monitoring for these flags. If being run in
|
|
console mode, it is just the main thread. If Asterisk is in the
|
|
background, a thread is created to do it. ........
|
|
|
|
2007-02-23 21:53 +0000 [r56457] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/sched.c: Change log notice to debug. It is possible for a
|
|
scheduled item to execute and be deleted at close to the same
|
|
time and unavoidable. If this happens this message creeps up.
|
|
|
|
2007-02-23 20:20 +0000 [r56407] Russell Bryant <russell@digium.com>
|
|
|
|
* /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) |
|
|
4 lines Don't destroy mutexes before unregistering all of the
|
|
entry points from the core. Also, fix a potential memory leak
|
|
from not destroying the locks for all of the possible call
|
|
numbers (about 32k of them). ........
|
|
|
|
2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* build_tools/make_version_h: build special version strings for
|
|
AADK/S800i builds
|
|
|
|
2007-02-23 17:58 +0000 [r56341] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_voicemail.c: The IMAP storage code uses the same code to
|
|
build the email that is used when voicemail is sent via email
|
|
using something like sendmail. In the patch from bug 8033 to fix
|
|
various IMAP storage problems, the line endings in the email file
|
|
were changed in the code from "\n" to "\r\n". However, this
|
|
breaks sending regular voicemail to email. So, this change
|
|
conditionally sets line endings to "\r\n" only if IMAP_STORAGE is
|
|
enabled. (issue #9128, patch by jarjarbinks, modified by me to
|
|
not break IMAP storage)
|
|
|
|
2007-02-22 23:25 +0000 [r56280] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /: Blocked revisions 56279 via svnmerge ........ r56279 | file |
|
|
2007-02-22 18:19:25 -0500 (Thu, 22 Feb 2007) | 2 lines Always
|
|
defer Agent logoff if any channels are up until they hang up.
|
|
(issue #9123 reported by arbrandes) ........
|
|
|
|
2007-02-22 23:08 +0000 [r56277] Russell Bryant <russell@digium.com>
|
|
|
|
* configs/sla.conf.sample, main/dial.c, apps/app_meetme.c,
|
|
doc/sla.txt: Merge changes from team/russell/sla_updates. This
|
|
batch of changes to the SLA code does a few different things. * I
|
|
made the SLA code event driven instead of having to act in a lot
|
|
of busy loops while dialing things to wait for state changes.
|
|
This makes the code more efficient and readable at the same time.
|
|
* I have implemented a couple of new features. The first is
|
|
inbound trunk ringing timeouts. This is an option that defines
|
|
how long to let an incoming call on a trunk to ring. * I have
|
|
also implemented ring timeouts for stations. They may be
|
|
specified for the entire station, meaning it is how long to let
|
|
the station ring before giving up. You can also specify a ring
|
|
timeout for a specific trunk on a station. So, you can say that
|
|
you only want a specific station to ring 5 seconds if it is line1
|
|
ringing, but otherwise, there is no timeout.
|
|
|
|
2007-02-22 18:49 +0000 [r56231] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/channel.c, /, channels/chan_sip.c: Merged revisions 56230
|
|
via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2
|
|
lines Only change the original or clone channel if it's the
|
|
channel behind the proxy channel, not if it's just a regular
|
|
bridged channel. ........
|
|
|
|
2007-02-22 14:06 +0000 [r56169] TransNexus OSP Development <support@transnexus.com>
|
|
|
|
* doc/osp.txt: Update OSP documentation for v1.4.
|
|
|
|
2007-02-22 10:33 +0000 [r56125] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Move message from verbose to debug
|
|
|
|
2007-02-22 02:39 +0000 [r56094] Steve Murphy <murf@digium.com>
|
|
|
|
* sounds/Makefile: updated the sound tarball versions in Makefile
|
|
|
|
2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_sip.c: Restructure a little bit of code to reduce
|
|
nesting. There is no functionality change here.
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) |
|
|
3 lines If we receive a frame that is not in any of the
|
|
negotiated formats, then drop it. (potentially issue #8781 and
|
|
SPD-12) ........
|
|
|
|
2007-02-22 00:35 +0000 [r56008] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/cli.c: Print out deprecation notice on usage output of CLI
|
|
commands. (issue #8925 reported by blitzrage)
|
|
|
|
2007-02-22 00:08 +0000 [r56006] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/loader.c: disable unloading of embedded modules... there is
|
|
a fundamental problem with doing so that will not be fixed in
|
|
this version of Asterisk due to its invasiveness
|
|
|
|
2007-02-21 20:35 +0000 [r55957] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2
|
|
lines Change naughty warning message to provide useful
|
|
information. If a write now fails on a channel in meetme it will
|
|
tell you the channel name instead of spitting out the wrong error
|
|
message. ........
|
|
|
|
2007-02-21 20:27 +0000 [r55954] Jason Parker <jparker@digium.com>
|
|
|
|
* channels/chan_gtalk.c: Fix locking issue, and accept
|
|
"transport-accept" as a valid accept message. This should solve
|
|
issues 8970 and 8503.
|
|
|
|
2007-02-21 20:22 +0000 [r55951] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_meetme.c: Simplify the last change to app_meetme, and
|
|
move the call to dispose_conf() up into the block where we know a
|
|
conf exists.
|
|
|
|
2007-02-21 20:16 +0000 [r55914-55949] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_meetme.c: Only dispose of the conference if one was
|
|
created.
|
|
|
|
* apps/app_speech_utils.c: Only start playing the next file if we
|
|
have not been quieted.
|
|
|
|
* channels/chan_sip.c: Add a flag that indicates whether a SIP
|
|
dialog is an outgoing call or not. SIP_OUTGOING originally did it
|
|
but it was repurposed to the direction of the last transaction,
|
|
which can cause update_call_counter to falsely decrease the wrong
|
|
counters. (please don't hurt me oej) (issue #8943 reported by
|
|
mdu113)
|
|
|
|
2007-02-21 14:06 +0000 [r55869] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* /, build_tools/make_version: Merged revisions 55868 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21
|
|
Feb 2007) | 2 lines use new tag version script ........
|
|
|
|
2007-02-21 08:32 +0000 [r55834] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly
|
|
after transfer (decrement inuse early on transferer's call leg)
|
|
|
|
2007-02-21 02:01 +0000 [r55799] Jason Parker <jparker@digium.com>
|
|
|
|
* channels/chan_gtalk.c: Fix segfault when buddy couldn't be found.
|
|
Issue 7764, patch by sailer
|
|
|
|
2007-02-21 01:03 +0000 [r55751-55758] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_meetme.c: Improve the reference counting to fix bugs
|
|
where people report seeing conferences listed that have no
|
|
members. (issue #9073)
|
|
|
|
* /: Blocked revisions 55750 via svnmerge ........ r55750 | russell
|
|
| 2007-02-20 18:19:14 -0600 (Tue, 20 Feb 2007) | 9 lines Fix
|
|
random crashes when using the MeetMe application. This patch
|
|
converts list handling to use the linked list macros and most
|
|
importantly, implements reference counting on the ast_conference
|
|
objects. The reference counting was first backported from 1.4.
|
|
However, that code has some problems that caused the reference
|
|
count to never hit zero. Those problems are fixed in this patch
|
|
and will be resolved in 1.4 and trunk next, with a different
|
|
patch. (issues #7647, #9073, #9106, BE-115). ........
|
|
|
|
2007-02-21 00:11 +0000 [r55670-55741] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c: Better handle dropped IMAP connections.
|
|
(issue #9054 reported by bsmithurst)
|
|
|
|
* channels/chan_sip.c: Return behavior I removed. I did not
|
|
remember that you could just add a localnet entry to make it
|
|
work.
|
|
|
|
* channels/chan_sip.c: Don't test our own address against the
|
|
localnet settings. At least one person has had issues as a result
|
|
of this from #7051 so I'm reversing it. (issue #8821 reported by
|
|
kokoskarokoska)
|
|
|
|
* /, channels/chan_agent.c: Merged revisions 55669 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb
|
|
2007) | 2 lines Defer clearing callback information if channels
|
|
are up until they are hung up. This ensures the hangup process
|
|
goes smoothly and no channels get hung in limbo. (issue #8088
|
|
reported by kebl0155) ........
|
|
|
|
2007-02-20 20:26 +0000 [r55589-55634] Russell Bryant <russell@digium.com>
|
|
|
|
* main/http.c: Add the Asterisk version information to the Server
|
|
header in HTTP responses. (requested by Pari)
|
|
|
|
* include/asterisk/manager.h: Increase the maximum number of
|
|
manager headers to 128, at the request of Pari.
|
|
|
|
* /: Blocked revisions 55588 via svnmerge ........ r55588 | russell
|
|
| 2007-02-20 13:49:50 -0600 (Tue, 20 Feb 2007) | 3 lines Convert
|
|
a tab to spaces so that the documentation is printed out properly
|
|
aligned. ........
|
|
|
|
2007-02-20 16:53 +0000 [r55555] Jason Parker <jparker@digium.com>
|
|
|
|
* channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free
|
|
with strdupa (thanks file) 55555!
|
|
|
|
2007-02-20 16:41 +0000 [r55553] Russell Bryant <russell@digium.com>
|
|
|
|
* configs/sla.conf.sample: Change the formatting of sla.conf.sample
|
|
to make it more readable. (issue #9112, blitzrage)
|
|
|
|
2007-02-19 21:12 +0000 [r55483] Olle Johansson <oej@edvina.net>
|
|
|
|
* res/res_jabber.c: - Not sending arguments to an application is
|
|
not "out of memory" - Making error messages a bit more clear
|
|
|
|
2007-02-19 18:11 +0000 [r55435] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007)
|
|
| 2 lines forcename and forcegreetings options should check to
|
|
see if the recording already exists ........
|
|
|
|
2007-02-19 14:52 +0000 [r55397] Doug Bailey <dbailey@digium.com>
|
|
|
|
* channels/chan_iax2.c: Changed iax2 process thread to detached to
|
|
correct memory leak due to left over thread context on thread
|
|
exit. Modified module unload process to avoid deadlocks on
|
|
pthread cancels
|
|
|
|
2007-02-18 12:35 +0000 [r55250-55278] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, apps/app_record.c: Merged revisions 55277 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2
|
|
lines Documentation update (#9053, jsmith) ........
|
|
|
|
* /: Block patch that was made only for 1.2 (already implemented in
|
|
1.4 and trunk)
|
|
|
|
2007-02-17 17:39 +0000 [r55219] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_queue.c: Add missing membername option to AddQueueMember
|
|
documentation. (issue #9088 reported by seanbright)
|
|
|
|
2007-02-17 17:10 +0000 [r55217] Jason Parker <jparker@digium.com>
|
|
|
|
* channels/chan_skinny.c: Fix an issue where callerid would not be
|
|
displayed on some phones. Issue 8995, initial patch and research
|
|
done by wedhorn
|
|
|
|
2007-02-17 03:55 +0000 [r55086-55154] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_dial.c, /: Merged revisions 55153 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2
|
|
lines Answer the channel before recording privacy information.
|
|
(issue #8926 reported by lmamane) ........
|
|
|
|
* apps/app_queue.c: Make the 'i' option of Queue actually work.
|
|
(issue #8986 reported by utis)
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2
|
|
lines Allow chan_sip to handle attended transfers from a SIP
|
|
phone that is sitting behind chan_agent. Yes folks, all it took
|
|
was one line of code. (issue #8784 reported by pzieba) ........
|
|
|
|
2007-02-17 00:40 +0000 [r55006-55052] Russell Bryant <russell@digium.com>
|
|
|
|
* configure, include/asterisk/autoconfig.h.in, configure.ac: If the
|
|
pg_config application is found, but there is probably executing
|
|
it, then consider postgres unavailable. (issue #8637)
|
|
|
|
* codecs/gsm/Makefile: Filter out yet another architecture that
|
|
does not work with the optimizations in the built-in libgsm.
|
|
(issue 8637, ovi)
|
|
|
|
* /, apps/app_meetme.c, configs/meetme.conf.sample: Merged
|
|
revisions 55005 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) |
|
|
9 lines Revert the change I did in revisions 54955, 54969, and
|
|
54970, in 1.2, 1.4, and trunk. I decided that once a conference
|
|
is created from meetme.conf, it is acceptable behavior that the
|
|
pin can not be changed until the conference goes away. I also
|
|
added a note in meetme.conf to describe this behavior. We still
|
|
have another issue in 1.4 and trunk where some conferences with
|
|
no users don't go away. That is the real bug that needs to be
|
|
addressed here. ........
|
|
|
|
2007-02-16 22:18 +0000 [r55002] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, channels/chan_agent.c: Merged revisions 54999 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb
|
|
2007) | 2 lines Do not send indications through ast_indicate in
|
|
chan_agent but instead go directly to the technology. This way
|
|
when indications are emulated they happen on the Agent channel
|
|
and do not screw up formats on the channels. (issue #8439
|
|
reported by punkgode) ........
|
|
|
|
2007-02-16 21:12 +0000 [r54969] Russell Bryant <russell@digium.com>
|
|
|
|
* /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) |
|
|
5 lines For conferences that are configured in meetme.conf, check
|
|
the configuration file every time someone joins the conference
|
|
instead of only when the conference is first created. This is to
|
|
ensure that changes to the pin numbers in the config file are
|
|
always honored. (issue #9073) ........
|
|
|
|
2007-02-16 18:51 +0000 [r54924] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_dial.c: Need to check macro extension as well as macro
|
|
context for directed pickup.
|
|
|
|
2007-02-16 18:03 +0000 [r54888-54898] Russell Bryant <russell@digium.com>
|
|
|
|
* pbx/pbx_config.c: Fix setting "autofallthrough" to yes by
|
|
default. It was set to enabled in pbx.c. However, if the option
|
|
was not present in extensions.conf, then pbx_config.c would set
|
|
it back to disabled.
|
|
|
|
* res/res_features.c: Clean up a few coding guidelines issues -
|
|
spaces to tabs, use sizeof() to pass the size of a static buffer,
|
|
add spaces ...
|
|
|
|
2007-02-16 17:25 +0000 [r54886] Jason Parker <jparker@digium.com>
|
|
|
|
* main/asterisk.c: Clarify a restart message. It's silly, but the
|
|
reporter had a very valid point. Issue 9079
|
|
|
|
2007-02-16 17:02 +0000 [r54884] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_dial.c: Allow directed pickup to pick up the real
|
|
context instead of the macro context if a Macro is used. (issue
|
|
#8984 reported by jamesb63)
|
|
|
|
2007-02-16 12:06 +0000 [r54772-54787] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Issue #7541 - Handle multipart attachments
|
|
to SIP messages - even if boundary is quoted.
|
|
|
|
* /, res/res_agi.c: Merged revisions 54771 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2
|
|
lines Issue #9069 - If we open with TH we should not close with
|
|
/TD. (seanbright) ........
|
|
|
|
2007-02-16 00:48 +0000 [r54481-54714] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_speech_utils.c: Don't let dtmf leak over into the engine
|
|
and let it skew the results... also give DTMF results priority.
|
|
(issue #9014 reported by surftek)
|
|
|
|
* apps/app_dial.c, /: Merged revisions 54622 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2
|
|
lines Use a separate variable to indicate execution should
|
|
continue instead of the return value. (issue #8842 reported by
|
|
pluto70) ........
|
|
|
|
* apps/app_dial.c: Forward begin DTMF frames as well as end. (issue
|
|
#9068 reported by mhardeman)
|
|
|
|
2007-02-14 18:44 +0000 [r54439] Olle Johansson <oej@edvina.net>
|
|
|
|
* /: Block patch only needed in 1.2
|
|
|
|
2007-02-14 16:56 +0000 [r54375] Matt Frederickson <creslin@digium.com>
|
|
|
|
* channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2
|
|
lines When handling glare on a PRI, move the requested channel
|
|
rather than hang up the old one. Fix for 8957 and 9011. ........
|
|
|
|
2007-02-14 01:09 +0000 [r54290] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/channel.c: Add G722 to ast_best_codec. If anyone disagrees
|
|
with it's placement, feel free to change it. (issue #9045
|
|
reported by gork)
|
|
|
|
2007-02-13 21:31 +0000 [r54204-54235] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_sip.c: Remove a couple of leftover debug messages
|
|
|
|
* include/asterisk/devicestate.h: Fix the documentation on the
|
|
return values from device state provider registration and
|
|
deletion.
|
|
|
|
* channels/chan_sip.c: If we fail to create the SIP socket, then
|
|
return -1 from reload_config() so that load_module() will return
|
|
AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get
|
|
spammed with error messages every time chan_sip tries to send a
|
|
message.
|
|
|
|
2007-02-13 18:41 +0000 [r54180] Olle Johansson <oej@edvina.net>
|
|
|
|
* /: Blocking patch for 1.2 only
|
|
|
|
2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant <russell@digium.com>
|
|
|
|
* main/dial.c, include/asterisk/dial.h: Change
|
|
ast_set_state_callback() to ast_dial_set_state_callback()
|
|
|
|
* main/dial.c, apps/app_meetme.c, apps/app_page.c,
|
|
include/asterisk/dial.h: - Add the ability to register a callback
|
|
to monitor state changes in an asynchronous dial operation. -
|
|
Rename the various references to "status" to "state" in the dial
|
|
API
|
|
|
|
2007-02-12 16:34 +0000 [r54026] Joshua Colp <jcolp@digium.com>
|
|
|
|
* configure, configure.ac: Make the --without-oss argument work.
|
|
(issue #9026 reported by puzzled)
|
|
|
|
2007-02-12 15:38 +0000 [r54002] Russell Bryant <russell@digium.com>
|
|
|
|
* configs/users.conf.sample: Fix a typo where "vmpassword" should
|
|
be "vmsecret"
|
|
|
|
2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/chan_h323.c: Fix VLDTMF reception
|
|
|
|
* apps/app_echo.c: Much simpler than previous one ;-)
|
|
|
|
* main/channel.c: Provide correct DTMF duration
|
|
|
|
* main/cli.c: Bring deprecated 'debug channel <x|all>' command back
|
|
|
|
2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* configure, configure.ac, acinclude.m4: don't display the
|
|
--with-imap message unless --with-imap was specified without a
|
|
path use '-n' instead of '! -z' for tests
|
|
|
|
2007-02-10 01:02 +0000 [r53783-53821] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_meetme.c: Add some output for "show application
|
|
SLAStation/SLATrunk"
|
|
|
|
* channels/chan_sip.c: Change some text to properly state "On
|
|
Hold", which was already done in trunk.
|
|
|
|
* configs/sla.conf.sample, include/asterisk/app.h,
|
|
include/asterisk/utils.h, main/dial.c, apps/app_meetme.c,
|
|
channels/chan_sip.c, doc/sla.txt (added),
|
|
include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge
|
|
team/russell/sla_rewrite This is a completely new implementation
|
|
of the SLA functionality introduced in Asterisk 1.4. It is now
|
|
functional and ready for testing. However, I will be adding some
|
|
additional features over the next week, as well. For information
|
|
on how to set this up, see configs/sla.conf.sample and
|
|
doc/sla.txt. In addition to the changes in app_meetme.c for the
|
|
SLA implementation itself, this merge brings in various other
|
|
changes: chan_sip: - Add the ability to indicate HOLD state in
|
|
NOTIFY messages. - Queue HOLD and UNHOLD control frames even if
|
|
the channel is not bridged to another channel. linkedlists.h: -
|
|
Add support for rwlock based linked lists. dial.c: - Add the
|
|
ability to run ast_dial_start() without a reference channel to
|
|
inherit information from.
|
|
|
|
* apps/app_echo.c: When the Echo() application receives the digit
|
|
'#', echo that back as well. Since we already sent the BEGIN
|
|
frame for that digit, it makes sense to send the END as well.
|
|
|
|
2007-02-09 23:52 +0000 [r53779-53781] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_gtalk.c: another dependency
|
|
|
|
* apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c,
|
|
funcs/func_odbc.c, res/res_adsi.c: add some inter-module
|
|
dependencies
|
|
|
|
* build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk
|
|
scripts to work when both MODULEINFO and MAKEOPTS are present in
|
|
a source file
|
|
|
|
2007-02-09 19:33 +0000 [r53749] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_dial.c: Temporarily change musicclass on channel to one
|
|
specified in Dial so that the 'm' option functions properly.
|
|
(issue #8969 reported by christianbee)
|
|
|
|
2007-02-09 16:42 +0000 [r53715] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* doc/imapstorage.txt, configure, configure.ac: clarify the fact
|
|
that voicemail IMAP storage cannot be built against a distro's
|
|
binary c-client library package (at least not at this time)
|
|
|
|
2007-02-08 23:18 +0000 [r53672] Olle Johansson <oej@edvina.net>
|
|
|
|
* main/acl.c: Don't output debug unless we asked for it
|
|
|
|
2007-02-08 17:54 +0000 [r53601] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_speech_utils.c: Fix timeout issue when utterance is
|
|
longer then timeout itself.
|
|
|
|
2007-02-08 13:47 +0000 [r53530-53532] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/loader.c: Issue 9007 - Mutex not released on early return
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007)
|
|
| 2 lines Issue 9003 - If fullname is empty, quote() passes back
|
|
"\"" ........
|
|
|
|
2007-02-07 23:52 +0000 [r53464-53497] Russell Bryant <russell@digium.com>
|
|
|
|
* main/db1-ast/Makefile: When building libdb1.a, put the additional
|
|
flags needed at the beginning of ASTCFLAGS, instead of at the
|
|
end. This way, we ensure that we find the local headers first
|
|
before accidentally trying to use headers that exist in locations
|
|
specified in the ASTCFLAGS passed from the main Makefile. (issue
|
|
#8637, ovi)
|
|
|
|
* main/Makefile: The clean target actually needs to run "distclean"
|
|
on editline. This is because we need to make sure that its
|
|
configure script gets executed again, because the CFLAGS we want
|
|
to pass to editline may have changed.
|
|
|
|
2007-02-07 17:53 +0000 [r53434] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: We can not reliably do P2P bridging with DTMF passing
|
|
back with compensation if we need to listen for DTMF frames.
|
|
(issue #8962 reported by caio1982)
|
|
|
|
2007-02-07 17:39 +0000 [r53429] Russell Bryant <russell@digium.com>
|
|
|
|
* main/rtp.c: When parsing the NTP timestamp in a sender report
|
|
message, you are supposed to take the low 16 bits of the integer
|
|
part, and the high 16 bits of the fractional part. However, the
|
|
code here was erroneously taking the low 16 bits of the
|
|
fractional part. It then shifted the result 16 bits down, so the
|
|
result was always zero. This fix makes it grab the appropriate
|
|
high 16 bits, instead. (issue #8991, pointed out by
|
|
andre_abrantes)
|
|
|
|
2007-02-07 17:04 +0000 [r53358-53399] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_playback.c: Directly load say.conf in load_module
|
|
instead of calling the reload function. (issue #8946 reported by
|
|
junky)
|
|
|
|
* /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2
|
|
lines Fix a few potential memory leaks with realtime users and
|
|
peers. (issue #8999 reported by bsmithurst) ........
|
|
|
|
2007-02-07 15:33 +0000 [r53355] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /, apps/app_macro.c: Merged revisions 53354 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007)
|
|
| 2 lines Issue 7440 - Macro called from Macro from the h
|
|
extension exits prematurely ........
|
|
|
|
2007-02-07 09:22 +0000 [r53324] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
|
|
revisions 52843 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) |
|
|
1 line fixed some possible segfaults. also fixed an very
|
|
important bug which occurs on high load (when calls are very fast
|
|
generated) ........
|
|
|
|
2007-02-07 05:24 +0000 [r53246-53294] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* res/res_jabber.c: Text fix for jabber reload command (reported by
|
|
bkruse via IRC)
|
|
|
|
* main/manager.c, /: Merged revisions 53245 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007)
|
|
| 2 lines Issue 8987 - Status could return two responses
|
|
(mnicholson) ........
|
|
|
|
2007-02-05 23:43 +0000 [r53222] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Formatting
|
|
|
|
2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_playback.c: Ensure say_cfg is NULL when the module is
|
|
loaded. (issue #8946 reported by junky)
|
|
|
|
* apps/app_playback.c: Unregister Playback CLI commands as well as
|
|
dialplan application. (issue #8946 reported by junky)
|
|
|
|
2007-02-05 00:18 +0000 [r53143] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Add some comments on queue system behaviour
|
|
and how it affects the SIP channel
|
|
|
|
2007-02-03 21:05 +0000 [r53138] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Make SIPDtmfMode application work with
|
|
recent capability changes, and also fix an RTP stack issue when
|
|
the auto option was used. (issue #8972 reported by mdu113)
|
|
|
|
2007-02-03 20:44 +0000 [r53135-53136] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_dial.c, /: Merged revisions 53133 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) |
|
|
4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when
|
|
the dial application exits early because of invalid arguments
|
|
instead of just leaving it empty. (issue #8975) ........
|
|
|
|
* /: Blocked revisions 53134 via svnmerge ........ r53134 | russell
|
|
| 2007-02-03 14:39:45 -0600 (Sat, 03 Feb 2007) | 2 lines Revert
|
|
some changes that accidentally got committed as a part of another
|
|
fix. ........
|
|
|
|
2007-02-03 10:02 +0000 [r53131] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string
|
|
because due to compatibilities with CS1000 reported at
|
|
www.voip-info.org
|
|
|
|
2007-02-02 21:26 +0000 [r53129] BJ Weschke <bweschke@btwtech.com>
|
|
|
|
* UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a
|
|
warning to the console that things might possibly be
|
|
misconfigured when queue member's states are still 'Not in Use'
|
|
when we're about to bridge them with a caller from queue. Also,
|
|
put some documentation quoted from oej's queues.txt efforts
|
|
started in /trunk today. This commit puts #7433 into feedback
|
|
state for 1.4, and pending no further negative feedback, it will
|
|
finally be closed.
|
|
|
|
2007-02-02 17:15 +0000 [r53114-53120] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Correct a copy/pasted error message line for RTCP.
|
|
|
|
* main/config.c, /: Merged revisions 53117 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2
|
|
lines Pass the glob expanded filename to process_text_line so
|
|
that error messages contain the actual filename, not the original
|
|
include one. (issue #8959 reported by tzafrir) ........
|
|
|
|
* Makefile: Add systemname to asterisk.conf generation per recent
|
|
discussions about it. (issue #8968 reported by blitzrage)
|
|
|
|
2007-02-02 00:24 +0000 [r53109] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c, configs/sip.conf.sample: Disable the direct
|
|
p2p RTP call setup in SIP. You can enable it in sip.conf, but it
|
|
is now considered experimental until we solve the
|
|
AST_CONTROL_ANSWER with payload and videocaps stuff.
|
|
|
|
2007-02-01 23:16 +0000 [r53108] Jason Parker <jparker@digium.com>
|
|
|
|
* /: Blocked revisions 53107 via svnmerge ........ r53107 | qwell |
|
|
2007-02-01 17:14:09 -0600 (Thu, 01 Feb 2007) | 2 lines Fix a
|
|
small typo. Synopsis lines shouldn't have a newline ........
|
|
|
|
2007-02-01 22:24 +0000 [r53097-53104] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2
|
|
lines Copy noncodeccapability over to the joint variable so that
|
|
telephone-event will get transmitted in the sent INVITE. ........
|
|
|
|
* main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile
|
|
here as well, but it apparently required both dev mode and no
|
|
optimizations to creep up.
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2
|
|
lines Don't negotiate RFC2833 when not configured to do so.
|
|
(issue #8799 reported by mdu113) ........
|
|
|
|
2007-02-01 21:24 +0000 [r53093] Russell Bryant <russell@digium.com>
|
|
|
|
* funcs/func_strings.c: Fix the FIELDQTY function to not crash.
|
|
(reported by blitzrage and Corydon on IRC)
|
|
|
|
2007-02-01 21:15 +0000 [r53091] Olle Johansson <oej@edvina.net>
|
|
|
|
* /: Going backwards, blame file.
|
|
|
|
2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb
|
|
2007) | 2 lines Return previous behavior of having MOH pick up
|
|
where it was left off. (issue #8672 reported by sinistermidget)
|
|
........
|
|
|
|
* funcs/func_strings.c: Make func_strings build under dev mode.
|
|
Didn't I do this today already in the berkeley DB?
|
|
|
|
2007-02-01 21:05 +0000 [r53079-53085] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: - Clean INC_COUNT flag when we decrement
|
|
call counter - If it's still set at time of dialog destruction,
|
|
make sure we decrement the device call counter properly before we
|
|
destroy the dialog
|
|
|
|
* apps/app_queue.c: Change debug level for state change message
|
|
that is not really informative when debugging app_queue
|
|
|
|
* channels/chan_sip.c: Cleaning up the devicestate callback
|
|
function
|
|
|
|
2007-02-01 20:13 +0000 [r53075-53077] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* funcs/func_strings.c: Oops.
|
|
|
|
* /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007)
|
|
| 2 lines Bug 8965 ........
|
|
|
|
2007-02-01 19:33 +0000 [r53072] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/asterisk.c: Add missing 'F' letter to getopt so it magically
|
|
becomes a valid option. (issue #8960 reported by tzafrir)
|
|
|
|
2007-02-01 19:21 +0000 [r53070] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via
|
|
svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007)
|
|
| 2 lines No wonder FIELDQTY doesn't work with functions... the
|
|
documentation in pbx.c was wrong ........
|
|
|
|
2007-02-01 17:37 +0000 [r53064] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Fix silly logic. We really want to write
|
|
UDPTL frames out when the call is up.
|
|
|
|
2007-02-01 16:35 +0000 [r53062] Olle Johansson <oej@edvina.net>
|
|
|
|
* configs/sip.conf.sample: Add explanation of port= in combination
|
|
with defaultip= (thanks jsmith)
|
|
|
|
2007-02-01 13:17 +0000 [r53060] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/chan_misdn.c: we update the name on any first reply of
|
|
our setup
|
|
|
|
2007-02-01 11:07 +0000 [r53057] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/chan_h323.c: chan_h323 is very stable, so let it built
|
|
by default
|
|
|
|
2007-02-01 00:24 +0000 [r53050-53052] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: When going on hold have the side that was put on hold
|
|
reinvite back to Asterisk. When going off hold have the side that
|
|
was taken off hold reinvited back to the other party.
|
|
|
|
* main/rtp.c: Add more frame types to forward in the RTP bridge
|
|
loops.
|
|
|
|
2007-01-31 21:32 +0000 [r52859-53046] Russell Bryant <russell@digium.com>
|
|
|
|
* main/cdr.c, main/manager.c, pbx/pbx_spool.c,
|
|
channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
|
|
pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c,
|
|
main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c,
|
|
channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c:
|
|
Merged revisions 53045 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) |
|
|
3 lines Fix a bunch of places where pthread_attr_init() was
|
|
called, but pthread_attr_destroy() was not. ........
|
|
|
|
* apps/app_userevent.c: Remove an extra \r\n from manager user
|
|
events. (issue #8955, mnicholson)
|
|
|
|
* main/rtp.c, /: Merged revisions 53039 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) |
|
|
3 lines Use the proper format string to print unsigned values in
|
|
the rtp debug output. (issue #8954, wmis) ........
|
|
|
|
* apps/app_queue.c: Only changed the paused status in an existing
|
|
queue member if the paused column exists.
|
|
|
|
* apps/app_queue.c: Instead of always creating a realtime queue
|
|
member as unpaused, read the "paused" column and use that value
|
|
for the paused status of the member. (issue #8949, jmls)
|
|
|
|
* contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10.
|
|
(issue #8363, johnlange)
|
|
|
|
* doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue
|
|
#8942, lters)
|
|
|
|
* configure, include/asterisk/autoconfig.h.in, configure.ac,
|
|
codecs/codec_gsm.c: When we are checking for a system installed
|
|
version of libgsm, we need to check for gsm.h as well.
|
|
Furthermore, when checking for this header, it may be located in
|
|
a gsm/ sub directory, so check for that, as well. (issue #8773)
|
|
|
|
* /: Blocked revisions 52954 via svnmerge ........ r52954 | russell
|
|
| 2007-01-30 13:41:52 -0600 (Tue, 30 Jan 2007) | 4 lines Don't
|
|
print a message indicating that we don't know what to do with a
|
|
proceeding control frame in ast_request_and_dial(). We just need
|
|
to ignore it. (reported by JerJer on #asterisk-dev) ........
|
|
|
|
* channels/chan_sip.c: Only set the DTMF flag on the rtp structure
|
|
if the DTMF mode is actually RFC2833, not just that it is not
|
|
INFO. This makes it get set for inband DTMF as well, which is not
|
|
valid. (issue #8936)
|
|
|
|
* main/asterisk.c, /: Merged revisions 52903 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) |
|
|
9 lines The SIGHUP handler was implemented to allow admins to
|
|
send SIGHUP to a running Asterisk process to reload the
|
|
configuration. However, doing the actual reload in the signal
|
|
handler itself is a very bad thing to do, because the reload
|
|
process includes calling non-reentrant functions such as
|
|
malloc/calloc/etc. If Asterisk is running in the background, then
|
|
the reload will happen immediately. However, if running in
|
|
console mode, the reload doesn't work until something is typed at
|
|
the console. That sort of defeats the purpose, but I don't see an
|
|
easy way to get around it at this point. ........
|
|
|
|
* /: Blocked revisions 52857 via svnmerge ........ r52857 | russell
|
|
| 2007-01-30 09:35:23 -0600 (Tue, 30 Jan 2007) | 5 lines Comment
|
|
out the parts in the Makefile that make codec_zap get built. It
|
|
will not yet build against zaptel 1.2, so I am disabling it to
|
|
prevent further bug reports until it gets merged. (issue #8940)
|
|
........
|
|
|
|
2007-01-30 15:29 +0000 [r52856] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: Drop the deprecated show commands since the
|
|
original ones were changed back. (issue #8937 reported by
|
|
PCadach)
|
|
|
|
2007-01-30 08:46 +0000 [r52807-52809] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/chan_h323.c: Revert reprecation of h.323 gk cycle
|
|
command from pre-1.4 version instead of duplicated h323 cycle gk
|
|
|
|
* res/res_odbc.c: Don't play with free()'d pointers
|
|
|
|
* configure, acinclude.m4: Handle non-standard OpenH323/PWLib
|
|
library names
|
|
|
|
2007-01-30 00:15 +0000 [r52763] Russell Bryant <russell@digium.com>
|
|
|
|
* /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) |
|
|
5 lines Fix the extraction of the timestamp from video frames. It
|
|
was using the mapping for a mini-frame instead of a video-frame,
|
|
which caused it to get invalid data. (issue #8795, mihai)
|
|
........
|
|
|
|
2007-01-29 23:43 +0000 [r52717] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan
|
|
2007) | 2 lines Now that filename is part of the structure and
|
|
since it comes before postprocess... we have to add it to our
|
|
postprocess line. (reported on asterisk-dev by Boris Bakchiev)
|
|
........
|
|
|
|
2007-01-29 22:58 +0000 [r52688-52695] Russell Bryant <russell@digium.com>
|
|
|
|
* main/Makefile: Add a missing quotation mark. This was pointed out
|
|
by jcmoore on #asterisk-dev.
|
|
|
|
* main/manager.c: Remove a recursive lock of the manager session.
|
|
This was pointed out by zandbelt in issue #8711.
|
|
|
|
2007-01-29 22:12 +0000 [r52679] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* pbx/pbx_config.c: Argument number correction
|
|
|
|
2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant <russell@digium.com>
|
|
|
|
* main/Makefile: ASTLDFLAGS needs to be passed to the editline
|
|
configure script as LDFLAGS. (issue #8928, zandbelt)
|
|
|
|
* main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF
|
|
mode translation. P2P bridging can only be used when the DTMF
|
|
modes don't match if the core is monitoring DTMF in both
|
|
directions. Then, the core will handle the translation.
|
|
Otherwise, this bridging method can not be used. (issue #8936)
|
|
|
|
* main/manager.c: The session lock can not be held while calling
|
|
action callbacks. If so, then when the WaitEvent callback gets
|
|
called, then no event can happen because the session can't be
|
|
locked by another thread. Also, the session needs to be locked in
|
|
the HTTP callback when it reads out the output string. This fixes
|
|
the deadlock reported in both 8711 and 8934. Regarding issue
|
|
8711, there still may be an issue. If there is a second action
|
|
requested before the processing of the first action is finished,
|
|
there could still be some corruption of the output string buffer
|
|
used to build the result. (issue #8711, #8934)
|
|
|
|
2007-01-29 18:59 +0000 [r52572] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c: Use ast_calloc instead of malloc.
|
|
|
|
2007-01-29 17:57 +0000 [r52535] Steve Murphy <murf@digium.com>
|
|
|
|
* apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR
|
|
backport to 1.4). It was committed to trunk via 7663. But it
|
|
wasn't so much an enhancement as a fix for the bad language
|
|
output for portuguese in Brazil, so, after a lot of prodding from
|
|
patient Brazilians, here is the same fix for 1.4
|
|
|
|
2007-01-29 17:33 +0000 [r52523] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c: Set quota information to 0 when creating a
|
|
vm_state. (issue #8924 reported by neutrino88)
|
|
|
|
2007-01-29 16:54 +0000 [r52506] Russell Bryant <russell@digium.com>
|
|
|
|
* main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in
|
|
the last commit to the adaptive jitterbuffer code. - Specifically
|
|
indicate to the compiler that the "dropem" variable only needs
|
|
one but. - Change formatting to conform to coding guidelines.
|
|
|
|
2007-01-29 04:18 +0000 [r52494] Jim Dixon <telesistant@hotmail.com>
|
|
|
|
* main/jitterbuf.c, include/jitterbuf.h: Fixed problem with
|
|
jitterbuf, whereas it would not complain about, and would allow
|
|
itself to be overfilled (per the max_jitterbuf parameter). Now it
|
|
rejects any data over and above that size, and complains about
|
|
it.
|
|
|
|
2007-01-28 05:15 +0000 [r52462] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* configure, configure.ac: Suggested change to fix normal usage of
|
|
--with-tds=/usr/local (Sean Bright, via asterisk-dev mailing
|
|
list)
|
|
|
|
2007-01-27 02:13 +0000 [r52335-52416] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, apps/app_queue.c: Merged revisions 52415 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2
|
|
lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log
|
|
follow documentation. (issue #7677 reported by amilcar) ........
|
|
|
|
* main/manager.c: Have the manager interface send back an "Already
|
|
logged in" message instead of "Invalid/Unknown Command" when the
|
|
client authenticates for a second time. (issue #8509 reported by
|
|
pari)
|
|
|
|
* /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2
|
|
lines Make the last context entry read in the dominant one.
|
|
(issue #8918 reported by pj) ........
|
|
|
|
* main/file.c: Fix core show file formats CLI command.
|
|
|
|
2007-01-25 19:18 +0000 [r52163-52265] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2
|
|
lines Allow dequeueing of frames with negative timestamp by
|
|
moving jitterbuffer frames check to jb_next. (issue #8546
|
|
reported by harmen) ........
|
|
|
|
* channels/chan_sip.c: Drop out variables I accidentally put in.
|
|
|
|
* channels/chan_sip.c: Decrement onHold count if we are hung up on
|
|
and still on hold. (issue #8909 reported by alexh42)
|
|
|
|
* apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan
|
|
2007) | 2 lines Add another note about audio files being played
|
|
back to each bridged party. (issue #8718 reported by ppyy)
|
|
........
|
|
|
|
2007-01-25 01:37 +0000 [r52107-52160] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_voicemail.c, configs/users.conf.sample: By suggestion
|
|
from kpfleming last week, change "vmpassword" to "vmsecret".
|
|
|
|
* configure, configure.ac: Remove libnsl as a required lib for
|
|
libiksemel to work. This change was already made in the trunk.
|
|
(issue #8762)
|
|
|
|
* /: Blocked revisions 52137 via svnmerge ........ r52137 | russell
|
|
| 2007-01-24 18:39:50 -0600 (Wed, 24 Jan 2007) | 3 lines Fix a
|
|
seg fault when running this application with no arguments from
|
|
AGI. (issue #8905, junky) ........
|
|
|
|
* include/asterisk/dial.h: Fix the formatting of doxygen comments
|
|
to properly indicate that the comment documents the previous
|
|
entity, as opposed to the next one.
|
|
|
|
2007-01-24 18:26 +0000 [r52052] Steve Murphy <murf@digium.com>
|
|
|
|
* utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via
|
|
svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1
|
|
line updated check_expr via 8322 (refactoring of expression
|
|
checking impl); elfring contributed a nice code reorg, I
|
|
contributed some time to get it working again, better messages
|
|
........
|
|
|
|
2007-01-24 18:20 +0000 [r52016-52049] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/dial.c (added), apps/app_page.c, main/Makefile,
|
|
include/asterisk/dial.h (added): Merge in dialing API and the
|
|
app_page that uses it. (issue #BE-118)
|
|
|
|
* channels/chan_sip.c: Fix changing channel formats when joint
|
|
capability changes and there are no audio formats... I didn't
|
|
break it originally! (issue #8535 reported by ivoc)
|
|
|
|
2007-01-24 17:14 +0000 [r52000] Russell Bryant <russell@digium.com>
|
|
|
|
* configure: rebuild configure script to reflect last chan_h323
|
|
related changes.
|
|
|
|
2007-01-24 12:57 +0000 [r51979-51989] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/chan_misdn.c: added fix from #8899
|
|
|
|
* channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24
|
|
Jan 2007) | 1 line fixed the busy problem (dialstatus was not
|
|
busy when we called a busy extension) ........
|
|
|
|
2007-01-24 09:30 +0000 [r51931] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Show capabilities *and* preference in
|
|
general settings in "sip show settings" (reported by Clona/Telio
|
|
- Thanks!)
|
|
|
|
2007-01-24 08:04 +0000 [r51895] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* acinclude.m4: Allow x64 builds of H.323 (please, rebuild
|
|
configure)
|
|
|
|
2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant <russell@digium.com>
|
|
|
|
* main/channel.c, /: Merged revisions 51843 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) |
|
|
6 lines Fix an issue related to synchronization of recordings
|
|
when using Monitor(). The bug is a miscalculation of the amount
|
|
to seek the stream for writing to disk when the number of samples
|
|
coming in and out of a channel do not match up. (issue #8298,
|
|
#8887, report and patch by guillecabeza, patch files created and
|
|
testing done by whoiswes) ........
|
|
|
|
* apps/app_while.c, /: Merged revisions 51828 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) |
|
|
4 lines Don't set a new value for the END_ variable on the
|
|
channel before using the old value. If you do, it will lead to
|
|
accessing a memory address that has been free()'d. (issue #8895,
|
|
arkadia) ........
|
|
|
|
2007-01-23 22:46 +0000 [r51788] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c,
|
|
channels/chan_sip.c, channels/chan_skinny.c,
|
|
channels/chan_features.c, channels/chan_alsa.c,
|
|
channels/chan_gtalk.c, channels/chan_iax2.c: Update channel
|
|
drivers to use module referencing so that unloading them while in
|
|
use will not result in crashes. (issue #8897 reported by junky)
|
|
|
|
2007-01-23 22:04 +0000 [r51750-51781] Russell Bryant <russell@digium.com>
|
|
|
|
* main/manager.c: Fix some bugs in process_message(). The manager
|
|
session lock needs to be held when sending some sort of response,
|
|
or calling one of the manager action callbacks. This resolves an
|
|
issue where people using the GUI would get random crashes when
|
|
they start clicking around a lot. (issue #8711, reported and
|
|
debugged by zandbelt)
|
|
|
|
* main/http.c: Fix setting the default port of 8088 on 64-bit or
|
|
big-endian machines.
|
|
|
|
* main/manager.c: When traversing the list of manager actions, the
|
|
iterator needs to be initialized to the list head *after* locking
|
|
the list. Also, lock the actions list in one place it is being
|
|
accessed where it was not being done.
|
|
|
|
2007-01-23 20:32 +0000 [r51683-51716] Steve Murphy <murf@digium.com>
|
|
|
|
* res/res_features.c: this mod from 8593 (dstchannel in cdr is
|
|
empty when transfer call).
|
|
|
|
* main/callerid.c: via 8748 (callerid.c loses name when returning
|
|
PRIVATE_NUMBER flag), the user suggested this mod, saying it
|
|
would allow 'WITHHELD' to appear in the name field, which would
|
|
be useful
|
|
|
|
2007-01-23 10:28 +0000 [r51648-51649] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
|
|
channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506
|
|
via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) |
|
|
6 lines * more additions to make the RESTART message work * added
|
|
fix for misdn_call to allow SETUPs with empty extensions,
|
|
replaced the strtok_r functions with strsep for that (inspired by
|
|
Sandro Cappellazzo, thanks) ........ r50506 | crichter |
|
|
2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get
|
|
L2 UP, the L1 is UP definitely too, so we set the L1 state up as
|
|
well. ........
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
|
|
channels/chan_misdn.c: manually merged r49922 and r50335, because
|
|
of conflicts. this commint includes addition of the ISDN RESTART
|
|
Message
|
|
|
|
2007-01-23 06:51 +0000 [r51615] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/chan_h323.c, channels/Makefile: Do not abort Asterisk
|
|
startup if h323 configuration file not found (reported by
|
|
mithraen)
|
|
|
|
2007-01-23 03:00 +0000 [r51513-51558] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Only change audio formats on the channel if
|
|
we have an audio format to change to. (issue #8535 reported by
|
|
ivoc)
|
|
|
|
* /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan
|
|
2007) | 2 lines Yield before reading from zaptel timing source
|
|
under Solaris so that other threads get a chance to do things.
|
|
(issue #7875 reported by bob) ........
|
|
|
|
2007-01-22 19:41 +0000 [r51411] Russell Bryant <russell@digium.com>
|
|
|
|
* /: Blocked revisions 51410 via svnmerge ........ r51410 | russell
|
|
| 2007-01-22 13:39:30 -0600 (Mon, 22 Jan 2007) | 3 lines Merge
|
|
codec_zap support for the transcoder card. This is a standalone
|
|
codec module so it will not affect anything else. ........
|
|
|
|
2007-01-22 19:28 +0000 [r51409] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/pbx_ael.c: This fixes 8836, according to dnatural
|
|
|
|
2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan
|
|
2007) | 2 lines Move filestream creation to Mixmonitor loop. This
|
|
will prevent a blank file from being created if no frames ever
|
|
pass through to be recorded. (issue #7589 reported by
|
|
steve_mcneil) ........
|
|
|
|
* /: Blocked revisions 51359 via svnmerge ........ r51359 | file |
|
|
2007-01-22 11:23:03 -0500 (Mon, 22 Jan 2007) | 2 lines Explicitly
|
|
declare what codecs are supported by default globally since using
|
|
a bitmask for all may include ones we don't need. (issue #8357
|
|
reported by gknispel_proformatique) ........
|
|
|
|
2007-01-20 06:53 +0000 [r51348-51350] Jason Parker <jparker@digium.com>
|
|
|
|
* configs/say.conf.sample: Fix Italian numeral support in say.conf
|
|
for "_[2-9]00" case. "2131" would've translated to something
|
|
along the lines of (pardon my..Italian {or lack thereof})
|
|
"duecentocentotrentuno", which makes no sense at all.
|
|
|
|
* configs/say.conf.sample: Fix German language support in say.conf
|
|
Properly support 21, 31, 41, 51, 61, 71, 81, and 91.
|
|
einundzwanzig has the same format as zweiundzwanzig (as do all
|
|
other "_ZX" spoken numerals) Fix support for numbers in the
|
|
10,000,000 to 99,999,999 range. Add support for numbers in the
|
|
100,000,000 to 999,999,999 range.
|
|
|
|
2007-01-20 00:13 +0000 [r51302-51343] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_meetme.c: Remove an unused instance of an unnamed enum.
|
|
|
|
* apps/app_meetme.c: Remove another duplicated definition
|
|
|
|
* apps/app_meetme.c: Remove a variable that was declared twice.
|
|
|
|
* codecs/gsm/Makefile: Add a couple more processors that need
|
|
optimizations excluded. (issue #8637)
|
|
|
|
* channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk.
|
|
AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same
|
|
thing. So, a digit would have been interpreted incorrectly here.
|
|
Since the channel driver will always have the begin and end
|
|
callbacks called for a digit, only support the button-down and
|
|
button-up messages.
|
|
|
|
* .cleancount: Bump the cleancount since my last commit changed the
|
|
channel structure.
|
|
|
|
* channels/chan_oss.c, main/rtp.c, main/channel.c,
|
|
channels/chan_phone.c, channels/chan_misdn.c,
|
|
channels/chan_skinny.c, channels/chan_features.c,
|
|
channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c,
|
|
channels/chan_zap.c, channels/chan_local.c, main/frame.c,
|
|
channels/chan_sip.c, channels/chan_agent.c,
|
|
include/asterisk/channel.h, channels/chan_gtalk.c,
|
|
channels/chan_iax2.c: Merge the changes from the
|
|
/team/group/vldtmf_fixup branch. The main bug being addressed
|
|
here is a problem introduced when two SIP channels using SIP INFO
|
|
dtmf have their media directly bridged. So, when a DTMF END frame
|
|
comes into Asterisk from an incoming INFO message, Asterisk would
|
|
try to emulate a digit of some length by first sending a DTMF
|
|
BEGIN frame and sending a DTMF END later timed off of incoming
|
|
audio. However, since there was no audio coming in, the DTMF_END
|
|
was never generated. This caused DTMF based features to no longer
|
|
work. To fix this, the core now knows when a channel doesn't care
|
|
about DTMF BEGIN frames (such as a SIP channel sending INFO
|
|
dtmf). If this is the case, then Asterisk will not emulate a
|
|
digit of some length, and will instead just pass through the
|
|
single DTMF END event. Channel drivers also now get passed the
|
|
length of the digit to their digit_end callback. This improves
|
|
SIP INFO support even further by enabling us to put the real
|
|
digit duration in the INFO message instead of a hard coded 250ms.
|
|
Also, for an incoming INFO message, the duration is read from the
|
|
frame and passed into the core instead of just getting ignored.
|
|
(issue #8597, maybe others...)
|
|
|
|
* main/asterisk.c: Merged revisions 51300 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) |
|
|
4 lines Fix a memory leak on command line tab completion. The
|
|
container for the matches was freed, but the individual matches
|
|
themselves were not. (issue #8851, arkadia) ........
|
|
|
|
2007-01-19 00:17 +0000 [r51272-51274] Dwayne M. Hubbard <dhubbard@digium.com>
|
|
|
|
* channels/chan_zap.c: chan_zap compiles without libpri after
|
|
committing 7877 patch
|
|
|
|
* channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007)
|
|
| 3 lines issue 7877: chan_zap module reload does not use
|
|
default/initialized values on subsequent loads. Reset
|
|
configuration variables to default values prior to parsing
|
|
configuration file. ........
|
|
|
|
2007-01-18 23:36 +0000 [r51270] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* /: block this patch since it is already here
|
|
|
|
2007-01-18 22:50 +0000 [r51265] Jason Parker <jparker@digium.com>
|
|
|
|
* apps/app_voicemail.c, main/channel.c, main/pbx.c,
|
|
funcs/func_strings.c, main/app.c: Add some more checks for
|
|
option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832,
|
|
patch(es) by tgrman
|
|
|
|
2007-01-18 21:54 +0000 [r51262] Russell Bryant <russell@digium.com>
|
|
|
|
* Makefile, configure, main/Makefile, acinclude.m4, makeopts.in:
|
|
Ensure that the locations given to the Asterisk configure script
|
|
for ncurses, curses, termcap, or tinfo are further passed along
|
|
to the editline configure script. This fixes some
|
|
cross-compilation environments. (issue #8637, reported by ovi,
|
|
patch by me)
|
|
|
|
2007-01-18 21:14 +0000 [r51256] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18
|
|
Jan 2007) | 2 lines If a timezone is not specified, assume
|
|
localtime (instead of gmtime) (Issue #7748) ........
|
|
|
|
2007-01-18 19:17 +0000 [r51251] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_speech_utils.c: Only start timeout once we reach the end
|
|
of the files to play back.
|
|
|
|
2007-01-18 18:42 +0000 [r51245] Jason Parker <jparker@digium.com>
|
|
|
|
* main/cli.c: Fix an issue with file name completion in "module
|
|
load" and "load". Issue 8846
|
|
|
|
2007-01-18 18:36 +0000 [r51243] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Copy MOH settings when calling a peer so
|
|
that if they put someone on hold or get put on hold themselves
|
|
they get the right music class. (issue #8840 reported by mdu113)
|
|
|
|
2007-01-18 18:28 +0000 [r51241] Jason Parker <jparker@digium.com>
|
|
|
|
* main/channel.c: Fix an issue with deprecated commands
|
|
|
|
2007-01-18 17:49 +0000 [r51236] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18
|
|
Jan 2007) | 2 lines Document all the fields, including the
|
|
indication that "uniqueid" should not be renamed. ........
|
|
|
|
2007-01-18 17:18 +0000 [r51233] Russell Bryant <russell@digium.com>
|
|
|
|
* main/manager.c: Make the "hasmanager" option in users.conf
|
|
actually have an effect. (issue #8740, LnxPrgr3)
|
|
|
|
2007-01-18 00:48 +0000 [r51211-51213] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c: Build the IMAP remote directory string
|
|
better and properly. Fix an issue with encoding the GSM voicemail
|
|
when attaching to the voicemail. (issue #8808 reported by
|
|
akohlsmith)
|
|
|
|
* main/rtp.c: Pass data as well for hold/unhold/vidupdate frames.
|
|
(issue #8840 reported by mdu113)
|
|
|
|
2007-01-17 23:31 +0000 [r51198-51205] Russell Bryant <russell@digium.com>
|
|
|
|
* funcs/func_odbc.c: Fix some instances where when loading
|
|
func_odbc, a double-free could occur. Also, remove an unneeded
|
|
error message. If the failure condition is actually a memory
|
|
allocation failure, a log message will already be generated
|
|
automatically.
|
|
|
|
* channels/chan_zap.c: Instead of dividing the offset by 2
|
|
directly, make it more clear that the offset is being scaled by
|
|
the size of the elements in the buffer. (Inspired by a discussing
|
|
on the asterisk-dev list about this code)
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) |
|
|
3 lines Move the check for a failure of ast_channel_alloc() to
|
|
before locking the pvt structure again. Otherwise, on a failure,
|
|
this will cause a deadlock. ........
|
|
|
|
2007-01-17 20:56 +0000 [r51195] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /, main/utils.c: Merged revisions 51194 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007)
|
|
| 4 lines When ast_strip_quoted was called with a zero-length
|
|
string, it would treat a NULL as if it were the quoting character
|
|
(and would thus return the string in memory immediately following
|
|
the passed-in string). ........
|
|
|
|
2007-01-17 17:36 +0000 [r51186] Jason Parker <jparker@digium.com>
|
|
|
|
* apps/app_voicemail.c: re-add "password" for realtime voicemail
|
|
|
|
2007-01-17 06:36 +0000 [r51182] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Return the correct result when directly writing out a
|
|
packet so that the core doesn't then decide to handle it the
|
|
regular way again. (issue #8833 reported by rcourtna)
|
|
|
|
2007-01-17 01:29 +0000 [r51176] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* apps/app_voicemail.c: a few more coding style cleanups and one
|
|
bug fix (from AnthonyL)
|
|
|
|
2007-01-17 00:46 +0000 [r51172] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: Move rescheduling of lagrq/pings into the
|
|
scheduler callback.
|
|
|
|
2007-01-17 00:20 +0000 [r51165-51170] Jason Parker <jparker@digium.com>
|
|
|
|
* main/rtp.c: Fix issue with dtmf continuation packets when the
|
|
dtmf digit is 0... Issue 8831
|
|
|
|
* apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with
|
|
IMAP storage and realtime voicemail. Also update the vmdb sql
|
|
script for IMAP specific options. Issue 8819, initial patches by
|
|
bsmithurst (slightly modified by me)
|
|
|
|
* doc/voicemail_odbc_postgresql.txt: change documentation to
|
|
reflect new procedure in 1.4/trunk
|
|
|
|
2007-01-16 21:51 +0000 [r51159-51162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions
|
|
51161 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007)
|
|
| 2 lines Add documentation walkthrough on getting Postgres to
|
|
work with voicemail (from Issue 8513) ........
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007)
|
|
| 2 lines Postgres driver doesn't like a NULL pointer when
|
|
retrieving the length (Bug 8513) ........
|
|
|
|
2007-01-16 17:46 +0000 [r51150] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* apps/app_voicemail.c: minor things i missed before i get jumped
|
|
on
|
|
|
|
2007-01-16 17:39 +0000 [r51148] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, res/res_features.c: Merged revisions 51145 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2
|
|
lines Return previous behavior. ParkedCalls will be able to do
|
|
DTMF based transfers again. trunk however will get an option to
|
|
allow this to be set on/off. (issue #8804 reported by nortex)
|
|
........
|
|
|
|
2007-01-16 17:36 +0000 [r51146] Jason Parker <jparker@digium.com>
|
|
|
|
* main/file.c: Display more useful output when streaming files.
|
|
Include the channel name to which the file is being played. Issue
|
|
8828, patch by junky.
|
|
|
|
2007-01-16 05:55 +0000 [r51087] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2
|
|
lines Add none as a valid callgroup/pickupgroup option. I
|
|
consider it a bug that it would inherit it all the way down and
|
|
not have any way to reset it to nothing - so that's why it is in
|
|
1.2. (issue #8296 reported by gkloepfer) ........
|
|
|
|
2007-01-16 01:15 +0000 [r51057] Russell Bryant <russell@digium.com>
|
|
|
|
* main/config.c: It is possible for the config pointer to be NULL
|
|
here, so it needs to be checked before dereferencing it.
|
|
|
|
2007-01-16 00:22 +0000 [r51030] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* apps/app_voicemail.c, configs/users.conf.sample: Patch allows for
|
|
changing voicemail password in users.conf from voicemail main,
|
|
written by AnthonyL bug #8436
|
|
|
|
2007-01-15 23:49 +0000 [r50994] Russell Bryant <russell@digium.com>
|
|
|
|
* Makefile.rules: Filter out a few CFLAGS that are not valid
|
|
CXXFLAGS.
|
|
|
|
2007-01-15 23:10 +0000 [r50988] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /: Blocked revisions 50987 via svnmerge ........ r50987 |
|
|
tilghman | 2007-01-15 17:09:02 -0600 (Mon, 15 Jan 2007) | 2 lines
|
|
Check return value before dereferencing (Bug 8822) ........
|
|
|
|
2007-01-15 21:08 +0000 [r50957] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from
|
|
https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946
|
|
| mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4
|
|
lines Solves issue with forwarding voicemails from folders other
|
|
than inbox. patch by anthonyl. ........
|
|
|
|
2007-01-15 18:23 +0000 [r50921] Jason Parker <jparker@digium.com>
|
|
|
|
* main/asterisk.c: re-add deprecated "show version" CLI command.
|
|
|
|
2007-01-15 16:36 +0000 [r50895] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/manager.c: Move event processing into do_message so that it
|
|
gets executed again when events are tripped.
|
|
|
|
2007-01-15 15:03 +0000 [r50867] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* configure, include/asterisk/autoconfig.h.in, main/Makefile,
|
|
configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the
|
|
ACX_PTHREAD macro from the Autoconf macro archive for setting up
|
|
compiler pthreads support... should improve portability to
|
|
platforms with unusual pthreads requirements
|
|
|
|
2007-01-14 21:59 +0000 [r50820] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/astmm.c: Add missing newlines for two memory CLI commands.
|
|
|
|
2007-01-14 05:13 +0000 [r50782] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c,
|
|
main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c,
|
|
main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c,
|
|
main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c,
|
|
main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c,
|
|
main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c,
|
|
main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c,
|
|
main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c,
|
|
main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c,
|
|
main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h,
|
|
main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c,
|
|
main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c,
|
|
main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c,
|
|
main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c,
|
|
main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c,
|
|
main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13
|
|
Jan 2007) | 2 lines Bug 8814 - db should look for its header
|
|
using a relative path, instead of the system path (Fixes FreeWRT)
|
|
........
|
|
|
|
2007-01-13 16:45 +0000 [r50754] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* Makefile, build_tools/make_sample_voicemail (added): when
|
|
building the sample greetings for maibox 1234@default during
|
|
'make samples', build a greeting for each language and file
|
|
format the user selected to install with menuselect (reported by
|
|
Brian Capouch on asterisk-dev)
|
|
|
|
2007-01-13 06:00 +0000 [r50674-50727] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/channel.c: Only write a frame out to the channel if one
|
|
exists. There are cases where one may not and would therefore
|
|
cause the channel driver to segfault. (issue #8434 reported by
|
|
slimey)
|
|
|
|
* res/res_snmp.c: Only join the snmp thread on an unload if the
|
|
thread is actually running. (issue #8810 reported by junky)
|
|
|
|
2007-01-12 19:24 +0000 [r50647] Jason Parker <jparker@digium.com>
|
|
|
|
* configs/voicemail.conf.sample: Update documentation to state that
|
|
you shouldn't use realtime static with voicemail.conf
|
|
|
|
2007-01-12 16:42 +0000 [r50602] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/manager.c: We need to check for res being 0 in do_message
|
|
itself, otherwise our headers will get lost.
|
|
|
|
2007-01-12 14:42 +0000 [r50562] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/pbx.c, /: Merged revisions 50561 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007)
|
|
| 2 lines minor documentation clarification ........
|
|
|
|
2007-01-11 05:53 +0000 [r50377-50468] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Remove check for channel state as it can
|
|
definitely be something other then ring, and also clean up the
|
|
code a bit. This should solve the parking issues and maybe some
|
|
attended transfer issues people have been seeing.
|
|
|
|
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add
|
|
support to see whether NAT was detected (yay symmetric RTP) and
|
|
also add a check in chan_sip so that if NAT has been detected and
|
|
the reinvite behind nat option has been turned off, then just do
|
|
partial bridge. (issue #8655 reported by mnicholson)
|
|
|
|
* apps/app_speech_utils.c: Merge speech-multi branch which adds
|
|
support for joining multiple sound files together to be played
|
|
one after another in SpeechBackground.
|
|
|
|
* main/config.c: Fix parsing when using something like ldap
|
|
settings. (done by anthonyl)
|
|
|
|
* channels/chan_sip.c: Fix chan_sip not working issue. Let's not
|
|
prematurely return 0. (issue #8783 reported by st41ker)
|
|
|
|
2007-01-10 16:45 +0000 [r50346] Jason Parker <jparker@digium.com>
|
|
|
|
* cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made
|
|
it fail to load if the config file existed. Issue 8777
|
|
|
|
2007-01-10 04:55 +0000 [r50266-50298] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_dial.c, /: Merged revisions 50295 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2
|
|
lines Add another return value to dial_exec_full that indicates
|
|
execution is going to continuing at a new
|
|
extension/context/priority and to just let it slide. (issue #8598
|
|
reported by jon) ........
|
|
|
|
* main/pbx.c: Ensure data's existence before trying to access it.
|
|
(issue #8774 reported by rcourtna)
|
|
|
|
2007-01-10 02:17 +0000 [r50228] Russell Bryant <russell@digium.com>
|
|
|
|
* Makefile, /: Merged revisions 50227 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) |
|
|
6 lines Make the number that represents the major version number
|
|
a single digit instead of 2. Using two digits makes it an octal
|
|
number when put into version.h, which breaks the compilation of
|
|
any out of tree module that checks the version for any version
|
|
after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev
|
|
mailing list, who gave credit to vihai for pointing it out)
|
|
........
|
|
|
|
2007-01-09 17:11 +0000 [r50186] Jason Parker <jparker@digium.com>
|
|
|
|
* main/cli.c: Re-add CLI command that should have only been
|
|
deprecated in 1.4. Thanks kshumard! (reported in person, so no
|
|
associated issue #)
|
|
|
|
2007-01-09 13:40 +0000 [r50151] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007)
|
|
| 4 lines The advent of realtime has enabled people to use commas
|
|
in the fullname field. This could cause an issue with sending
|
|
voicemails, when the field is unquoted. (Issue 8595) ........
|
|
|
|
2007-01-09 11:25 +0000 [r50124] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: - handle re-invites properly in sip_hangup()
|
|
- Add some invitestate status changes just to be sure
|
|
|
|
2007-01-08 23:39 +0000 [r50098] Jason Parker <jparker@digium.com>
|
|
|
|
* apps/app_voicemail.c: Fix an issue with voicemail and users.conf,
|
|
where it wouldn't ever parse a password, since it was using
|
|
"secret" instead of "password" Issue 8761, reported by and patch
|
|
suggestion from ssokol.
|
|
|
|
2007-01-08 21:11 +0000 [r50073] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* apps/app_senddtmf.c: we can't unlock a channel if we cant find
|
|
it. - AnthonyL bug #8741
|
|
|
|
2007-01-08 18:21 +0000 [r50032] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Disable the more intense packet2packet bridging until
|
|
the bugs can be worked out.
|
|
|
|
2007-01-08 14:26 +0000 [r49925-50006] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Issue #8677 - Handle failure of T.38
|
|
re-invite This is not a fix, but adding an error message to tell
|
|
the admin that we have a bad configuration. We should not send
|
|
T.38 re-invites to devices that can't handle it (with the current
|
|
architecture where you have to hard-code t.38 support per
|
|
device). To really fix this, we need to figure out a way to tell
|
|
the incoming call that the re-invite failed, so we can signal
|
|
failure on that end and go back to the original call.
|
|
|
|
* channels/chan_sip.c: Issue #8524, support multiple via header
|
|
values (tardieu) Thanks!
|
|
|
|
* channels/chan_sip.c: We only need one forward declaration
|
|
|
|
* channels/chan_sip.c: Issue 8735: Terminate state when extension
|
|
is unavailable for subscription
|
|
|
|
2007-01-08 05:11 +0000 [r49890] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2
|
|
lines Ensure we use the default refresh value of 60 if the remote
|
|
server does not send one. (issue #8746 reported by maethor)
|
|
........
|
|
|
|
2007-01-08 03:53 +0000 [r49866] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* configure, configure.ac: since we use AC_PATH_TOOL to find tools,
|
|
we should use the results it provides for us (reported by Brian
|
|
Capouch on the asterisk-dev list)
|
|
|
|
2007-01-07 21:44 +0000 [r49831-49834] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007)
|
|
| 2 lines If openstream fails, then we crash (Issue 8564)
|
|
........
|
|
|
|
* channels/chan_sip.c: Second condition was a subset of the first,
|
|
so hold was never decremented, thus hint stayed stuck (Issue
|
|
8747)
|
|
|
|
2007-01-06 00:24 +0000 [r49742] Jason Parker <jparker@digium.com>
|
|
|
|
* main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping
|
|
byte of allocated memory! This looks like it may have been a
|
|
chicken/egg scenario.. You had to call a cleanup func, because
|
|
everything was allocated. Then since you had to call a cleanup
|
|
func, you were forced to allocate - ie; strdup("").
|
|
|
|
2007-01-05 23:51 +0000 [r49710-49715] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* configure, acinclude.m4: one more time...
|
|
|
|
* configure, acinclude.m4: proper fix for r49712
|
|
|
|
* configure, acinclude.m4: if --with-foo=<path> is specific for a
|
|
configure option, ensure that it is used for header file checking
|
|
as well
|
|
|
|
* main/manager.c: ast_func_read() needs a writable copy of the
|
|
function name to be passed
|
|
|
|
2007-01-05 23:16 +0000 [r49705] Jason Parker <jparker@digium.com>
|
|
|
|
* channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and
|
|
chan_zap also depend on zaptel. This fixes an issue (8727) with
|
|
zaptel being in a different directory, using --with-zaptel.
|
|
|
|
2007-01-05 22:52 +0000 [r49676-49680] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/manager.c: don't 'consume' the params list before we try to
|
|
use it again
|
|
|
|
* res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c,
|
|
main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c,
|
|
main/db.c, channels/chan_zap.c, channels/chan_sip.c,
|
|
apps/app_meetme.c, res/res_features.c, channels/chan_agent.c,
|
|
utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c,
|
|
apps/app_queue.c, res/res_jabber.c: reduce stack consumption for
|
|
AMI and AMI/HTTP requests by nearly 20K in most cases
|
|
|
|
2007-01-05 22:14 +0000 [r49675] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/channel.c: Don't keep repeating the warning over and over
|
|
when the end of the call is reached. (issue #8724 reported by
|
|
xrg)
|
|
|
|
2007-01-05 17:09 +0000 [r49581-49636] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* /, channels/chan_sip.c, channels/chan_skinny.c,
|
|
channels/chan_iax2.c: Merged revisions 49635 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007)
|
|
| 2 lines ensure that threads which are supposed to be detached
|
|
(because we aren't going to wait on them) are created properly
|
|
........
|
|
|
|
* channels/chan_iax2.c: revert the dynamic_list insertion change...
|
|
that was not the right thing to do
|
|
|
|
* channels/chan_iax2.c: create the IAX2 processing threads as
|
|
background threads so they will use smaller stacks when we create
|
|
a dynamic thread, put it on the dynamic_list right away so we
|
|
don't lose track of it
|
|
|
|
2007-01-04 23:00 +0000 [r49568] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: It's possible for the iax2 pvt to
|
|
disappear, so if it has... don't bother looking for dpentries.
|
|
|
|
2007-01-04 22:51 +0000 [r49553] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* include/asterisk/threadstorage.h, main/asterisk.c,
|
|
build_tools/cflags.xml, include/asterisk.h, main/Makefile,
|
|
main/threadstorage.c (added), main/utils.c: add support for
|
|
tracking thread-local-storage objects that exist via
|
|
'threadstorage' CLI commands
|
|
|
|
2007-01-04 22:28 +0000 [r49551] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/config.c: Only free comments and line buffer once we reach
|
|
the first level. (issue #8678 reported by ssokol, fixed by
|
|
anthonyl)
|
|
|
|
2007-01-04 21:58 +0000 [r49460-49536] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/iax2-parser.c, main/frame.c: don't mark these
|
|
allocations as 'cache' allocations when caching has been disabled
|
|
|
|
* channels/iax2-parser.c: if we're going to decrement the frame
|
|
count when we free a frame, we should inrement it when we create
|
|
one :-)
|
|
|
|
* channels/iax2-parser.c, channels/iax2-parser.h,
|
|
channels/chan_iax2.c: only do IAX2 frame caching for voice and
|
|
video frames
|
|
|
|
* main/frame.c: don't do frame header caching in the core if
|
|
LOW_MEMORY is defined
|
|
|
|
* channels/iax2-parser.c: don't define this type either if
|
|
LOW_MEMORY is enabled
|
|
|
|
2007-01-04 18:11 +0000 [r49459] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from
|
|
https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447
|
|
| mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2
|
|
lines converted a lot of 256 to PATH_MAX and some white space
|
|
fixes. ........
|
|
|
|
2007-01-04 18:06 +0000 [r49457-49458] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode
|
|
|
|
* codecs/Makefile: make building of codec_gsm against the system
|
|
GSM library actually work
|
|
|
|
2007-01-04 16:50 +0000 [r49413] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from
|
|
https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412
|
|
| mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3
|
|
lines good catch russell sorry i missed that. fix magic number
|
|
with proper sizeof ........
|
|
|
|
2007-01-04 04:33 +0000 [r49388] Russell Bryant <russell@digium.com>
|
|
|
|
* funcs/func_realtime.c: Fix the REALTIME() dialplan function.
|
|
ast_build_string() advances the string pointer to the position to
|
|
begin the next write into the buffer. So, this pointer can not be
|
|
used to copy the contents of the string later. The beginning of
|
|
the buffer must be saved. Interestingly enough, this code could
|
|
not have ever worked. (Pointed out by Sebb on IRC, thanks!)
|
|
|
|
2007-01-03 23:32 +0000 [r49355] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from
|
|
https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354
|
|
| mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6
|
|
lines When using ODBC_STORAGE VoicemailMain doesn't create the
|
|
subdirectories for a mailbox such as the INBOX directory. this
|
|
patch solves that problem, was written by anthony be-125 ........
|
|
|
|
2007-01-03 09:06 +0000 [r49313] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
|
|
doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c,
|
|
/, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
|
|
configs/misdn.conf.sample: Merged revisions
|
|
48319,48321,48467,48552,48576,49135,49303 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) |
|
|
1 line changed a few debugs to higher debug levels ........
|
|
r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) |
|
|
1 line added the export and import of the MISDN_ADDRESS_COMPLETE
|
|
Variable to inidcate wether the extension is already completely
|
|
dialed or if there might come additional digits by information
|
|
elements. also added some docs for that. ........ r48467 |
|
|
crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
|
|
removed FIXUP state. added check for channel allocation conflict
|
|
when we create a setup while the other site creates a setup on
|
|
the same channel, besides the check we resolve this conflict.
|
|
........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18
|
|
Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a
|
|
preselected channel we just accept it, even when we're NT. added
|
|
some checks for segfaults. ........ r48576 | crichter |
|
|
2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we
|
|
reject a channel, because it's in use already, we shouldn't
|
|
process the setup anymore. made the channel allocation a bit
|
|
easier and more understandable, removed a few unused lines
|
|
........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02
|
|
Jan 2007) | 1 line added check for channel ranges in the
|
|
set/empty channel functions. set pmp_l1_check default to no.
|
|
added misdn restart pid cli command. added cleaning of channel
|
|
when we send a RELEASE_COMPLETE. ........ r49303 | crichter |
|
|
2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added
|
|
check for bridging in misdn_call to avoid setting
|
|
echocancellation when 2 mISDN channels are involved and when
|
|
bridging is set. That lead to a kernel panic before under
|
|
different situations, because we switched about 2 times between
|
|
hardware bridging and echocancelation * readded MISDN_URATE
|
|
variable which got lost before, this should make app_v110 work
|
|
again * fixed typo ........
|
|
|
|
2007-01-03 03:21 +0000 [r49282] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* Makefile, Makefile.rules: various Makefile improvements to get
|
|
chan_vpb (and any other C++ modules) to build properly
|
|
|
|
2007-01-03 01:19 +0000 [r49259] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: Check pvt structure presence before passing
|
|
to send_command. This gets rid of the irritating message about a
|
|
packet without pvt structure. This happens because the scheduled
|
|
item is getting cancelled at almost the exact moment it is
|
|
getting executed.
|
|
|
|
2007-01-02 22:30 +0000 [r49237] Steve Murphy <murf@digium.com>
|
|
|
|
* main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
|
|
pbx/ael/ael.flex: This is a slight modification to Josh's edits
|
|
for #8579; both files edited were the produced by flex; so the
|
|
source files need to be changed instead, and the generated files
|
|
regenerated.
|
|
|
|
2007-01-02 19:58 +0000 [r49212] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Small cleanup of add_t38sdp - it's always
|
|
enabled at that point in the code
|
|
|
|
2007-01-02 17:33 +0000 [r49189] Jason Parker <jparker@digium.com>
|
|
|
|
* main/pbx.c: Allow fractions of a second in the Wait()
|
|
application, like it says it allows.
|
|
|
|
2007-01-02 13:59 +0000 [r49165] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_zap.c: remove comment that is unrelated to this
|
|
function
|
|
|
|
2007-01-02 12:08 +0000 [r49145] Olle Johansson <oej@edvina.net>
|
|
|
|
* configs/features.conf.sample: Adding note on effect of
|
|
applicationmap features on re-invites
|
|
|
|
2007-01-01 23:34 +0000 [r49098-49102] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_zap.c, build_tools/menuselect-deps.in, configure,
|
|
configure.ac, codecs/codec_zap.c: check specifically for VLDTMF
|
|
and transcoding support in the system's Zaptel installation, and
|
|
make only the modules that need those features dependent on them
|
|
(this will allow building the other Zaptel-using parts of
|
|
Asterisk against older versions of Zaptel or those on other
|
|
platforms that haven't caught up yet to the Linux version)
|
|
|
|
* Makefile: use a simpler (and portable) method to ensure that
|
|
menuselect is built as a host binary
|
|
|
|
* Makefile: revert this change until a better solution can be
|
|
found... 'env -i' was not being used properly, but even when
|
|
changed to do so, this process fails during cross-compilation
|
|
because the menuselect build still sees 'CC' as set to the
|
|
cross-compiler
|
|
|
|
2007-01-01 20:14 +0000 [r49096] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: remove incomplete implementation of dnsmgr.
|
|
Let's fix this in trunk.
|
|
|
|
2006-12-30 18:31 +0000 [r49063-49073] Joshua Colp <jcolp@digium.com>
|
|
|
|
* pbx/pbx_config.c: IAX has been deprecated for quite some time so
|
|
we had better use IAX2 when creating the dial string for users.
|
|
(issue #8697 reported by ssokol)
|
|
|
|
* channels/chan_zap.c: Use asprintf to build the channel names
|
|
instead of custom function. I believe the custom function is
|
|
doing some things that are not portable across all
|
|
implementations. (issue #8570 reported by hterag & issue #8692
|
|
reported by nicolasg)
|
|
|
|
* main/rtp.c: If the Packet2Packet bridge is being broken because
|
|
of a masquerade then attempt to read a frame in so the masquerade
|
|
actually happens. Otherwise weirdness will occur. (issue #8696
|
|
reported by kjotte)
|
|
|
|
* channels/chan_iax2.c: Initialize the packet queue in load_module
|
|
instead of just declaring the list with the default value. (issue
|
|
#8695 reported by ssokol)
|
|
|
|
2006-12-30 00:40 +0000 [r49061] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have
|
|
comma args converted to vertical bars. I hope this change does
|
|
little harm.
|
|
|
|
2006-12-29 00:50 +0000 [r49042-49048] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* /: put this value into the correct property
|
|
|
|
* /, BUGS: Merged revisions 49045 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006)
|
|
| 2 lines location of the bug posting guidelines has changed
|
|
........
|
|
|
|
* sample.call: simple commit to test CIA integration
|
|
|
|
2006-12-28 21:26 +0000 [r49032-49035] Jason Parker <jparker@digium.com>
|
|
|
|
* main/cli.c: Fix some deprecated commands. Issue 8682, patch by me
|
|
|
|
* main/http.c: saw this in passing... fix a small typo
|
|
|
|
2006-12-28 20:08 +0000 [r49028] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* sounds/Makefile: new versions of sounds
|
|
|
|
2006-12-28 19:52 +0000 [r49024] Jason Parker <jparker@digium.com>
|
|
|
|
* main/http.c: make the uris_lock a rwlock instead of a mutex lock
|
|
- needs to be forward ported to trunk
|
|
|
|
2006-12-28 19:43 +0000 [r49022] Joshua Colp <jcolp@digium.com>
|
|
|
|
* configure, include/asterisk/autoconfig.h.in, configure.ac,
|
|
include/asterisk/lock.h: Backport support for read/write locks.
|
|
|
|
2006-12-28 19:21 +0000 [r49020] Steve Murphy <murf@digium.com>
|
|
|
|
* main/ast_expr2.fl, main/ast_expr2.c, main/frame.c,
|
|
pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c,
|
|
pbx/ael/ael_lex.c, include/asterisk/ael_structs.h,
|
|
pbx/ael/ael.tab.h, utils/ael_main.c: removed <err.h> as in trunk
|
|
from the ael stuff. Also, threw in a minor fix to frame.c to
|
|
avoid build-killing compiler warnings.
|
|
|
|
2006-12-27 22:28 +0000 [r49009] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not
|
|
available when LOW_MEMORY is used and things are being built in
|
|
the utils directory, so we need to resort to the old method of
|
|
strncpy. (issue #8579 reported by mottano)
|
|
|
|
2006-12-27 22:06 +0000 [r48998-49006] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/enum.c, main/asterisk.c, main/rtp.c, main/term.c,
|
|
main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c,
|
|
main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c,
|
|
main/http.c, main/logger.c: since these variables all have static
|
|
duration, none of them need initializers (they default to zero
|
|
anyway)
|
|
|
|
* include/asterisk/options.h, main/asterisk.c, main/file.c: move
|
|
extern declaration for this option to a header file where it
|
|
belongs provide an initial value for 'languageprefix' option,
|
|
instead of relying on randomness to provide a useful value
|
|
|
|
2006-12-27 21:06 +0000 [r48993-48997] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Only include acl.h and lock.h once
|
|
|
|
* channels/chan_sip.c: Only set rfc2833compensate flag once
|
|
(handle_common_options)
|
|
|
|
* channels/chan_sip.c: - Remove checking for T38 options twice.
|
|
Keeping them in handle_common_options
|
|
|
|
2006-12-27 18:33 +0000 [r48987-48988] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_sip.c: make the option actually match the
|
|
documentation
|
|
|
|
* channels/iax2-parser.c, include/asterisk/utils.h,
|
|
include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show
|
|
memory' and 'show memory summary' to distinguish memory
|
|
allocations that were done for caching purposes, so they don't
|
|
look like memory leaks
|
|
|
|
2006-12-27 17:59 +0000 [r48975-48985] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c, configs/sip.conf.sample: Be a bit more
|
|
politically correct
|
|
|
|
* channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy
|
|
cisco MWI support. Normally we try not to change our software for
|
|
bugs in other devices. But in this case, the Cisco phones are so
|
|
widespread so we try to implement a fix while waiting for a
|
|
bugfix from Cisco.
|
|
|
|
* channels/chan_sip.c: - Make sure handle_common_options return 1
|
|
when we found a common option - Move uncommon (only global)
|
|
option away from handle_common_options Reported by rizzo. Thanks!
|
|
|
|
* channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before
|
|
re-sending invite with auth.
|
|
|
|
* /, channels/chan_sip.c: Fix bogus content-length in t38 sdp.
|
|
(rizzo, #8600)
|
|
|
|
2006-12-26 05:20 +0000 [r48960-48966] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_meetme.c: Get rid of a needless memory allocation and
|
|
only create a conference structure in find_conf_realtime if data
|
|
was read from realtime. (issue #8669 reported by robl)
|
|
|
|
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an
|
|
API call that initializes an RTP structure. We need this because
|
|
chan_sip is cheeky and uses a temporary RTP structure for codec
|
|
purposes, and the API calls that are used rely on the lock.
|
|
(Pointed out on asterisk-dev by Andy Wang)
|
|
|
|
* configure, configure.ac: Clean up autoconf file (gets rid of
|
|
warnings seen when rebuilding configure) and rebuild configure.
|
|
|
|
2006-12-25 05:21 +0000 [r48931-48956] Russell Bryant <russell@digium.com>
|
|
|
|
* /, funcs/func_math.c: Merged revisions 48955 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) |
|
|
6 lines Fix an error introduced by copying and pasting the
|
|
handling of the >= operator for the MATH function. If a single
|
|
equal sign was used as an operator, the function would treat it
|
|
is as if it were the >= operator. Now, it properly handles it as
|
|
an invalid operator. (issue #8665, patch by tempest1) ........
|
|
|
|
* channels/chan_oss.c: Fix a typo in an error message that
|
|
indicated that the MGCP channel type could not be registered,
|
|
instead of the correct type, OSS.
|
|
|
|
* /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) |
|
|
3 lines Check for the proper return value on an error in a call
|
|
to mmap(). This was reported by Andy Wang on the asterisk-dev
|
|
list. Thanks! ........
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) |
|
|
3 lines Remove a couple of misplaced dots in log messages. This
|
|
was reported by Andrea Spadaccini on the asterisk-dev mailing
|
|
list. ........
|
|
|
|
* main/http.c: Implement locking for the list of URI handlers to
|
|
make it thread-safe.
|
|
|
|
2006-12-23 Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* Asterisk 1.4.0 released.
|
|
|
|
2006-12-22 22:33 +0000 [r48870-48906] Jason Parker <jparker@digium.com>
|
|
|
|
* Makefile, main/stdtime/localtime.c: Minor fixes for Solaris.
|
|
|
|
* channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia
|
|
|
|
2006-12-21 20:26 +0000 [r48783] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2
|
|
lines Add new silence sound files to the spec for Redhat. (issue
|
|
#8652 reported by alvaro_palma_aste) ........
|
|
|
|
2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c: vms doesn't exist on non-IMAP storage
|
|
builds.
|
|
|
|
* apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so
|
|
it is then passed to the IMAP store file function. (issue #8614
|
|
reported by punknow)
|
|
|
|
* doc/snmp.txt: find is not the same as bind when it comes to
|
|
documentation. (issue #8626 reported by johann8384)
|
|
|
|
2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/Makefile: suppress compiler warnings in this module
|
|
until it can be improved
|
|
|
|
2006-12-19 21:12 +0000 [r48585] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_dial.c, /: Merged revisions 48584 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2
|
|
lines Free localuser structure when we fail to dial (issue #8612
|
|
reported by rizzo) ........
|
|
|
|
2006-12-19 21:03 +0000 [r48583] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* apps/app_sms.c: fix a bogus datalen in the frames generated by
|
|
app_sms (causing noisy output if you listen to the output!) This
|
|
affects trunk as well, whereas 1.2 is ok.
|
|
|
|
2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable
|
|
type for these unixODBC API calls, eliminating warnings on 64-bit
|
|
platforms that use the 'new' 64-bit types for ODBC API calls
|
|
|
|
2006-12-19 03:46 +0000 [r48571] Joshua Colp <jcolp@digium.com>
|
|
|
|
* Makefile: Use env -i to start a fresh environment when going to
|
|
build menuselect. This is more portable then using unset. (issue
|
|
#8543 reported by jtodd)
|
|
|
|
2006-12-18 17:23 +0000 [r48566] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* include/asterisk/channel.h: unbreak the macro used for
|
|
incrementing the frame counters. I don't know when the bug was
|
|
introduced, but with the typical usage c->fin =
|
|
FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects
|
|
trunk as well (fix coming).
|
|
|
|
2006-12-18 17:15 +0000 [r48564] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: Put thread into proper list if we abort
|
|
handling due to an error, and also hold the lock while putting it
|
|
back into the proper idle list so we don't prematurely get a
|
|
signal. (issue #8604 reported by arkadia)
|
|
|
|
2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile,
|
|
utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile,
|
|
utils/ael_main.c: remove some now-unnecessary explicit includes
|
|
of autoconfig.h clean up per-file dependencies during 'make
|
|
clean'
|
|
|
|
* build_tools/prep_tarball: need an additional argument here to
|
|
make the downloads actually occur
|
|
|
|
* configure, include/asterisk/autoconfig.h.in, configure.ac,
|
|
acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep
|
|
these calls from thinking they have multiple arguments
|
|
|
|
* codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile,
|
|
funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast,
|
|
main, codecs/gsm, pbx, res, channels, codecs, utils, agi,
|
|
main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr:
|
|
simplify dependency tracking system, using the compiler's
|
|
built-in method for generating them, and only doing dependency
|
|
tracking if developer mode is enabled via the configure script
|
|
|
|
* Makefile, include/asterisk.h, main/stdtime/localtime.c: since we
|
|
really, really have to have autoconfig.h included before all
|
|
other headers (especially system headers), the Makefile will now
|
|
force it to happen (this will fix build problems with files like
|
|
ast_expr2f.c, where we can't control the inclusion order in the
|
|
file itself)
|
|
|
|
* funcs/func_curl.c: instead of initializing the curl library every
|
|
time the CURL() function is invoked, do it only once per thread
|
|
(this allows multiple calls to CURL() in the dialplan for a
|
|
channel to run much more quickly, and also to re-use connections
|
|
to the server) (thanks to JerJer for frequently complaining about
|
|
this performance problem)
|
|
|
|
2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Turn payload_lock into bridge_lock and make it
|
|
encompass all RTP structure contents that may relate to bridge
|
|
information, including who we are bridged to.
|
|
|
|
* channels/chan_iax2.c: Hold call structure lock in places where a
|
|
qualify or peer action can destroy it.
|
|
|
|
* channels/chan_iax2.c: Lock network retransmission queue in all
|
|
places that it is used.
|
|
|
|
2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported
|
|
from 1.2)
|
|
|
|
* channels/chan_sip.c: Update to latest IANA spec
|
|
|
|
2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: Use a wakeup variable so that we don't wait
|
|
on IO indefinitely if packets need to be retransmitted.
|
|
|
|
* main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP
|
|
structure can change AFTER a bridge has started. This comes from
|
|
the packet handling of the SIP response when indication that it
|
|
was answered has been sent. Therefore we need to protect this
|
|
data with a lock when we read/write. (issue #8232 reported by
|
|
tgrman)
|
|
|
|
* main/rtp.c: Remove direct RTCP bridging. I've come to the
|
|
conclusion that we should handle this through the core and not
|
|
just forward it on. Should solve a few bugs.
|
|
|
|
2006-12-12 Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* Asterisk 1.4.0-beta4 released.
|
|
|
|
2006-12-12 04:13 +0000 [r48401] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This
|
|
is the way it should have been done.
|
|
|
|
2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* sounds/Makefile: new sounds package with 100% more silence
|
|
|
|
* /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge
|
|
from https://svn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
|
|
| 4 lines app_externalivr needs a real silence file, and
|
|
additional changes to add silence files into core instead of
|
|
extra patch provided by bug 8177 with minor additions. ........
|
|
|
|
2006-12-11 21:31 +0000 [r48391] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c: Return non-existant callerid handling to
|
|
that which it was before. In 1.4 and trunk callerid can be
|
|
allocated but not have any contents so we have to use
|
|
ast_strlen_zero before passing it to the relevant functions.
|
|
(issue #8567 reported by pabelanger)
|
|
|
|
2006-12-11 05:37 +0000 [r48382] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* funcs/func_strings.c: STRFTIME() does not actually require an
|
|
argument (issue 8540)
|
|
|
|
2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Merge in my latest RTP changes. Break out RTP and
|
|
RTCP callback functions so they no longer share a common one.
|
|
|
|
* apps/app_meetme.c: Use the correct API call to say a device state
|
|
changed. (Yes, I'm a nub.)
|
|
|
|
* apps/app_meetme.c: Don't access the conference structure after it
|
|
has been freed.
|
|
|
|
2006-12-11 00:47 +0000 [r48375] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
|
|
res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
|
|
apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374
|
|
via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006)
|
|
| 5 lines When doing a fork() and exec(), two problems existed
|
|
(Issue 8086): 1) Ignored signals stayed ignored after the exec().
|
|
2) Signals could possibly fire between the fork() and exec(),
|
|
causing Asterisk signal handlers within the child to execute,
|
|
which caused nasty race conditions. ........
|
|
|
|
2006-12-10 03:04 +0000 [r48372] Steve Murphy <murf@digium.com>
|
|
|
|
* channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
|
|
line This version applies the patch suggested by stevens in bug
|
|
7836 (make inbound channel RINGING state consistent with other
|
|
channels). ........
|
|
|
|
2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_iax2.c: Use locking when accessing the
|
|
registrations list. This list is not actually used very often, so
|
|
the likelihood of there being a problem is pretty small, but
|
|
still possible. For example, if the CLI command to list the
|
|
registrations was called at the same time that a reload was
|
|
occurring and the registrations list was getting destroyed and
|
|
rebuilt, a crash could occur. In passing, go ahead and convert
|
|
this list to use the linked list macros.
|
|
|
|
* /: Blocked revisions 48361 via svnmerge ........ r48361 | russell
|
|
| 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use
|
|
locking when accessing the registrations list. This list is not
|
|
actually used very often, so the likelihood of there being a
|
|
problem is pretty small, but still possible. For example, if the
|
|
CLI command to list the registrations was called at the same time
|
|
that a reload was occurring and the registrations list was
|
|
getting destroyed and rebuilt, a crash could occur. ........
|
|
|
|
2006-12-07 18:17 +0000 [r48357] Russell Bryant <russell@digium.com>
|
|
|
|
* /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
|
|
Dec 2006) | 3 lines Ensure that the file position is not
|
|
incremented beyond the total number of files available for
|
|
playback. (issue #8539, ulogic) ........
|
|
|
|
2006-12-07 15:33 +0000 [r48349] Steve Murphy <murf@digium.com>
|
|
|
|
* main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that
|
|
killed bug 8423 -- OriginateSuccess and OriginateError incomplete
|
|
channel name. May it rest in peace.
|
|
|
|
2006-12-06 16:25 +0000 [r48326] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being
|
|
retransmitted to Asterisk
|
|
|
|
2006-12-06 16:15 +0000 [r48323] Russell Bryant <russell@digium.com>
|
|
|
|
* configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
|
|
Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
|
|
in the sample configuration file. (issue #8526, arkadia) ........
|
|
|
|
2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Don't send Contact on MESSAGE
|
|
|
|
2006-12-05 20:42 +0000 [r48279] Jason Parker <jparker@digium.com>
|
|
|
|
* configure.ac: Fix curl version number testing to be much more
|
|
friendly to non-bash shells. Issue 8508, patch by me. This
|
|
*SHOULD* be POSIX compliant now..
|
|
|
|
2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Merging the invitestate-1.4 branch after
|
|
successful testing. Will check if I can solve this with less
|
|
changes in 1.2.
|
|
|
|
* configs/sip.conf.sample: Add missing s from another repository.
|
|
(thanks jcmoore!)
|
|
|
|
* configs/sip.conf.sample: Updating sip.conf.sample with
|
|
information about T38 not working when chan_local or chan_agent
|
|
is involved in the call. I don't know how big a fix that would be
|
|
to solve, but this is the current state of affairs. (Chan_sip
|
|
currently checks if the other side of the bridge has a SIP tech.
|
|
We could/should implement another check, possibly for udptl_write
|
|
or some flag in the ast_channel structure).
|
|
|
|
2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_voicemail.c: Oops, forgot to release the odbc handle
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006)
|
|
| 6 lines If the recording in the database is too large, it will
|
|
fail to retrieve with an mmap error. Not too sure why this
|
|
doesn't happen when we put it in the database, also, but since
|
|
that doesn't seem to be broken, I'm not going to fix it (at least
|
|
until someone reports it). Solution is to ask for the file in
|
|
smaller chunks. (Bug 8385) ........
|
|
|
|
2006-12-04 21:48 +0000 [r48237-48248] Jason Parker <jparker@digium.com>
|
|
|
|
* apps/app_voicemail.c: Fix an issue which didn't allow
|
|
unavail/greet/busy/etc messages from being saved into ODBC (and
|
|
probably IMAP).
|
|
|
|
* /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell |
|
|
2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert
|
|
change from 8016 - this breaks other stuff... Needs further
|
|
review. Tip: When you've reported a bug about something and
|
|
somebody has put up a patch for it.. It's not a good idea to open
|
|
a completely new bug and say that something is broken because of
|
|
the patch in the other bug - PLEASE mention something in the bug
|
|
where the patch was actually created. ........
|
|
|
|
* /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell |
|
|
2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an
|
|
issue where a message isn't saved correctly when using ODBC
|
|
storage and reviewing a message. Issue 8016 - patch by sokhapkin.
|
|
........
|
|
|
|
2006-12-04 18:16 +0000 [r48234] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /: Blocked revisions 48233 via svnmerge ........ r48233 | file |
|
|
2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the
|
|
generic bridge tells us not to retry, and we have a frame to spit
|
|
out then break the bridge. Props to markit in #asterisk-bugs for
|
|
bringing this up. ........
|
|
|
|
2006-12-04 17:54 +0000 [r48228-48230] Jason Parker <jparker@digium.com>
|
|
|
|
* configs/voicemail.conf.sample: Add documentation to
|
|
voicemail.conf.sample for ODBC storage. Issue 8499 - patch by
|
|
blitzrage.
|
|
|
|
* doc/snmp.txt: Attempt to document some of the dependencies that
|
|
are needed for net-snmp Issue 8499 - initial patch by blitzrage.
|
|
|
|
2006-12-03 06:34 +0000 [r48223] Russell Bryant <russell@digium.com>
|
|
|
|
* sounds/Makefile: When "fetch" is in use, instead of "wget",
|
|
--continue is not a valid option. (issue #8451)
|
|
|
|
2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: - Removing one of two pieces of code to
|
|
handle 481 response on INVITE - Move handling of REFER response
|
|
to handle_response_refer()
|
|
|
|
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
|
|
configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax
|
|
transmission happens - Encapsulate RTP timers in the rtp
|
|
structure so we have one for video and one for audio The video
|
|
one is not used in 1.4, really. Will be used for RTP keepalives
|
|
when we can send something that video phones support in the RTP
|
|
stream. I now this is a big architectual change at this stage for
|
|
1.4, but decided it was needed to avoid future bug reports. -
|
|
Document the RTP NAT keepalive option in sip.conf.sample Issue
|
|
7679 in the bug tracker. Please test.
|
|
|
|
2006-12-02 03:50 +0000 [r48195] Russell Bryant <russell@digium.com>
|
|
|
|
* include/asterisk/utils.h: Backport the comment containing the
|
|
warning regarding the limitations on the usage of this function.
|
|
It is thread safe, but not technically reentrant.
|
|
|
|
2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* apps/app_dial.c, /: Merged revisions 48192 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006)
|
|
| 2 lines if Dial() is going to send music-on-hold to the calling
|
|
party, it has to send PROGRESS first to ensure that the reverse
|
|
audio path has been setup first (BE-106) ........
|
|
|
|
2006-12-01 23:16 +0000 [r48190] Russell Bryant <russell@digium.com>
|
|
|
|
* Makefile, configure, configure.ac, makeopts.in, sounds/Makefile:
|
|
FreeBSD 6.1 does not include wget by default. However, it has
|
|
fetch which will work just fine for our purposes of downloading
|
|
the sounds packages. So, check for both wget and fetch and the
|
|
configure script and use what was found to download them. If
|
|
neither one was found, and sound packages are selected that must
|
|
be downloaded, the install process will print out an informative
|
|
error message indicating the situation. Also, fix a couple places
|
|
where "make" was hard coded into some output messages by
|
|
replacing them with the $(MAKE) variable. (issue #8451, initial
|
|
patch by pabelanger, with additional modifications by me)
|
|
|
|
2006-12-01 20:25 +0000 [r48184-48186] Jason Parker <jparker@digium.com>
|
|
|
|
* configs/extensions.conf.sample, /: Merged revisions 48183 via
|
|
svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
|
|
lines Fix a small typo - issue 8848, reported by pabelanger
|
|
........
|
|
|
|
2006-12-01 19:38 +0000 [r48179] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/cli.c: Double-unlock error (reported by blitzrage on IRC)
|
|
|
|
2006-12-01 17:41 +0000 [r48177] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c, configs/sip.conf.sample: - Backport of the
|
|
"limitonpeers" patch from trunk, to fix a lot of issues with
|
|
queues and SIP device states - Remove support for T.38 early
|
|
media, since it's impossible. (Two patches in one - extra friday
|
|
evening offer due to being off line from svn today... :-)
|
|
|
|
2006-11-30 21:18 +0000 [r48168] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not
|
|
do a partial bridge for Google Talk since we need to handle STUN.
|
|
(issue #8448 reported by phsultan)
|
|
|
|
2006-11-30 20:51 +0000 [r48166] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Issue 8319 - change noncecount before
|
|
using it.
|
|
|
|
2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /: Blocked revisions 48161 via svnmerge ........ r48161 | file |
|
|
2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't
|
|
write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel
|
|
driver. (issue #8390 reported by hselasky) ........
|
|
|
|
* /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
|
|
lines Only print out debug message if bridged channel is not
|
|
NULL. (issue #8412 reported by jubilex) ........
|
|
|
|
* /, res/res_features.c: Merged revisions 48154 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
|
|
lines Do not listen for DTMF on the bridge that comes into
|
|
existence when ParkedCall is executed. This means native bridging
|
|
can now occur for this. (issue #8406 reported by kebl0155)
|
|
........
|
|
|
|
* main/cdr.c, /: Merged revisions 48151 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
|
|
lines Print certain CDR messages out at the NOTICE level versus
|
|
WARNING since they can occur when used with the CDR applications
|
|
and are perfectly fine. (issue #8367 reported by dartvader)
|
|
........
|
|
|
|
* /: Blocked revisions 48146 via svnmerge ........ r48146 | file |
|
|
2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember
|
|
the pointer to the allocated block of memory so that we can free
|
|
it and not cause a memory leak. (issue #8449 reported by arkadia)
|
|
........
|
|
|
|
* /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov
|
|
2006) | 2 lines Document 'port' for SIP peers, came up because of
|
|
the current mailing list thread. (issue #8450 reported by
|
|
blitzrage) ........
|
|
|
|
2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson <oej@edvina.net>
|
|
|
|
* doc/manager.txt: Explain status reports and make codefreeze more
|
|
happy :-)
|
|
|
|
* /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by
|
|
GS 487 adapter without CSEQ on separate line in the REGISTER
|
|
request. Imported from 1.2.
|
|
|
|
2006-11-29 21:05 +0000 [r48115] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in
|
|
mm_login. (issue #8420 reported by slimey)
|
|
|
|
2006-11-29 19:56 +0000 [r48113] Olle Johansson <oej@edvina.net>
|
|
|
|
* configs/sip.conf.sample: Explain the use device status system
|
|
implemented in SIP for subscriptions, queues and manager a bit
|
|
better. Like in 1.2, you will get more detailed information if
|
|
you set a call limit for a device. When the call limit is
|
|
reached, the status system will report a device as busy. For
|
|
queues, setting a call limit per SIP device is propably a
|
|
requirement. In most cases, it will work much better if you only
|
|
use type=peer and not type=friend. We might decide to backport
|
|
the new setting from trunk to apply all call limits to the peer
|
|
part of a friend only.
|
|
|
|
2006-11-29 16:50 +0000 [r48107] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c, /: Merged revisions 48106 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
|
|
lines If the frame was duplicated before writing out then we need
|
|
to free it. (issue #8429 reported by edguy3) ........
|
|
|
|
2006-11-29 08:03 +0000 [r48105] Olle Johansson <oej@edvina.net>
|
|
|
|
* configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma.
|
|
|
|
2006-11-29 04:26 +0000 [r48101] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c: Don't crash if the mailstream was not
|
|
created.
|
|
|
|
2006-11-28 18:26 +0000 [r48095] Jason Parker <jparker@digium.com>
|
|
|
|
* Makefile: Export several more variables in top level Makefile.
|
|
Inspired by issue 8438.
|
|
|
|
2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_phone.c, /: Merged revisions 48087 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov
|
|
2006) | 2 lines According to the research I have done we never
|
|
needed to include compiler.h in the first place so let's not!
|
|
(issue #8430 reported by edguy3) ........
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2
|
|
lines Use the proper function to get the new message count
|
|
instead of always using the filesystem. (issue #8421 reported by
|
|
slimey) ........
|
|
|
|
2006-11-27 17:20 +0000 [r48049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27
|
|
Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381)
|
|
........
|
|
|
|
2006-11-27 17:17 +0000 [r48046] Russell Bryant <russell@digium.com>
|
|
|
|
* main/manager.c: Remove a couple of unused variables (issue #8380,
|
|
casper)
|
|
|
|
2006-11-27 15:32 +0000 [r48038] Joshua Colp <jcolp@digium.com>
|
|
|
|
* pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2
|
|
lines Do not reference the freed outgoing structure in the debug
|
|
message. (issue #8425 reported by arkadia) ........
|
|
|
|
2006-11-27 06:41 +0000 [r48031] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Change logging message
|
|
|
|
2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy <murf@digium.com>
|
|
|
|
* funcs/func_cdr.c: might as well also document the raw values of
|
|
the flag vars
|
|
|
|
* /, funcs/func_cdr.c: A little bit of func_cdr documentation
|
|
upgrade-- no bug# involved, although 8221 may have inspired it.
|
|
|
|
2006-11-25 09:28 +0000 [r48002] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4
|
|
and future releases, you can disable subscription support totally
|
|
or per peer in sip.conf with allowsubscribe = yes | no
|
|
|
|
2006-11-24 17:17 +0000 [r47992] Steve Murphy <murf@digium.com>
|
|
|
|
* main/translate.c: bug 8189 posted this fix for main/translate.c
|
|
for PLC
|
|
|
|
2006-11-24 15:46 +0000 [r47989] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
|
|
channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23
|
|
Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE.
|
|
beatufied some logs, changed some loglevels. changed the default
|
|
value of block_on_alarm ........
|
|
|
|
2006-11-23 11:01 +0000 [r47959] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Don't allocate unused variable.
|
|
|
|
2006-11-22 21:47 +0000 [r47944] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Video will never reach Packet2Packet bridging and can
|
|
do more harm then good.
|
|
|
|
2006-11-21 17:32 +0000 [r47897] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: If we have the non standard G726-32 setting turned on
|
|
we want to return G726-32 to the SDP, not our AAL2 string. (issue
|
|
#8330 reported by voipgate)
|
|
|
|
2006-11-21 15:20 +0000 [r47892] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Apparently Exosip sends a 101 after a 100
|
|
provisional response. Let's not treat that as early media.
|
|
(discovered at the AVTF meeting in Paris).
|
|
|
|
2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_voicemail.c: Oops, merge missed release of odbc object
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006)
|
|
| 2 lines Failing to trap -1 error from mmap causes segfault
|
|
(Issue 8385) ........
|
|
|
|
2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/frame.c, /: Merged revisions 47859 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2
|
|
lines Don't forget to byte swap if we are exiting the smoother
|
|
feed early. (issue #8287 reported by arturs) ........
|
|
|
|
* /: Blocked revisions 47855 via svnmerge ........ r47855 | file |
|
|
2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines Free
|
|
history items at the end of use of the temporary SIP pvt
|
|
structure. (issue #8383 reported by benh) ........
|
|
|
|
* main/rtp.c: Only remove/destroy the RTCP I/O item if it exists.
|
|
|
|
* .cleancount, apps/app_dial.c, apps/app_directed_pickup.c,
|
|
include/asterisk/channel.h: Use a separate variable in the
|
|
channel structure to store the context that the channel was
|
|
dialed from. (issue #8382 reported by jiddings)
|
|
|
|
2006-11-20 11:45 +0000 [r47843-47845] Olle Johansson <oej@edvina.net>
|
|
|
|
* configs/sip.conf.sample: Explain properly how videosupport works.
|
|
Committ from Asterisk Video Task Force meeting in Paris!
|
|
|
|
* /, channels/chan_sip.c: Make sure we destroy scheduled items and
|
|
not use them ever again after destruction (rizzo)
|
|
|
|
2006-11-18 17:59 +0000 [r47823] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* channels/chan_sip.c: fix bug 7450 - Parsing fails if From header
|
|
contains angle brackets (the bug was only in a corner case where
|
|
the < was right after the opening quote, and the fix is trivial).
|
|
|
|
2006-11-16 23:19 +0000 [r47781-47782] Jason Parker <jparker@digium.com>
|
|
|
|
* apps/app_db.c, apps/app_dial.c: Fix a couple of typos. Initially
|
|
pointed out by mrobinson.
|
|
|
|
* /: Blocked revisions 47780 via svnmerge ........ r47780 | qwell |
|
|
2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a
|
|
couple of typos in applications.. Initially spotted by mrobinson.
|
|
........
|
|
|
|
2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* /, doc/billing.txt: update documentation regarding IAX2 transfers
|
|
and CDRs Merged revisions 47776 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006)
|
|
| 2 lines update clearly wrong documentation regarding cdr_custom
|
|
........
|
|
|
|
2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Compare technology using the pointers
|
|
instead of a straight comparison based on name. (issue #8228
|
|
reported by dean bath)
|
|
|
|
* /: Blocked revisions 47761 via svnmerge ........ r47761 | file |
|
|
2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines Look for
|
|
the header file specifically in all cases, not just the existence
|
|
of the directory. (issue #8358 reported by mrness) ........
|
|
|
|
2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* configure, configure.ac: check for pre-1.4 versions of Zaptel and
|
|
abort the configure script if found with an appropriate error
|
|
message
|
|
|
|
2006-11-16 19:24 +0000 [r47755] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD
|
|
notification optional, in order to avoid a lot of extra database
|
|
lookups for all those realtime users out there.
|
|
|
|
2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_local.c, /: Merged revisions 47750 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov
|
|
2006) | 2 lines Because of the way chan_local is written we
|
|
should be extra careful and make sure our callback functions have
|
|
a tech_pvt. (issue #8275 reported by mflorell) ........
|
|
|
|
* apps/app_meetme.c: Don't unreference the SLA object if there is
|
|
no SLA object in the devicestate callback. (issue #8354 reported
|
|
by loloski)
|
|
|
|
2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Don't fixup if there's nothing to fixup
|
|
|
|
* UPGRADE.txt: Warn users about change in canreinvite
|
|
|
|
* channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never
|
|
authenticated (according to the RFC) - Update docs on
|
|
canreinvite. "nonat" is the recommended setting for most users
|
|
with phones behind a NAT.
|
|
|
|
2006-11-15 22:31 +0000 [r47712] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_local.c, /: Merged revisions 47711 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov
|
|
2006) | 2 lines Make sure that the pvt structure exists before
|
|
trying to do fixup on Local channels. (issue #7937 reported by
|
|
mada123, fix by alamantia with mods by me) ........
|
|
|
|
2006-11-15 21:56 +0000 [r47709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL
|
|
|
|
2006-11-15 21:33 +0000 [r47707] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/channel.c: We need to ensure timelimit stuff is included as
|
|
well so warnings get played. (issue #8050 reported by KNK)
|
|
|
|
2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/file.c: don't try to call fclose() if fopen() failed
|
|
|
|
2006-11-15 20:31 +0000 [r47698] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: - Improve SIP history - Never send reply to
|
|
ACK (again...)
|
|
|
|
2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006)
|
|
| 4 lines ensure that message duration is included in email
|
|
notifications for forwarded messages (BE-96, fix by me after
|
|
corydon used his clue-bat on me) ensure that duration in the
|
|
message metadata is updated if prepending is done during
|
|
forwarding (related to BE-96) remove prototype for API call that
|
|
does not exist ........
|
|
|
|
* main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15
|
|
Nov 2006) | 2 lines clear the category's variable tail pointer as
|
|
well when variables are detached from it ........ r47688 |
|
|
kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2
|
|
lines when appending a list of variable to a category, ensure the
|
|
tail pointer points to the last variable in the list ........
|
|
r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006)
|
|
| 2 lines when re-writing the config file, don't repeat the path
|
|
if it hasn't changed ........
|
|
|
|
* main/config.c, /: Merged revisions 47682 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006)
|
|
| 2 lines ouch... don't use printf, use ast_log/ast_verbose
|
|
........
|
|
|
|
2006-11-15 17:46 +0000 [r47672] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* main/cli.c: fix longest match search in find_cli. Trunk already
|
|
fixed. 1.2 not affected (well, i have no idea, the code is
|
|
totally different there).
|
|
|
|
2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Send error message when we can't allocate
|
|
SIP dialog, possibly due to limitation of file descriptors.
|
|
(imported from 1.2)
|
|
|
|
2006-11-15 04:45 +0000 [r47645] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: If NAT detection is turned on or already detected
|
|
then say NAT is active when setting the remote RTP peer when
|
|
doing early bridging. (issue #8365 reported by marcelbarbulescu)
|
|
|
|
2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/term.c: more formatting cleanup, and avoid running off the
|
|
end of the string
|
|
|
|
2006-11-15 00:14 +0000 [r47639] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Turn notice about unknown RTCP packet type into a
|
|
debug message instead.
|
|
|
|
2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/misdn/isdn_lib.c: silence compiler warning on 64-bit
|
|
platforms (this variable is an 'int' anyway, comparing it to
|
|
'signed long' is not useful)
|
|
|
|
2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2
|
|
lines Update copyright information in the ADSI logo blob.
|
|
........
|
|
|
|
* channels/chan_sip.c: Only keep the video RTP structure around if
|
|
1. Video support is enabled and 2. A video codec is enabled on
|
|
the dialog
|
|
|
|
* funcs/func_uri.c: Small documentation clarification for
|
|
URIENCODE. (issue #8294 reported by salaud)
|
|
|
|
2006-11-14 18:54 +0000 [r47621] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_voicemail.c: Conversion of res_odbc API to include ast_
|
|
prefix did not completely transition app_voicemail when
|
|
ODBC_STORAGE is used (reported on IRC by caio1982, not in
|
|
bugtracker)
|
|
|
|
2006-11-14 16:45 +0000 [r47617] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_amd.c: Use LOG_DEBUG to print out the indication that
|
|
app_amd is using default settings instead of using LOG_NOTICE.
|
|
This stops needless logging of this information under normal
|
|
circumstances. (issue #8361 reported by Seb7)
|
|
|
|
2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Update documentation to fit the
|
|
implementation...
|
|
|
|
* /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in
|
|
retransmission system if it's an OPTION packet from peerpoke
|
|
|
|
2006-11-13 21:28 +0000 [r47584] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2
|
|
lines Initialize global pointers for connection and result to
|
|
NULL. (issue #8356 reported by james) ........
|
|
|
|
2006-11-13 20:20 +0000 [r47581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006)
|
|
| 2 lines Having more than 255 old messages caused corruption in
|
|
the new/old count ........
|
|
|
|
2006-11-13 19:15 +0000 [r47576] Steve Murphy <murf@digium.com>
|
|
|
|
* main/config.c: This solves bug 8342, whereby a crash occurs under
|
|
certain circumstances while reading a config file with comments--
|
|
a call to CB_ADD shouldn't happen if withcomments is zero
|
|
|
|
2006-11-13 19:11 +0000 [r47573] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/cli.c, channels/chan_sip.c: Re-enable old deprecated
|
|
commands
|
|
|
|
2006-11-13 19:10 +0000 [r47572] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: - Don't reply to INVITE already replied
|
|
to when we get BYE - Declare errmsg as int. Oops.
|
|
|
|
2006-11-13 18:18 +0000 [r47564] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing
|
|
the messed if, but we all forgot to update the regressions. Until
|
|
now.
|
|
|
|
2006-11-13 17:13 +0000 [r47553] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/pbx_ael.c: AEL need not complain about parkedcalls not being
|
|
found... just confuses users
|
|
|
|
2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, apps/app_sms.c: Merged revisions 47549 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2
|
|
lines When sending an SMS with a user data header properly set
|
|
the UDH flag in the first byte. (issue #8347 reported by
|
|
hoffmeis) ........
|
|
|
|
* main/cli.c: Free full command string upon unregistering of CLI
|
|
command. Backported from revision 47536 from rizzo.
|
|
|
|
2006-11-13 16:00 +0000 [r47540] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Only produce error message about sip history
|
|
once
|
|
|
|
2006-11-13 05:48 +0000 [r47527] Russell Bryant <russell@digium.com>
|
|
|
|
* configure, acinclude.m4: AC_PROG_SED is included in autoconf
|
|
2.60, but apparently it is not included in 2.59. So, to maintain
|
|
compatability with 2.59 since it is a small change, copy this
|
|
macro into acinclude.m4 and rename it to AST_PROG_SED. (issue
|
|
#8345)
|
|
|
|
2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* res/res_odbc.c, /: Merged revisions 47525 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006)
|
|
| 2 lines If the execute fails a second time, make sure that we
|
|
don't pass back a stale handle ........
|
|
|
|
* channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006)
|
|
| 2 lines Don't play dialtone if the seizing the channel fails
|
|
(Bug 7754) ........
|
|
|
|
2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks
|
|
DEA!!!)
|
|
|
|
* channels/chan_sip.c: Part of issue 8078 - parse even if udptl is
|
|
UDPTL in sdp...
|
|
|
|
* channels/chan_sip.c: - Don't destroy SIP dialog because of a
|
|
failed T.38 re-invite. Wait for a bye. Final response to a
|
|
re-invite does not mean that the session dies, only that the
|
|
re-invite fails. - Keep RTP active during processing of T.38
|
|
re-invite. If the re-invite fails, RTP needs to remain as before
|
|
the re-invite. Issue 8338 - darren1713. Please test.
|
|
|
|
* channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp
|
|
-Add some comments to t.38 code
|
|
|
|
2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant <russell@digium.com>
|
|
|
|
* /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) |
|
|
4 lines Only do the check to determine whether the channel
|
|
calling this function is an IAX2 channel when getting the IP
|
|
address using the special argument, CURRENTCHANNEL. (issue #8341,
|
|
jcovert) ........
|
|
|
|
* Makefile: Add the target "menuconfig" as an alias for the
|
|
"menuselect" target. This is just a favor to users so that if you
|
|
accidentally type "make menuconfig" instead of "make menuselect",
|
|
it still works. (inspired by a comment on IRC from wangster
|
|
calling me an "especially devious asterisk developer" for having
|
|
it be menuselect instead of menuconfig. :) )
|
|
|
|
* main/term.c: Tweak the formatting of this new function to better
|
|
conform to coding guidelines.
|
|
|
|
2006-11-11 02:04 +0000 [r47490] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo
|
|
safe output!
|
|
|
|
2006-11-10 22:23 +0000 [r47480] Matt Frederickson <creslin@digium.com>
|
|
|
|
* channels/chan_zap.c: Make sure we don't use 32 bits when we only
|
|
need one bit.
|
|
|
|
2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: ...and make sure that the dialog is
|
|
destroyed, even if we don't get any answer on the bye... This is
|
|
the channel that remains dead after the SIP transfer
|
|
|
|
* channels/chan_sip.c: Add debug output while trying to trace bug
|
|
in bug report
|
|
|
|
* channels/chan_sip.c: Make sure we destroy dialog...
|
|
|
|
* /, channels/chan_sip.c: Small cleanup of handle_request_invite()
|
|
- imported from 1.2 with changes
|
|
|
|
2006-11-10 19:47 +0000 [r47462] Matt Frederickson <creslin@digium.com>
|
|
|
|
* channels/chan_zap.c: Fix for #7321. Be able to explicitly hide
|
|
callerid name for switches that bork on it.
|
|
|
|
2006-11-10 18:56 +0000 [r47454] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Issue 8010 - Fix support for multipart
|
|
SDP (alphaque)
|
|
|
|
2006-11-10 17:13 +0000 [r47444] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* build_tools/prep_moduledeps: grep -m is not available on BSD, so
|
|
use head -1 instead
|
|
|
|
2006-11-10 16:53 +0000 [r47437] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_chanspy.c: Only split up extension and context if a
|
|
value exists. (issue #8332 reported by loloski)
|
|
|
|
2006-11-10 16:51 +0000 [r47436] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c,
|
|
channels/chan_skinny.c, channels/chan_h323.c,
|
|
channels/chan_iax2.c: Discussion of these CLI changes resulted in
|
|
more consistency (Bug 8236)
|
|
|
|
2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* apps/app_queue.c: if adding a queue member is LOG_NOTICE, then
|
|
removing them should be LOG_NOTICE, not LOG_DEBUG
|
|
|
|
* apps/app_queue.c: reflect addition/removal of dynamic queue
|
|
members in queue_log, so that people using dialplan replacement
|
|
for AgentCallbackLogin can still track login/logout (issue #7736,
|
|
reported/patched by whoiswes but this commit was written by me
|
|
and covers all three paths for AQM/RQM)
|
|
|
|
2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Rip out half implementation of 491 response
|
|
support, since it wasn't implemented properly and caused memory
|
|
leaks in the case of us getting 491's, which Asterisk actually
|
|
sends... Since it is a bit too complicated to fix this, I'll rip
|
|
it out of 1.4 and put it on the to-do-list for future releases.
|
|
Now, we handle this as congestion, which it really is. Issue
|
|
#8331
|
|
|
|
* channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD.
|
|
Thanks fenlander!
|
|
|
|
2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_h323.c: Fix building of chan_h323 by completeing
|
|
some structure definitions. (issue #8327 reported by Mithraen)
|
|
|
|
* apps/app_voicemail.c: Do conversion in a more easier to read and
|
|
working way for \r, \n, and \t. (issue #8324 reported by
|
|
johnlange)
|
|
|
|
2006-11-09 21:26 +0000 [r47391] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_voicemail.c, channels/chan_zap.c,
|
|
build_tools/prep_moduledeps: Work around an issue that caused
|
|
menuselect to display a bogus description for app_voicemail and
|
|
chan_zap. These modules use some preprocessor directives to
|
|
determine what it will report to Asterisk as its description.
|
|
However, the way we extract this information from the source
|
|
files for menuselect is not smart enough to figure this out.
|
|
(issue #8326, #8328)
|
|
|
|
2006-11-09 16:53 +0000 [r47380] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_phone.c, /: Merged revisions 47379 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov
|
|
2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and
|
|
higher as, well, it's apparently going to be removed. This should
|
|
make all you FC6 fans happy as your Asterisk will now build
|
|
without any mods. ........
|
|
|
|
2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant <russell@digium.com>
|
|
|
|
* main/cli.c: fix tab completion for "core debug channel" and "core
|
|
no debug channel"
|
|
|
|
* main/cli.c: Fix "core show channel". Also, fix tab completion for
|
|
both "core show channel" and "core show channels".
|
|
|
|
* main/cli.c: Fix "core debug channel <whatever>". I guess someone
|
|
needs to go through and audit every CLI command that changed
|
|
number of arguments ...
|
|
|
|
* main/asterisk.c: revert the previous change, which actually
|
|
modified the deprecated command, "show profile". Now, actually
|
|
apply the change to "core show profile".
|
|
|
|
* main/asterisk.c: Fix argument parsing for the "core show profile"
|
|
CLI command (fixed by rizzo in his branch, team/rizzo/astobj2)
|
|
|
|
* main/cli.c: Fix another CLI command, "core show uptime" ...
|
|
(issue #8323, reported by johnlange, fixed by myself)
|
|
|
|
* main/asterisk.c: fix "core show version" to reflect the new
|
|
number of arguments for this CLI command (issue #8316, kshumard)
|
|
|
|
2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy <murf@digium.com>
|
|
|
|
* main/channel.c: This update fixes 7531
|
|
|
|
* channels/chan_skinny.c: Committed in behalf of 8190.
|
|
|
|
2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/frame.c: the battle over CLI command formats has broken
|
|
stuff...
|
|
|
|
* channels/chan_sip.c: add simple fix for SDP to report proper
|
|
sample rate for G.722 media sessions
|
|
|
|
2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant <russell@digium.com>
|
|
|
|
* utils/streamplayer.c: I occasionally get email from users that
|
|
are trying to figure out what this does, or due to some
|
|
misunderstanding as to what it is supposed to do, can't get it to
|
|
work. So, I have added some text here to hopefully explain what
|
|
this application does and does not do.
|
|
|
|
* channels/chan_gtalk.c: Make this module build again
|
|
|
|
* configure, configure.ac, acinclude.m4: Copy the macros from
|
|
libtool.m4 to our own acinclude.m4 such that libtool is no longer
|
|
required to be installed to be able to generated the configure
|
|
script.
|
|
|
|
2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo)
|
|
|
|
2006-11-07 23:46 +0000 [r47303] Steve Murphy <murf@digium.com>
|
|
|
|
* channels/chan_oss.c, main/channel.c, channels/chan_phone.c,
|
|
channels/chan_misdn.c, channels/chan_skinny.c,
|
|
channels/chan_features.c, channels/chan_h323.c,
|
|
channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c,
|
|
include/asterisk/stringfields.h, apps/app_voicemail.c,
|
|
main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c,
|
|
channels/chan_zap.c, channels/chan_sip.c, res/res_features.c,
|
|
channels/chan_agent.c, main/utils.c, include/asterisk/channel.h,
|
|
channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to
|
|
solve the problem in bug 7506. It's a lot of rework to solve a
|
|
fairly small problem... such is life.
|
|
|
|
2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_local.c: Make MOH work as it did before in
|
|
chan_local, without this then it can go funky when transfers and
|
|
MOH are involved. (issue #7671 reported by jmls)
|
|
|
|
2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* configs/musiconhold.conf.sample: clean up sample config, and make
|
|
native file playback the more obvious default choice
|
|
|
|
2006-11-07 18:38 +0000 [r47275] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* apps/app_voicemail.c: large overhaul to voicemail imap support.
|
|
Allows support for more imap servers, also a better
|
|
implementation of several parts of the original work. patch
|
|
provided by 8033 with major upgrades.
|
|
|
|
2006-11-07 17:30 +0000 [r47268] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of
|
|
continue.
|
|
|
|
2006-11-07 13:13 +0000 [r47250] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Fixing the attack shield so it doesn't
|
|
produce attacks... Issue 8265 - never reply to an ACK
|
|
|
|
2006-11-07 01:25 +0000 [r47239] Russell Bryant <russell@digium.com>
|
|
|
|
* /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06
|
|
Nov 2006) | 5 lines If random order is enabled for files mode
|
|
music on hold, set a random initial position, instead of always
|
|
starting at the first file, and doing the random operation only
|
|
when switching to the next file. (bug reported by John Lange on
|
|
the asterisk-dev mailing list) ........
|
|
|
|
2006-11-04 18:32 +0000 [r47199] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and
|
|
transfer from "john" Thank you!
|
|
|
|
2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant <russell@digium.com>
|
|
|
|
* main/cli.c: Fix another bug in "core set debug" ...
|
|
|
|
* main/asterisk.c, main/cli.c: Really fix the "core set debug" and
|
|
"core set verbose" CLI commands.
|
|
|
|
* main/cli.c: fix the "atleast" option to the "core set verbose"
|
|
and "core set debug" CLI commands
|
|
|
|
2006-11-03 23:17 +0000 [r47176] Steve Murphy <murf@digium.com>
|
|
|
|
* channels/chan_sip.c: This fix introduced via bug 8233
|
|
|
|
2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* bootstrap.sh: align bootstrap.sh with the version in trunk (needs
|
|
to be blocked as it is already in trunk)
|
|
|
|
* configure.ac: add proper environment vars to detect modules on
|
|
freebsd. (already applied to trunk so it needs to be blocked
|
|
there)
|
|
|
|
2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c,
|
|
channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More
|
|
changes making the CLI more consistent with "category verb
|
|
arguments" (continuation of issue 8236)
|
|
|
|
* main/config.c, main/cli.c, main/channel.c, main/manager.c,
|
|
channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c,
|
|
main/http.c, main/file.c, main/logger.c, main/image.c,
|
|
res/res_indications.c, main/asterisk.c, res/res_odbc.c,
|
|
channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
|
|
channels/chan_local.c, main/frame.c, channels/chan_sip.c,
|
|
res/res_features.c, channels/chan_agent.c, res/res_crypto.c,
|
|
res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c:
|
|
Reverse change of "show" to "list" and make several other
|
|
commands more consistent with "category verb arguments"
|
|
|
|
2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Move check for codec translation to
|
|
sip_call() instead of in add_sdp. No one bothers with the result
|
|
of add_sdp anyway... Yet...
|
|
|
|
* channels/chan_sip.c: Disable code for T38 over TCP and RTP since
|
|
there's no trace of actual functionality for it :-)
|
|
|
|
2006-11-02 17:49 +0000 [r46965] Russell Bryant <russell@digium.com>
|
|
|
|
* /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02
|
|
Nov 2006) | 3 lines ignore files in a music on hold directory
|
|
that begin with '.' (issue #8249, cboie) ........
|
|
|
|
2006-11-02 17:17 +0000 [r46963] Nadi Sarrar <ns@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix
|
|
|
|
2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_sip.c: don't send INVITE when we have determined
|
|
that we can't offer any audio formats due to lack of transcoding
|
|
support (or incorrect configuration)
|
|
|
|
2006-11-02 16:06 +0000 [r46930] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2
|
|
lines Repeat after me oej: I will at least make sure my code
|
|
compiles before I commit it. ........
|
|
|
|
2006-11-02 15:24 +0000 [r46901] Olle Johansson <oej@edvina.net>
|
|
|
|
* /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2)
|
|
|
|
2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant <russell@digium.com>
|
|
|
|
* /, main/callerid.c: Add the missing call to free described in
|
|
issue #8268. Also, add a bunch of missing calls to free in
|
|
callerid_feed_jp().
|
|
|
|
* main/say.c: fix saying one hundred and two hundred in hebrew
|
|
(issue #7810, eldadran)
|
|
|
|
* Makefile, configure, codecs/gsm/Makefile, configure.ac,
|
|
build_tools/strip_nonapi, makeopts.in: Fixes for
|
|
cross-compilation on mips (issue #8058, ywalther, with some
|
|
modifications)
|
|
|
|
* aclocal.m4, build_tools/menuselect-deps.in, configure,
|
|
build_tools/embed_modules.xml, configure.ac: Add a check in the
|
|
configure script to determine whether ld is GNU ld or not. This
|
|
is needed because module embedding only works for gnu ld. GNU ld
|
|
is now listed as a dependency for all of the module embedding
|
|
options in menuselect. (issue #8143)
|
|
|
|
2006-11-01 20:35 +0000 [r46822] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* channels/chan_gtalk.c: bind address support from bug 8164
|
|
|
|
2006-11-01 19:49 +0000 [r46802] Steve Murphy <murf@digium.com>
|
|
|
|
* res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
|
|
accept longer strings or mass confusion and a lot of lost time is
|
|
the result
|
|
|
|
2006-11-01 18:39 +0000 [r46780] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/Makefile: Force poll() emulation for Darwin to always be on.
|
|
It's too broken to consider being used. This resolves the console
|
|
issue OSX users have been seeing. I would have liked to autoconf
|
|
this but I haven't been able to come up with a test case that
|
|
works. Que sera.
|
|
|
|
2006-11-01 18:26 +0000 [r46778] Russell Bryant <russell@digium.com>
|
|
|
|
* res/res_monitor.c, /: Merged revisions 46776 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) |
|
|
9 lines soxmix and Asterisk expect different file extensions for
|
|
certain formats. This was already handled for the wav49 format.
|
|
However, it was not handled for ulaw and alaw. I fixed this in
|
|
such a way that using the alternate extensions for ulaw and alaw
|
|
will only happen if we know we're calling soxmix, and not a
|
|
custom script defined using the MONITOR_EXEC variable. The wav49
|
|
processing was left alone so that external scripts will see no
|
|
behavior change. (issue #7550, reported by mnicholson, proposed
|
|
patch by junky, committed fix is a bit different) ........
|
|
|
|
2006-11-01 18:21 +0000 [r46775] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: It's another round of chan_iax2 fixes!
|
|
Should hopefully fix the deadlock issues people have been
|
|
reporting. IAXtel now has qualify turned on for 800 peers and it
|
|
is handling it fine.
|
|
|
|
2006-11-01 17:48 +0000 [r46760] Steve Murphy <murf@digium.com>
|
|
|
|
* main/config.c: Cleanups suggested by Russell.
|
|
|
|
2006-11-01 16:39 +0000 [r46744] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_zap.c: Prevent an infinite loop when config
|
|
processing gets to a jitterbuffer option
|
|
|
|
2006-10-31 22:02 +0000 [r46716] Jason Parker <jparker@digium.com>
|
|
|
|
* main/translate.c: Fix "core show translation" output. Issue
|
|
#8243, patch by Damin.
|
|
|
|
2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* include/asterisk/translate.h, main/translate.c: add an API so
|
|
that translators can activate/deactivate themselves when needed
|
|
|
|
* include/asterisk/translate.h, main/translate.c: revert changes
|
|
that were the wrong way to address this... proper fix coming
|
|
|
|
* main/translate.c: let's set the seen flag early enough to
|
|
actually make a difference...
|
|
|
|
* include/asterisk/translate.h, main/translate.c: don't re-do setup
|
|
operations for translators that can dynamically register
|
|
themselves
|
|
|
|
2006-10-31 15:49 +0000 [r46663] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* /: Blocked revisions 46662 via svnmerge ........ r46662 |
|
|
tilghman | 2006-10-31 09:46:04 -0600 (Tue, 31 Oct 2006) | 3 lines
|
|
Move thread-unsafe initializer to the module loading code; add
|
|
the corresponding function to the module unload to fix a memory
|
|
leak. ........
|
|
|
|
2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson <oej@edvina.net>
|
|
|
|
* main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue
|
|
#8089 - Fix the ENUM support (picking one record by number).
|
|
Thanks otmar!
|
|
|
|
* /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport
|
|
when we're supposed to support ;rport. Issue #7473.
|
|
|
|
* /, channels/chan_sip.c: If peer fails ACL check, fail peer at
|
|
REGISTER
|
|
|
|
* channels/chan_sip.c: Fix T38 too. Thanks, tgrman !
|
|
|
|
2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant <russell@digium.com>
|
|
|
|
* contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the
|
|
boot process to ensure it starts after stuff like MySQL (issue
|
|
#8253, Alric)
|
|
|
|
* /, main/utils.c: Merged revisions 46560 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) |
|
|
3 lines When handling the case where the hostname is just an IPV4
|
|
numeric address, be sure to set the address type. (issue #8247,
|
|
alexr) ........
|
|
|
|
* /, res/res_agi.c: Merged revisions 46557 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) |
|
|
3 lines fix some copy/paste bugs in the checking of arguments for
|
|
the "control stream file" AGI command (issue #8255, mnicholson)
|
|
........
|
|
|
|
* main/translate.c: Add a small tweak to the code that checks to
|
|
see whether destination formats are translatable based on the
|
|
source format. If we have already determined that there is no
|
|
translation path in one direction, don't bother checking the
|
|
other direction.
|
|
|
|
2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/translate.c: when unregistering a translator, don't rebuild
|
|
the translation matrix unless needed when filtering formats out
|
|
of an offer, ensure we check for translation ability in both
|
|
directions
|
|
|
|
* include/asterisk/linkedlists.h: ensure that items removed from a
|
|
list are always unlinked from the list (next pointer set to NULL)
|
|
|
|
2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp <jcolp@digium.com>
|
|
|
|
* configure, configure.ac: Don't explicitly link in crypt as it is
|
|
not used on some platforms.
|
|
|
|
* channels/chan_iax2.c: We need to lock the pvt structure during
|
|
retransmission as another worker thread may be doing something as
|
|
well.
|
|
|
|
2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson <oej@edvina.net>
|
|
|
|
* main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h,
|
|
include/asterisk/doxyref.h, channels/chan_sip.c,
|
|
main/ast_expr2f.c, include/asterisk/module.h,
|
|
formats/format_ogg_vorbis.c, main/app.c,
|
|
include/asterisk/channel.h, include/asterisk/lock.h,
|
|
include/asterisk/frame.h: Issue #8246 - Doxygen fixes from
|
|
kshumard. An extra big thankyou is given to everyone that
|
|
contributes to doxygen! THANK YOU!
|
|
|
|
* main/rtp.c, /: Bind RTCP to the same IP as RTP
|
|
|
|
* /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302
|
|
redirects (imported from 1.2)
|
|
|
|
* /, channels/chan_sip.c: Issue #7608 - Notifications sent with
|
|
wrong content-type (imported from 1.2, modified)
|
|
|
|
* channels/chan_sip.c, CHANGES: Backport of patch for #7828 that
|
|
was reported for trunk, but obviously exists in 1.4 too.
|
|
|
|
* channels/chan_sip.c: Restoring the old logic, since working
|
|
around it and fixing it seemed too complicated. - The
|
|
SIP_OUTGOING flag indicates the direction of the last transaction
|
|
in the dialog. - The initreq stores the last request in the
|
|
dialog, the request that opened the latest transaction. Please
|
|
now retry all the 1.4 bug reports with mixed to/from headers,
|
|
tags etc in ACK, BYE, CANCEL. Thanks!
|
|
|
|
* channels/chan_sip.c: Accepting a message twice may be
|
|
misinterpreted...
|
|
|
|
* channels/chan_sip.c: - 183 is not reliable message... - Error
|
|
should not have SDP
|
|
|
|
2006-10-28 16:37 +0000 [r46377] Joshua Colp <jcolp@digium.com>
|
|
|
|
* utils/Makefile: Don't build muted on OpenBSD, it is not
|
|
supported.
|
|
|
|
2006-10-27 19:03 +0000 [r46370] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_zap.c: move the copy of the default settings to the
|
|
global settings back out of process_zap, so that they aren't
|
|
overwritten when process_zap is called multiple times
|
|
|
|
2006-10-27 18:29 +0000 [r46367] Olle Johansson <oej@edvina.net>
|
|
|
|
* contrib/asterisk-ng-doxygen: Put some doxygen pressure on
|
|
Christian :-)
|
|
|
|
2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant <russell@digium.com>
|
|
|
|
* main/asterisk.c, res/res_agi.c, apps/app_externalivr.c,
|
|
res/res_musiconhold.c: We should always be using _exit() after a
|
|
fork() or vfork() instead of exit(). This is because exit() does
|
|
some extra cleanup which in some implementations of vfork(), for
|
|
example, can actually modify the state of the parent process,
|
|
causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
|
|
|
|
* /: Blocked revisions 46361 via svnmerge ........ r46361 | russell
|
|
| 2006-10-27 12:36:07 -0500 (Fri, 27 Oct 2006) | 5 lines We
|
|
should always be using _exit() after a fork() or vfork() instead
|
|
of exit(). This is because exit() does some extra cleanup which
|
|
in some implementations of vfork(), for example, can actually
|
|
modify the state of the parent process, causing very weird bugs
|
|
or crashes. (issue #7971, Nick Gavrikov) ........
|
|
|
|
* channels/chan_zap.c: Instead of iterating all of the options once
|
|
to look for jitterbuffer options, and then again for everything
|
|
else, move the processing of jitterbuffer options into the main
|
|
loop so that there are no erroneous messages about ignoring
|
|
unknown options. (issue #8226)
|
|
|
|
2006-10-27 10:03 +0000 [r46351-46353] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
|
|
channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
|
|
Merged revisions 46350 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) |
|
|
1 line fixed a bug which caused chan_misdn to try to allocate 2
|
|
times the same channel on high load, which then caused
|
|
instability of mISDN. removed a useless function from isdn_lib.c
|
|
........
|
|
|
|
* channels/misdn_config.c: fixed not compile issue, which was just
|
|
introduced
|
|
|
|
* channels/misdn_config.c, channels/chan_misdn.c, /,
|
|
channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
|
|
Merged revisions 46176 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) |
|
|
1 line added nttimeout option to configure wether we disconnect
|
|
calls on NT timeouts or not during an overlapdial session
|
|
........
|
|
|
|
2006-10-26 17:57 +0000 [r46335-46340] Jason Parker <jparker@digium.com>
|
|
|
|
* /, contrib/scripts/astgenkey.8: Merged revisions 46337 via
|
|
svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2
|
|
lines oops - somebody forgot to change this - long ago, probably.
|
|
........
|
|
|
|
* CHANGES: grammar check
|
|
|
|
2006-10-26 16:38 +0000 [r46331] Olle Johansson <oej@edvina.net>
|
|
|
|
* CHANGES: Corrections to changes (Multiparking is not included)
|
|
|
|
2006-10-26 16:31 +0000 [r46329] Russell Bryant <russell@digium.com>
|
|
|
|
* main/translate.c: - If the source has no audio or no video
|
|
portion, do not call powerof() to get the format index. - Don't
|
|
run through the audio and video loops if there is no audio or
|
|
video portion of the source If 0 is passed to powerof, it will
|
|
return -1. This value of -1 was then being used as an array index
|
|
in these loops, which caused a crash on some systems. Other than
|
|
this issue, this code works as we expected it to. If a format is
|
|
not in the source, and we have to translation path to it, it is
|
|
not offered in the list of acceptable destination formats. (fixes
|
|
issue #8231)
|
|
|
|
2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* CHANGES: update to reflect G.722 addition
|
|
|
|
2006-10-26 04:18 +0000 [r46298] Russell Bryant <russell@digium.com>
|
|
|
|
* doc/backtrace.txt: update backtrace documentation to reflect
|
|
changes in 1.4 (issue #8230, kshumard)
|
|
|
|
2006-10-26 01:37 +0000 [r46287] Mark Spencer <markster@digium.com>
|
|
|
|
* main/config.c, main/manager.c: Fix config comment code
|
|
preservation code (thanks murf!)
|
|
|
|
2006-10-25 20:14 +0000 [r46276] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Old todo note - Don't add Contact header on
|
|
BYE and Cancel
|
|
|
|
2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant <russell@digium.com>
|
|
|
|
* configure.ac: fix error output when checking for openh323 to
|
|
refer to openh323 instead of pwlib (issue #8222, misaksen)
|
|
|
|
2006-10-25 19:16 +0000 [r46252] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Somewhat ugly code to try to fix issue
|
|
#7608. Since the problem was not very well defined, the fix is a
|
|
bit fuzzy too... Thanks to Luigi for accidentally spotting the
|
|
possible problem!
|
|
|
|
2006-10-25 19:08 +0000 [r46249] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_queue.c: update warning message to include "agi" option
|
|
(issue #8225, jmls)
|
|
|
|
2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* sounds/Makefile: use 1.4.3 extra sounds with corrected silence
|
|
files
|
|
|
|
* sounds/sounds.xml, sounds/Makefile: add support for prebuilt
|
|
G.722 prompts and music on hold files
|
|
|
|
2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: show settings doesn't produce a list of
|
|
similar objects, it should stay a "show"
|
|
|
|
2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c,
|
|
channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c,
|
|
pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c,
|
|
main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c,
|
|
cdr/cdr_custom.c, channels/chan_mgcp.c,
|
|
apps/app_parkandannounce.c, apps/app_voicemail.c,
|
|
channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c,
|
|
res/res_adsi.c, main/utils.c, apps/app_ices.c,
|
|
pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c,
|
|
apps/app_getcpeid.c: apparently developers are still not aware
|
|
that they should be use ast_copy_string instead of strncpy... fix
|
|
up many more users, and fix some bugs in the process
|
|
|
|
2006-10-25 04:58 +0000 [r46165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/pbx.c: WaitExten truncates decimals of times to wait,
|
|
instead of accepting them (Bug 8208)
|
|
|
|
2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c,
|
|
channels/chan_h323.c, channels/chan_iax2.c,
|
|
include/asterisk/frame.h: add passthrough and file format support
|
|
for G.722 16KHz audio (issue #5084, original patch by andrew,
|
|
updated by mithraen)
|
|
|
|
* channels/chan_sip.c, main/translate.c: code zone experiment:
|
|
don't offer formats in the outbound INVITE that aren't either
|
|
passthrough or translatable
|
|
|
|
* main/translate.c: if multiple translators are registered for the
|
|
same source/dest combination, ensure that the lowest-cost one is
|
|
always inserted earlier in the list
|
|
|
|
2006-10-24 20:30 +0000 [r46142] Mark Spencer <markster@digium.com>
|
|
|
|
* res/res_agi.c: Fix FastAGI when there is no pid (bug #7628,
|
|
#8147)
|
|
|
|
2006-10-24 19:29 +0000 [r46130] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: We need to initialize our scheduler pthread
|
|
condition... yes.
|
|
|
|
2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* main/http.c: merge 45152 don't leak descriptors in http.c
|
|
|
|
* channels/chan_sip.c: merge 45966 refer_to_domain potentially
|
|
containing options
|
|
|
|
* channels/chan_sip.c: merge 46026 improper checks on get_header()
|
|
return values
|
|
|
|
* channels/chan_sip.c: merge 46045 prevent NULL args to
|
|
ast_strdupa() in chan_sip.c
|
|
|
|
2006-10-24 05:23 +0000 [r46093] Russell Bryant <russell@digium.com>
|
|
|
|
* Makefile: Restore the ability to remove the firmware directory
|
|
without causing the installation to fail (issue #8111)
|
|
|
|
2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/translate.c: ensure that the translation matrix is properly
|
|
lock-protected every place it is used
|
|
|
|
* include/asterisk/translate.h, main/translate.c: add an API call
|
|
to allow channel drivers to determine which media formats are
|
|
compatible (passthrough or transcode) with the format an existing
|
|
channel is already using
|
|
|
|
* doc/imapstorage.txt: simplify and correct voicemail IMAP storage
|
|
build instructions
|
|
|
|
2006-10-24 03:01 +0000 [r46078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* main/channel.c: Pass through a frame if we don't know what it is,
|
|
rather than trying to pass a NULL, which will segfault a channel
|
|
driver (Bug 8149)
|
|
|
|
2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant <russell@digium.com>
|
|
|
|
* utils/muted.c, utils/ael_main.c: In muted.c, check the return
|
|
value of strdup. In ael_main.c, check the return value of calloc.
|
|
(issue #8157) In passing fix a few minor bugs in ael_main.c. The
|
|
last argument to strncpy() was a hard-coded 100, where it should
|
|
have been 99. I changed this to use sizeof() - 1.
|
|
|
|
* apps/app_meetme.c: Fix the descriptions of some of the
|
|
MeetMeAdmin options (issue #8098, mflorell)
|
|
|
|
* res/res_jabber.c: don't crash when an incoming message has no
|
|
"from" (issue #8205, jmls)
|
|
|
|
2006-10-23 00:27 +0000 [r45928] Joshua Colp <jcolp@digium.com>
|
|
|
|
* /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
|
|
r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2
|
|
lines Don't leak memory mmmk? ........
|
|
|
|
2006-10-22 21:44 +0000 [r45916] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21
|
|
Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and
|
|
couldn't be initialized it would cause a segfault after 'reload'.
|
|
Reported by Drew/Matt thx. ........
|
|
|
|
2006-10-21 18:49 +0000 [r45818] Russell Bryant <russell@digium.com>
|
|
|
|
* res/res_monitor.c: Add a couple missing unregistrations of
|
|
manager actions and remove duplicate unregistrations of
|
|
applications. (issue #8194, jmls)
|
|
|
|
2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/loader.c: Don't use promotion on Darwin because it doesn't
|
|
seem to work quite right in all cases, this should solve the
|
|
unresolved symbol issue people have been seeing.
|
|
|
|
* Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get
|
|
installed in the proper location (reported on asterisk-dev
|
|
mailing list)
|
|
|
|
2006-10-20 07:44 +0000 [r45741] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Let's understand SIP: - REFER can create
|
|
dialog, Asterisk does not support it yet - NOTIFY can create
|
|
dialog in Asterisk's implementation (voicemail) even though we
|
|
don't support the server side of it. In this case, the standard
|
|
is a side issue ;-) - Added extened functionality for unsupported
|
|
methods (PING, PUBLISH) so we don't create PVT's for those
|
|
either. Russellb needs to judge what to do with this in 1.2, but
|
|
I think the current implementation n 1.2 is a bug since we're
|
|
sending bad replies to NOTIFY and REFER outside of dialogs
|
|
|
|
2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp <jcolp@digium.com>
|
|
|
|
* res/res_jabber.c: Let's remember to unregister JabberStatus too
|
|
(issue #8184 reported by jmls)
|
|
|
|
* /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge
|
|
from https://origsvn.digium.com/svn/asterisk/branches/1.2
|
|
........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct
|
|
2006) | 2 lines Respect language selection when seeing if the
|
|
file exists (issue #8178 reported by mnicholson) ........
|
|
|
|
* channels/chan_sip.c: If the jitterbuffer is forced on then we
|
|
can't partially bridge (reported by wangster on #asterisk-dev)
|
|
|
|
2006-10-19 00:59 +0000 [r45622] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_sip.c: Don't leak the actual thread-specific
|
|
sip_pvt struct
|
|
|
|
2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_sip.c: don't leak memory when a chan_sip thread is
|
|
destroyed that has a thread-local temp_pvt allocated
|
|
|
|
2006-10-18 21:03 +0000 [r45595] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/asterisk.c: Don't modify things if we are using vfork as
|
|
this is very bad and may cause unexpected behavior (issue #7970
|
|
reported by Nick Gavrikov)
|
|
|
|
2006-10-18 11:54 +0000 [r45517] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: remove duplicate declarations
|
|
|
|
2006-10-18 04:09 +0000 [r45464] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* main/http.c: merge from trunk: move ast_variables_destroy() to a
|
|
better place in handle_uri() to avoid leaking memory on non
|
|
existing files.
|
|
|
|
2006-10-18 03:02 +0000 [r45452] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Don't segfault if you're using a channel driver that
|
|
doesn't turn RTCP on
|
|
|
|
2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant <russell@digium.com>
|
|
|
|
* main/channel.c: Don't attempt to access private data members of
|
|
the pthread_mutex_t object, because this does not work on all
|
|
linux systems. Instead, just access the reentrancy field in the
|
|
ast_mutex_info struct when DEBUG_THREADS is enabled. If
|
|
DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
|
|
DEBUG_THREADS on as well. (issue #8139, me)
|
|
|
|
* configs/sip_notify.conf.sample: update entry to reboot a snom
|
|
phone (issue #7850, pnlarsson)
|
|
|
|
2006-10-17 Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* Asterisk 1.4.0-beta3 released.
|
|
|
|
2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* include/asterisk/stringfields.h, main/ast_expr2.c,
|
|
main/channel.c, channels/chan_sip.c, channels/chan_iax2.c:
|
|
optimize the 'quick response' code a bit more... no more malloc()
|
|
or memset() for each response expand stringfields API a bit to
|
|
allow reusing the stringfield pool on a structure when needed,
|
|
and remove some unnecessary code when the structure was being
|
|
freed
|
|
|
|
2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Don't create a "real" pvt structure for
|
|
requests that shouldn't be able to create one. Instead use a
|
|
temporary pvt and fill it with enough information so we can send
|
|
a reply.
|
|
|
|
2006-10-17 17:39 +0000 [r45329] Olle Johansson <oej@edvina.net>
|
|
|
|
* configs/sip.conf.sample: Adding information about Marks
|
|
direct-RTP hack to the docs...
|
|
|
|
2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* LICENSE: provide licensing language for IAXy firmware file
|
|
|
|
2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_dial.c, apps/app_directed_pickup.c: Backport of new
|
|
directed pickup (BE-85).
|
|
|
|
2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson <oej@edvina.net>
|
|
|
|
* CREDITS: Adding Inotel to credits for SIP transfers. Thanks for
|
|
your support!
|
|
|
|
* channels/chan_sip.c: Don't destroy dialog for unexpected REFER
|
|
response...
|
|
|
|
2006-10-14 04:38 +0000 [r45143] Steve Murphy <murf@digium.com>
|
|
|
|
* funcs/func_rand.c: update the doc string for both AEL and
|
|
extensions.conf users.
|
|
|
|
2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/acl.c don't drop the entire permit/deny list when an attempt
|
|
is made to add an invalid entry (BE-92)
|
|
|
|
2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp <jcolp@digium.com>
|
|
|
|
* res/res_speech.c: Clear the quiet flag too since we are
|
|
restarting a recognition again (reported on -dev by Stephan
|
|
Edelman)
|
|
|
|
* res/res_speech.c: Check return value from engine in case of
|
|
failure (ie: out of licenses) (reported on -dev mailing list)
|
|
|
|
2006-10-13 20:52 +0000 [r45103] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/ael/ael-test/ref.ael-vtest17 (added),
|
|
pbx/ael/ael-test/ael-vtest17/extensions.ael (added),
|
|
pbx/ael/ael-test/ael-vtest17 (added),
|
|
pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in
|
|
this release via these changes
|
|
|
|
2006-10-13 19:19 +0000 [r45088] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/chan_misdn.c: avoiding warning, fixing potential bug
|
|
|
|
2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp <jcolp@digium.com>
|
|
|
|
* codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c,
|
|
codecs/lpc10/decode.c, codecs/lpc10/dcbias.c,
|
|
codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c,
|
|
codecs/lpc10/difmag.c, codecs/lpc10/hp100.c,
|
|
codecs/lpc10/synths.c, codecs/lpc10/preemp.c,
|
|
codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c,
|
|
codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c,
|
|
codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c,
|
|
codecs/lpc10/lpcini.c, codecs/lpc10/random.c,
|
|
codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c,
|
|
codecs/lpc10/placea.c, codecs/lpc10/tbdm.c,
|
|
codecs/lpc10/analys.c, codecs/lpc10/onset.c,
|
|
codecs/lpc10/energy.c, codecs/lpc10/deemp.c,
|
|
codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c,
|
|
codecs/lpc10/median.c, codecs/lpc10/encode.c,
|
|
codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c,
|
|
codecs/lpc10/invert.c: And file said... let the compiler warnings
|
|
STOP!
|
|
|
|
* apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136
|
|
reported by mnicholson)
|
|
|
|
* apps/app_playback.c: Move say.conf existence check to do_say
|
|
function since it is called from multiple places (issue #8144
|
|
reported by kshumard)
|
|
|
|
2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_iax2.c: when sending a call to a peer, use the proper socket if
|
|
we have multiple bindings (reported on asterisk-dev)
|
|
|
|
2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Complete merging in RPID screen changes
|
|
(issue #8101 reported by hristo, patch by oej in revision 44757)
|
|
|
|
* main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add
|
|
the background refresh item back into the scheduler if enabled
|
|
since it is deleted during reload. (issue #8142 reported by
|
|
p_lindheimer)
|
|
|
|
2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* configure, include/asterisk/autoconfig.h.in, configure.ac,
|
|
main/utils.c: use a configure script test for PMTU discovery
|
|
control instead of just assuming it's available on Linux
|
|
|
|
2006-10-13 14:45 +0000 [r44994-45026] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some
|
|
echocandisable issues when bridged. this caused a kernel panic
|
|
sometimes.. also some minor formatting fixes
|
|
|
|
* channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause
|
|
got a wrong isdn cause at RELEASE_COMPLETE
|
|
|
|
2006-10-12 22:07 +0000 [r44992] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* channels/chan_sip.c: merge formatting and minor code
|
|
simplifications from trunk
|
|
|
|
2006-10-12 20:34 +0000 [r44982] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* channels/chan_gtalk.c: fix for bug 7764.
|
|
|
|
2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_sip.c: we can only send one 'a=ptime' attribute per
|
|
media session, not one for each format
|
|
|
|
* main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c,
|
|
main/utils.c: ensure that IAX2 and SIP sockets allow UDP
|
|
fragmentation when running on Linux (thanks to Brian Candler on
|
|
the asterisk-dev list for the tip)
|
|
|
|
2006-10-12 16:56 +0000 [r44945] Russell Bryant <russell@digium.com>
|
|
|
|
* main/manager.c: fix a silly typo in a comment that I saw while
|
|
reading the commit list
|
|
|
|
2006-10-12 16:08 +0000 [r44942] Joshua Colp <jcolp@digium.com>
|
|
|
|
* Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue
|
|
#8135 reported by ssokol)
|
|
|
|
2006-10-12 12:55 +0000 [r44921] Nadi Sarrar <ns@beronet.com>
|
|
|
|
* main/manager.c: append_event must be called while holding the
|
|
session lock
|
|
|
|
2006-10-12 10:24 +0000 [r44911] Russell Bryant <russell@digium.com>
|
|
|
|
* res/res_jabber.c: change some debug output to use LOG_DEBUG
|
|
instead of verbose output
|
|
|
|
2006-10-11 16:57 +0000 [r44888] Jason Parker <jparker@digium.com>
|
|
|
|
* main/db1-ast/Makefile: These are already set by the parent
|
|
Makefile.. There is no need to have this here (it doesn't
|
|
actually work anyways).
|
|
|
|
2006-10-11 09:18 +0000 [r44854] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c: removed warning because of missing
|
|
prototype declaration
|
|
|
|
2006-10-10 19:23 +0000 [r44830] Olle Johansson <oej@edvina.net>
|
|
|
|
* channels/chan_sip.c: Do not set default/global values in the
|
|
variable declaration, set it in reload_config()
|
|
|
|
2006-10-10 17:21 +0000 [r44819] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Move some stuff around so that a NOTIFY
|
|
dialog won't hang around until the end of the world under certain
|
|
circumstances
|
|
|
|
2006-10-10 16:44 +0000 [r44809] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* main/channel.c, funcs/func_channel.c, include/asterisk/channel.h:
|
|
CHANNEL() function sometime mix parameter and value
|
|
|
|
2006-10-10 16:42 +0000 [r44808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* funcs/func_logic.c: Lost of a bit of logic when this was
|
|
simplified between 1.2 and 1.4 (Bug 8117)
|
|
|
|
2006-10-10 16:30 +0000 [r44806] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Bail out if we have no refer structure and
|
|
we get a refer response
|
|
|
|
2006-10-10 16:21 +0000 [r44805] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* channels/chan_sip.c: more merge from trunk (comments and change a
|
|
static function name)
|
|
|
|
2006-10-10 15:23 +0000 [r44788] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Only set DTMF information if an RTP
|
|
structure exists
|
|
|
|
2006-10-10 13:50 +0000 [r44786] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added
|
|
support of dynamically enabling hdlc on bchannels
|
|
|
|
2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* channels/chan_sip.c: whitespace changes related to previous
|
|
commit
|
|
|
|
* channels/chan_sip.c: merge a few code simplifications that have
|
|
gone into trunk during last week, to reduce differences between
|
|
the two branches and make porting fixes easier.
|
|
|
|
2006-10-09 16:12 +0000 [r44764] Jason Parker <jparker@digium.com>
|
|
|
|
* channels/chan_skinny.c: Fix a problem where phones that go
|
|
"missing" never got unregistered. Issue #8067, reported by pj,
|
|
patch by Anthony LaMantia (with minor whitespace modifications)
|
|
|
|
2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid
|
|
the deadlock
|
|
|
|
* channels/chan_iax2.c: Properly avoid a collision with iax2_hangup
|
|
(issue #8115 reported by vazir)
|
|
|
|
2006-10-08 14:14 +0000 [r44746] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* channels/chan_sip.c: do not dereference p if we
|
|
know it is NULL
|
|
|
|
2006-10-07 14:39 +0000 [r44684] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/ast_h323.cxx, channels/chan_h323.c,
|
|
channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate
|
|
caller's transfer capability too
|
|
|
|
2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* channels/chan_sip.c: put common code in a
|
|
function to avoid repetitions.
|
|
|
|
* channels/chan_sip.c: remove hardwired usage of 5060, use
|
|
DEFAULT_SIP_PORT instead
|
|
|
|
* channels/chan_sip.c: option_debug checking
|
|
before printing to debug channel.
|
|
|
|
* channels/chan_sip.c: backport simplifications on sip_register,
|
|
usage of ast_set2_flag(), and fixes to the handling of failed
|
|
module loading.
|
|
|
|
* channels/chan_sip.c: improve and document function
|
|
get_in_brackets(), introducing a helper function
|
|
find_closing_quote() of more general use.
|
|
|
|
2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* include/asterisk/linkedlists.h: ensure that mutex locks inside
|
|
list heads are initialized properly on platforms that require
|
|
constructor initialization (issue #8029, patch from timrobbins)
|
|
|
|
* CHANGES: remove Jingle as per mog
|
|
|
|
2006-10-06 21:08 +0000 [r44628] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Remove the seqno check for RFC2833, the handler is
|
|
smart enough to not need it.
|
|
|
|
2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* CHANGES: various cleanups
|
|
|
|
2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: When the sequence number rolls over then reset the
|
|
recorded sequence number for DTMF (issue #8106 reported by
|
|
bungalow)
|
|
|
|
* main/file.c: Even more frames to treat as though the remote side
|
|
disappeared (issue #8097 reported by eldadran)
|
|
|
|
2006-10-06 15:59 +0000 [r44567] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* main/manager.c, main/http.c: make sure sockets are blocking when
|
|
they should be blocking.
|
|
|
|
2006-10-06 12:53 +0000 [r44559-44563] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/chan_misdn.c: fixed segfault which happens during
|
|
hold/transfer action
|
|
|
|
* channels/chan_misdn.c: if INFORMATION Message come with keypad
|
|
instead of called party number, we just use the keypad as called
|
|
party number.
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
|
|
channels/misdn/isdn_lib.h, channels/chan_misdn.c,
|
|
channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
|
|
added the option 'reject_cause' to make it possible to set
|
|
the RELEASE_COMPLETE - cause on the 3. incoming PMP channel,
|
|
which is automatically rejected because chan_misdn does not
|
|
support that kind of callwaiting. Therefore chan_misdn supports
|
|
now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc
|
|
now gets the info if the requested channel is incoming or
|
|
outgoing to make the 3. channel possible
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
|
|
channels/chan_misdn.c: fixed the hold/retrieve/transfer issues,
|
|
removed a useless bc field, added setting of frame.delivery fields,
|
|
some minor code cleanups
|
|
|
|
2006-10-05 19:57 +0000 [r44502] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/file.c: Treat busy control frames as hangup in the file streaming
|
|
core (issue #8097 reported by eldadran)
|
|
|
|
2006-10-05 18:21 +0000 [r44488] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang.
|
|
Many thanks to Doug!
|
|
|
|
2006-10-05 18:01 +0000 [r44486] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite
|
|
hanging by a thread if the other side is already setup with T.38
|
|
|
|
2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/app.c: don't segfault when an argument without a close
|
|
parenthesis is found stop parsing as soon as that situation
|
|
occurs
|
|
|
|
2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy <murf@digium.com>
|
|
|
|
* CHANGES: I put the accumulated changes from the commit logs and
|
|
inspection, into CHANGES. Hope everyone approves!
|
|
|
|
* configs/muted.conf.sample, utils/muted.c: Hang on a minute, the
|
|
install process sticks muted.conf in /etc/asterisk, so that's
|
|
where muted should look for it, right?
|
|
|
|
2006-10-05 02:40 +0000 [r44450] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Don't totally bail out if T.38 was
|
|
negotiated
|
|
|
|
2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_sip.c: fix Polycom presence notification again
|
|
|
|
2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* utils/Makefile: as far as i can tell astman only uses newt...
|
|
|
|
* Makefile: put linker flags in ASTLDFLAGS where they belong
|
|
|
|
2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE
|
|
requests add workaround for new Polycom firmware SUBSCRIBE
|
|
requests (bug is known to exist in 2.0.1 firmware)
|
|
|
|
* include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually
|
|
work
|
|
|
|
2006-10-04 19:57 +0000 [r44380] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c,
|
|
pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
|
|
pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
|
|
pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
|
|
pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6,
|
|
pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8,
|
|
pbx/ael/ael-test/ael-test16/extensions.ael (added),
|
|
pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y,
|
|
pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
|
|
pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
|
|
pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the
|
|
problems reported in bug 8090
|
|
|
|
2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* channels/chan_oss.c, main/cdr.c, channels/chan_phone.c,
|
|
main/manager.c, pbx/pbx_spool.c, res/res_smdi.c,
|
|
channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
|
|
channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c,
|
|
main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c,
|
|
include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c,
|
|
channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c,
|
|
main/devicestate.c, main/utils.c, res/res_musiconhold.c,
|
|
channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update
|
|
thread creation code a bit reduce standard thread stack size
|
|
slightly to allow the pthreads library to allocate the stack+data
|
|
and not overflow a power-of-2 allocation in the kernel and waste
|
|
memory/address space add a new stack size for 'background'
|
|
threads (those that don't handle PBX calls) when LOW_MEMORY is
|
|
defined
|
|
|
|
2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy <murf@digium.com>
|
|
|
|
* configs/muted.conf.sample: I've been meaning to add some
|
|
explanation about muted... here it is
|
|
|
|
* configs/manager.conf.sample: CLI reverbification update to this
|
|
config file
|
|
|
|
* apps/app_macro.c: In response to bug 7776, a Warning has been
|
|
added to the doc string for Macro().
|
|
|
|
2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/asterisk.c, main/loader.c, main/term.c, Makefile,
|
|
include/asterisk.h: ensure that local include files are always
|
|
used avoid a duplicate function name (term_init())
|
|
|
|
2006-10-03 22:35 +0000 [r44312] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing
|
|
client without resource.
|
|
|
|
2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* apps/app_queue.c: fix a logic error in my previous fix to the queue
|
|
reload code
|
|
|
|
2006-10-03 18:42 +0000 [r44286] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/ast_h323.cxx: Change default presentation indicator
|
|
to "user provided not screened" if octet 3a missed in
|
|
CallingPartyNumber IE
|
|
|
|
2006-10-03 18:35 +0000 [r44284] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Use VideoSupport instead so it is considered
|
|
a valid XML attribute name. (issue #8075 reported by renemendoza)
|
|
|
|
2006-10-03 18:30 +0000 [r44283] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/ast_h323.cxx: Fix preparation of type and
|
|
presentation of calling number
|
|
|
|
2006-10-03 00:01 +0000 [r44240] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* doc/jingle.txt, channels/chan_jingle.c (removed),
|
|
include/asterisk/jabber.h, configs/jingle.conf.sample (removed),
|
|
res/res_jabber.c: updated res_jabber for even better component
|
|
support, soon will be jep-0100 compliant. also removed
|
|
chan_jingle and infromed info from jingle.txt, chan_gtalk still
|
|
works and should be used in this version.
|
|
|
|
2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Change the fd on the I/O context in case it
|
|
changed during the reload, which is indeed possible. (issue #7943
|
|
reported by eclubb)
|
|
|
|
* contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
|
|
instead of hardcoding the path for the error message (issue #7942
|
|
reported by eclubb)
|
|
|
|
2006-10-02 18:52 +0000 [r44186] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* configs/users.conf.sample, pbx/pbx_config.c: Missed part of
|
|
userconf functionality for chan_h323
|
|
|
|
2006-10-02 17:25 +0000 [r44169] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/io.c: Shrink when current_ioc is unused. It is set to -1 when
|
|
unused, not 0. (issue #7941 reported by eclubb)
|
|
|
|
2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* doc/realtime.txt: Typo fix
|
|
|
|
* channels/chan_h323.c: Optimization of oh323_indicate(): less
|
|
locks - less problems, plus single exit point
|
|
|
|
2006-10-02 02:38 +0000 [r44146] Mark Spencer <markster@digium.com>
|
|
|
|
* channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when
|
|
you're not talking about a channel :)
|
|
|
|
2006-10-01 19:32 +0000 [r44135] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/chan_h323.c: Do not simulate any audio tones if we got
|
|
PROGRESS message
|
|
|
|
2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant <russell@digium.com>
|
|
|
|
* Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to
|
|
be empty. The cause is that since ASTDATADIR is explicitly
|
|
exported using "export ASTDATADIR" at the top of the Makefile,
|
|
make no longer considers the variable "undefined", so the
|
|
Makefile can't use ?= to set ASTDATADIR if not yet set. (issue
|
|
#8063, reported by akohlsmith, fixed by me)
|
|
|
|
* configs/queues.conf.sample: Fix the name of the "eventmemberstatus"
|
|
option in the sample queues.conf (issue #8065, adamg)
|
|
|
|
2006-10-01 15:01 +0000 [r44109] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* channels/chan_sip.c: sync with trunk - move variable declarations
|
|
to the beginning of a block.
|
|
|
|
2006-09-30 19:20 +0000 [r44090] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* main/rtp.c: Allow one-way RTP streams (device->Asterisk)
|
|
|
|
2006-09-30 16:28 +0000 [r44080] Luigi Rizzo <rizzo@icir.org>
|
|
|
|
* codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent
|
|
build problems: - with AST_DEVMODE, building codecs/lpc10 fails
|
|
because of lots of warnings, and the configure step in editline
|
|
fails as well. Fix this by removing the -Werror in these steps. -
|
|
on FreeBSD (but probably on other platforms as well), the final
|
|
link of asterisk fails because AST_LIBS was not exported to the
|
|
subdirs Makefiles. Add a proper fix in the top-level Makefile (a
|
|
possible alternative way is to add "export AST_LIBS" near the
|
|
beginning of the file). With this fix, i believe that some of the
|
|
platform-specific conditionals in main/Makefile are redundant
|
|
(because they should be already dealt with in the top level
|
|
Makefile) but i don't have a platform to check. Merging to head
|
|
will happen in a moment.
|
|
|
|
2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment
|
|
of previous fix: Issue #7928 - Don't send both 404 and 503. Fix
|
|
by phsultan with a small fix by me, myself or I. Thanks,
|
|
Philippe! (This was caused by my changes to the transaction
|
|
handling)
|
|
|
|
* channels/chan_sip.c: Found some buggy SIP clients (phones Planet
|
|
VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
|
|
sends ACK not on OK message only (when remote party answers) but
|
|
on RINGING message too, so when we send 200 OK message, we get
|
|
unidentified ACK message (because INVITE acknowledged on RINGING
|
|
message already), so 200 OK retransmits within its retransmission
|
|
interval then call gets dropped. If someone else knows how to
|
|
provide workaround for such cases, please, fix it in correct way.
|
|
Thanks to ssh from #asteriskru for provide access to his box to
|
|
study and fix this case.
|
|
|
|
2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* agi, utils: ignore temporary files made by the Makefiles during a
|
|
build
|
|
|
|
* codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile,
|
|
codecs/Makefile, utils/Makefile, configure,
|
|
build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac,
|
|
Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile,
|
|
pbx/Makefile, res/Makefile, channels/Makefile: fix a few build
|
|
system bugs, and convert Makefiles to be compatible with GNU make
|
|
3.80
|
|
|
|
2006-09-29 22:35 +0000 [r44053] Jason Parker <jparker@digium.com>
|
|
|
|
* main/asterisk.c, main/cli.c: Fix a bug with the removal of
|
|
'atleast' argument to 'core verbose' and 'core debug'. Add that
|
|
argument back in.
|
|
|
|
2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more
|
|
carefully when no CallingNumber IE available
|
|
|
|
* channels/h323/ast_h323.cxx: Fake display name by called number on
|
|
incoming calls (until passing connected number/connected name is
|
|
not implemented)
|
|
|
|
* channels/h323/ast_h323.cxx: Ported code refers to H.450 - add
|
|
includes
|
|
|
|
* channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly
|
|
pass TON/PRESENTATION information - original
|
|
H323Connection::SendSignalSetup() destroys Q.931 fields.
|
|
|
|
2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/Makefile: yet another place where we were not using the
|
|
correct CFLAGS by default
|
|
|
|
* main/Makefile: missed one conversion to ASTCFLAGS
|
|
|
|
2006-09-29 18:30 +0000 [r44009] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/ast_h323.cxx, channels/chan_h323.c,
|
|
channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass
|
|
TON/PRESENTATION information too
|
|
|
|
2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile,
|
|
main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
|
|
Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse
|
|
CFLAGS and LDFLAGS for build of Asterisk components, because they
|
|
are also then used for non-Asterisk components (like menuselect);
|
|
use our own variables instead
|
|
|
|
* configure, configure.ac: support --without-curl in configure
|
|
script
|
|
|
|
* Makefile.rules: another cross-compile fix
|
|
|
|
* Makefile: a couple more environment settings that can't leak into
|
|
the menuselect build
|
|
|
|
* main/cli.c: proper fix for ast_group_t change
|
|
|
|
* include/asterisk/lock.h: eliminate compiler warning when
|
|
DEBUG_CHANNEL_LOCKS is enabled and users of this header file
|
|
don't also include channel.h
|
|
|
|
2006-09-28 20:11 +0000 [r43944] Jason Parker <jparker@digium.com>
|
|
|
|
* apps/app_queue.c: Fix incorrect argument order for member names,
|
|
on persisted members. Issue 8047, patch by jmls.
|
|
|
|
2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_playback.c, res/res_monitor.c,
|
|
include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c,
|
|
channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c,
|
|
main/udptl.c, main/frame.c, funcs/func_timeout.c,
|
|
channels/chan_sip.c, apps/app_festival.c,
|
|
channels/iax2-provision.c, apps/app_alarmreceiver.c,
|
|
res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c:
|
|
Put in missing \ns on the end of ast_logs (issue #7936 reported
|
|
by wojtekka)
|
|
|
|
2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* apps/app_queue.c: fix buggy (and overly complex) loop used during reload
|
|
of app_queue for static member list updating
|
|
|
|
2006-09-28 17:34 +0000 [r43918] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/ast_h323.cxx: Extend call establishment timeout
|
|
|
|
2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_iax2.c: Make sure the pvt exists before accessing
|
|
it again as it may have gone away (issue #7562 reported by Seb7
|
|
and issue #7939 reported by sorg)
|
|
|
|
* main/cli.c: Warning be gone!
|
|
|
|
2006-09-28 16:41 +0000 [r43899] BJ Weschke <bweschke@btwtech.com>
|
|
|
|
* apps/app_queue.c: app_queue is comparing the device names incorrectly
|
|
while checking their statuses. It's internal list of interfaces
|
|
includes the dial string, while the argument passed to this
|
|
function does not have the dial string (/n for a local channel).
|
|
This causes it to ignore the device state changes because it
|
|
thinks it belongs to none of its members. (#8040 reported and
|
|
patch by tim_ringenbach)
|
|
|
|
2006-09-28 16:17 +0000 [r43893] Joshua Colp <jcolp@digium.com>
|
|
|
|
* apps/app_meetme.c: Stop the stream after waitstream returns so that our
|
|
formats get restored. (issue #7370 reported by kryptolus)
|
|
|
|
2006-09-28 15:56 +0000 [r43877] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/ast_h323.cxx: Fix compiler warning
|
|
|
|
2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke <bweschke@btwtech.com>
|
|
|
|
* apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 -
|
|
tim_ringenbach reported and patched)
|
|
|
|
* apps/app_queue.c: Autopause not working for queue members. (#8042
|
|
- jmls reported and patch)
|
|
|
|
2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force
|
|
remote side to start media on outgoing PROGRESS message
|
|
|
|
* include/asterisk/compiler.h: Put attribute tag at correct place
|
|
|
|
2006-09-28 11:03 +0000 [r43852] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
|
|
channels/chan_misdn.c: fixed a bug which led to chan_list zombies,
|
|
when the call could not be properly established in misdn_call.
|
|
also removed the ACK_HDLC stuff which is not really needed.
|
|
|
|
2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/ast_h323.cxx: Do not open transmit channel until
|
|
TCS is received
|
|
|
|
* main/file.c: Don't warn on HOLD/UNHOLD control frames
|
|
|
|
* main/file.c: Don't treat unknown control frames as voice
|
|
|
|
2006-09-27 20:21 +0000 [r43816] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_voicemail.c: Avoid inability to lock directory log message by
|
|
creating the directory ahead of time. (Issue 7631)
|
|
|
|
2006-09-27 19:44 +0000 [r43801-43803] Jason Parker <jparker@digium.com>
|
|
|
|
* apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS
|
|
not being set under certain circumstances. Fix a minor issue, to
|
|
make it use the filenames that were parsed, instead of the entire
|
|
argument string. Fix Background() to return -1 like Playback(),
|
|
if no args are specified.
|
|
|
|
2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp <jcolp@digium.com>
|
|
|
|
* main/rtp.c: Compensate for out of order packets better if RFC2833
|
|
compensation is turned on.
|
|
|
|
* channels/chan_iax2.c: Get rid of two functions from a time now
|
|
past (we THINK these are from pre-recursive lock time) that may
|
|
be contributing to two open issues on the bug tracker (7562/7939)
|
|
and that has the potential to just make bad things happen if the
|
|
timing is right.
|
|
|
|
2006-09-27 16:55 +0000 [r43779] Russell Bryant <russell@digium.com>
|
|
|
|
* main/channel.c,res/res_features.c: Fix a problem that occurred if
|
|
a user entered a digit
|
|
that matched a bridge feature that was configured using multiple
|
|
digits, and the digit that was pressed timed out in the feature
|
|
digit timeout period. For example, if blind transfer is
|
|
configured as '##', and a user presses just '#'. In this
|
|
situation, the call would lock up and no longer pass any frames.
|
|
(issue #7977 reported by festr, and issue #7982 reported by
|
|
michaels and valuable input provided by mneuhauser and kuj. Fixed
|
|
by me, with testing help and peer review from Joshua Colp). There
|
|
are a couple of issues involved in this fix: 1) When
|
|
ast_generic_bridge determines that there has been a timeout, it
|
|
returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets
|
|
this result, it calls ast_generic_bridge over again with the same
|
|
timestamp for the next event. This results in an endless loop of
|
|
nothing until the call is terminated. This is resolved by simply
|
|
changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it
|
|
sees a timeout. 2) I also changed ast_channel_bridge such that if
|
|
in the process of calculating the time until the next event, it
|
|
knows a timeout has already occured, to immediately return
|
|
AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
|
|
anyway. 3) In the process of testing the previous two changes, I
|
|
ran into a problem in res_features where ast_channel_bridge would
|
|
return because it determined that there was a timeout. However,
|
|
ast_bridge_call in res_features would then determine by its own
|
|
calculation that there was still 1 ms before the timeout really
|
|
occurs. It would then proceed, and since the bridge broke out and
|
|
did *not* return a frame, it interpreted this as the call was
|
|
over and hung up the channels. The reason for this was because
|
|
ast_bridge_call in res_features and ast_channel_bridge in
|
|
channel.c were using different times for their calculations.
|
|
channel.c uses the start_time on the bridge config, which is the
|
|
time that the feature digit was recieved. However, res_features
|
|
had another time, 'start', which was set right before calling
|
|
ast_channel_bridge. 'start' will always be slightly after
|
|
start_time in the bridge config, and sometimes enough to round up
|
|
to one ms. This is fixed by making ast_bridge_call use the same
|
|
time as ast_channel_bridge for the timeout calculation. ........
|
|
|
|
2006-09-27 16:24 +0000 [r43775] Christian Richter <christian.richter@beronet.com>
|
|
|
|
* channels/chan_misdn.c, channels/Makefile: removed the chan_misdn
|
|
versioning, since Asterisk has it's own
|
|
|
|
2006-09-27 16:23 +0000 [r43774] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_sip.c: Make rfc2833compensate a global option.
|
|
|
|
2006-09-27 04:35 +0000 [r43756] Russell Bryant <russell@digium.com>
|
|
|
|
* apps/app_voicemail.c: Backport revision 43754 from the trunk,
|
|
which removes an unused buffer from mm_login to close bug 8038,
|
|
as well as addresses some formatting and coding guidelines issues
|
|
in passing. Originally, I did not commit this to 1.4 since it is
|
|
not necessarily fixing a bug. However, since the IMAP storage
|
|
code is brand new, I decided it would be better to make the
|
|
change here as well, in case someone has to work on this code to
|
|
address issues in the very near future. I don't want to make
|
|
unnecessary merge problems going to the trunk.
|
|
|
|
2006-09-27 02:32 +0000 [r43739] Steve Murphy <murf@digium.com>
|
|
|
|
* configs/extensions.ael.sample: This change to extensions.ael was
|
|
to fix bug 8031; the install scripts are causing it to be copied
|
|
to /etc/asterisk/extensions.ael, and because it is a fairly
|
|
direct conversion of the original extensions.conf, the macro and
|
|
context names clash with the existing extensions.conf. So, I put
|
|
an ael- in front of all macros and contexts, and checked every
|
|
goto and macro call. Also, this file compiles under aelparse.
|
|
|
|
2006-09-26 20:56 +0000 [r43710] Russell Bryant <russell@digium.com>
|
|
|
|
* main/asterisk.c: Back in revision 4798, this message was changed from
|
|
using ast_cli() to directly calling write(). During this change,
|
|
checking if this was a remote console was removed. This caused
|
|
this message about using "exit" or "quit" to exit an Asterisk
|
|
console to come up in times where it did not make sense. This
|
|
change restores the check to see if this is a remote console
|
|
before printing the message. (fixes BE-65)
|
|
|
|
2006-09-26 20:47 +0000 [r43707] Joshua Colp <jcolp@digium.com>
|
|
|
|
* .cleancount, main/cli.c, channels/chan_sip.c,
|
|
include/asterisk/channel.h: Use proper type to represent the group variable
|
|
(issue #8025 reported by makoto)
|
|
|
|
2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/chan_sip.c: Add missing newline character in the warning
|
|
message about deprecated TOS values in configuration.
|
|
|
|
* apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain
|
|
mailbox definitions, don't introduce a length limit on the
|
|
definition by using a 256 byte temporary storage buffer. Instead,
|
|
make the temporary buffer just as big as it needs to be to hold
|
|
the entire mailbox definition. (fixes BE-68)
|
|
|
|
2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_local.c: Strip options off the argument passed for
|
|
devicestate in chan_local. (issue #8034 reported by pcardozo)
|
|
|
|
* apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight
|
|
overhaul of the whisper support. 1. We need to duplicate the
|
|
frame from ast_translate 2. We need to ensure we always have
|
|
signed linear coming in for signed linear combining. 3. We need
|
|
to ensure we are always feeding signed linear out. 4. Properly
|
|
store and restore write format when beeping on the channel we are
|
|
whispering on. 5. Properly discontinue the stream on the channel
|
|
for the beep. (issue #8019 reported by timkelly1980)
|
|
|
|
2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* sounds/Makefile: update to use 1.4.3 core sounds, with corrected
|
|
beep/beeperr/tt-monkeys files
|
|
|
|
2006-09-26 18:08 +0000 [r43650-43674] Jason Parker <jparker@digium.com>
|
|
|
|
* doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by
|
|
Dan Austin. Maximum values were incorrect, which is why this is
|
|
being put in 1.4
|
|
|
|
* channels/chan_skinny.c: Add proper codec support to chan_skinny.
|
|
Works with at least ulaw, alaw, and g729a. This is technically a
|
|
"new feature", but there are justifications for it. I found a bug
|
|
with the recent rtp packetization changes, which caused the media
|
|
setup to fail under certain circumstances, particularly when
|
|
using allow=all, or having no allow= statements (globally or on
|
|
the device). I could have either removed the rtp packetization
|
|
features, or I could add proper codec support (which, without, I
|
|
think most people would consider to be a bug anyways).
|
|
|
|
2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_voicemail.c: Should have moved these lines up in the
|
|
merge, instead of removing them
|
|
|
|
* apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1)
|
|
delete=yes was ignored 2) maxmessages was ignored
|
|
|
|
2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h,
|
|
channels/h323/cisco-h225.asn: Fix ASN1 description of
|
|
non-standard Cisco extensions
|
|
|
|
* channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport
|
|
changes of trunk: 1) r43540: Avoid possible deadlock on channel
|
|
destruction 2) r43590: Disable fastStart if requested by remote
|
|
side
|
|
|
|
2006-09-25 15:23 +0000 [r43616] Jason Parker <jparker@digium.com>
|
|
|
|
* sounds/Makefile: One more fix for sounds installation - this time
|
|
for portability. Reported to asterisk-dev mailing list.
|
|
|
|
2006-09-25 14:52 +0000 [r43605] Steve Murphy <murf@digium.com>
|
|
|
|
* formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from
|
|
crashing if trying to play an OGG moh file.
|
|
|
|
2006-09-25 06:15 +0000 [r43582] Paul Cadach <paul@odt.east.telecom.kz>
|
|
|
|
* channels/h323/caps_h323.cxx, channels/h323/compat_h323.h,
|
|
channels/chan_h323.c: Merged revisions 43472,43495 from trunk
|
|
|
|
2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant <russell@digium.com>
|
|
|
|
* channels/iax2-provision.c: Fix a CLI command registration issue
|
|
where an erroneous message claiming that "iax2 show provisioning"
|
|
was already registered. This was because this command was
|
|
registering itself as both the command, as well as the command it
|
|
is deprecating. (issue #8022, reported by bjweeks, fixed by
|
|
myself)
|
|
|
|
* channels/chan_iax2.c:Check to see if the channel that is activating the
|
|
IAXPEER function is actually an IAX2 channel before proceeding to
|
|
process it to avoid crashing. (issue #8017, reported by admott,
|
|
fixed by myself)
|
|
|
|
2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* Makefile: don't output the 'build complete' message when the
|
|
target being run is already going to do an installation
|
|
|
|
2006-09-22 22:12 +0000 [r43518] Jason Parker <jparker@digium.com>
|
|
|
|
* channels/chan_skinny.c: Allow chan_skinny.so to be unloaded
|
|
properly. Remove reload support, since it doesn't
|
|
actually...work.
|
|
|
|
2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy <murf@digium.com>
|
|
|
|
* pbx/pbx_ael.c: This commits a change to return
|
|
MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all
|
|
goes well for bug 8004
|
|
|
|
* pbx/pbx_ael.c: If the extensions.ael file not found, or
|
|
unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004.
|
|
|
|
2006-09-22 17:25 +0000 [r43492] Jason Parker <jparker@digium.com>
|
|
|
|
* main/cli.c: Make sure we explicitly set the CLI command to not be
|
|
deprecated, if it isn't.
|
|
|
|
2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* sounds/Makefile: use rebuilt extra sounds
|
|
|
|
* main/channel.c: all the Linux systems I have don't use
|
|
'__m_count' for this field, so I don't know where this came
|
|
from...
|
|
|
|
2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant <russell@digium.com>
|
|
|
|
* include/asterisk/threadstorage.h: backport the compatability fix
|
|
to use attribute_malloc instaed of __attribute__ ((malloc))
|
|
|
|
* channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN
|
|
could not be configured (issue #8006, Mithraen)
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* main/frame.c: Suppress a compiler warning about the use of a
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potentially uninitialized variable. It couldn't actually happen,
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though.
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2006-09-22 03:01 +0000 [r43469] Jason Parker <jparker@digium.com>
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* channels/chan_skinny.c: First shot at unload_module in
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chan_skinny.. More to come.
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2006-09-21 23:50 +0000 [r43466] Matt O'Gorman <mogorman@digium.com>
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* include/asterisk/jabber.h, channels/chan_gtalk.c,
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res/res_jabber.c: updates for better compontent support
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2006-09-21 23:24 +0000 [r43464] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
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* res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we
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actually documented how the new features in res_odbc actually
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work. (Oops)
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2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp <jcolp@digium.com>
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* channels/chan_oss.c: Some more clean up in the load function for
|
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chan_oss (issue #8002 reported by Mithraen with minor mods by
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moi)
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* channels/chan_mgcp.c: Clean up chan_mgcp's module load function
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(issue #8001 reported by Mithraen with mods by moi)
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2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming <kpfleming@digium.com>
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* main/Makefile, build_tools/strip_nonapi (added): add another
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attempt to strip non-API symbols from the final binary... script
|
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will need to be extended to work on non-Linux systems
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2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
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* apps/app_url.c: Fix documentation to reflect how Url() really
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works
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* cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates
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2006-09-21 Kevin P. Fleming <kpfleming@digium.com>
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* Asterisk 1.4.0-beta2 released.
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2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming <kpfleming@digium.com>
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* main/Makefile: remove this change... it requires binutils 2.17
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2006-09-20 23:19 +0000 [r43396] Jason Parker <jparker@digium.com>
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* build_tools/make_version: fix minor typo in the way version is
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handled
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2006-09-20 Kevin P. Fleming <kpfleming@digium.com>
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* Asterisk 1.4.0-beta1 released.
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