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git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.0-beta4@48430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2674 lines
110 KiB
Plaintext
2674 lines
110 KiB
Plaintext
2006-12-12 Kevin P. Fleming <kpfleming@digium.com>
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* Asterisk 1.4.0-beta4 released.
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2006-12-12 04:13 +0000 [r48401] Joshua Colp <jcolp@digium.com>
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* apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This
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is the way it should have been done.
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2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman <mogorman@digium.com>
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* sounds/Makefile: new sounds package with 100% more silence
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* /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge
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from https://svn.digium.com/svn/asterisk/branches/1.2 ........
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r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
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| 4 lines app_externalivr needs a real silence file, and
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additional changes to add silence files into core instead of
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extra patch provided by bug 8177 with minor additions. ........
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2006-12-11 21:31 +0000 [r48391] Joshua Colp <jcolp@digium.com>
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* apps/app_voicemail.c: Return non-existant callerid handling to
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that which it was before. In 1.4 and trunk callerid can be
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allocated but not have any contents so we have to use
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ast_strlen_zero before passing it to the relevant functions.
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(issue #8567 reported by pabelanger)
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2006-12-11 05:37 +0000 [r48382] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
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* funcs/func_strings.c: STRFTIME() does not actually require an
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argument (issue 8540)
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2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp <jcolp@digium.com>
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* main/rtp.c: Merge in my latest RTP changes. Break out RTP and
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RTCP callback functions so they no longer share a common one.
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* apps/app_meetme.c: Use the correct API call to say a device state
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changed. (Yes, I'm a nub.)
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* apps/app_meetme.c: Don't access the conference structure after it
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has been freed.
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2006-12-11 00:47 +0000 [r48375] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
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* apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
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res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
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apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374
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via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006)
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| 5 lines When doing a fork() and exec(), two problems existed
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(Issue 8086): 1) Ignored signals stayed ignored after the exec().
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2) Signals could possibly fire between the fork() and exec(),
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causing Asterisk signal handlers within the child to execute,
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which caused nasty race conditions. ........
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2006-12-10 03:04 +0000 [r48372] Steve Murphy <murf@digium.com>
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* channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
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line This version applies the patch suggested by stevens in bug
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7836 (make inbound channel RINGING state consistent with other
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channels). ........
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2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant <russell@digium.com>
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* channels/chan_iax2.c: Use locking when accessing the
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registrations list. This list is not actually used very often, so
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the likelihood of there being a problem is pretty small, but
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still possible. For example, if the CLI command to list the
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registrations was called at the same time that a reload was
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occurring and the registrations list was getting destroyed and
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rebuilt, a crash could occur. In passing, go ahead and convert
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this list to use the linked list macros.
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* /: Blocked revisions 48361 via svnmerge ........ r48361 | russell
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| 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use
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locking when accessing the registrations list. This list is not
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actually used very often, so the likelihood of there being a
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problem is pretty small, but still possible. For example, if the
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CLI command to list the registrations was called at the same time
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that a reload was occurring and the registrations list was
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getting destroyed and rebuilt, a crash could occur. ........
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2006-12-07 18:17 +0000 [r48357] Russell Bryant <russell@digium.com>
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* /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/branches/1.2
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........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
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Dec 2006) | 3 lines Ensure that the file position is not
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incremented beyond the total number of files available for
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playback. (issue #8539, ulogic) ........
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2006-12-07 15:33 +0000 [r48349] Steve Murphy <murf@digium.com>
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* main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that
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killed bug 8423 -- OriginateSuccess and OriginateError incomplete
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channel name. May it rest in peace.
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2006-12-06 16:25 +0000 [r48326] Olle Johansson <oej@edvina.net>
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* /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being
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retransmitted to Asterisk
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2006-12-06 16:15 +0000 [r48323] Russell Bryant <russell@digium.com>
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* configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/branches/1.2
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........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
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Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
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in the sample configuration file. (issue #8526, arkadia) ........
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2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson <oej@edvina.net>
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* /, channels/chan_sip.c: Don't send Contact on MESSAGE
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2006-12-05 20:42 +0000 [r48279] Jason Parker <jparker@digium.com>
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* configure.ac: Fix curl version number testing to be much more
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friendly to non-bash shells. Issue 8508, patch by me. This
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*SHOULD* be POSIX compliant now..
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2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson <oej@edvina.net>
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* channels/chan_sip.c: Merging the invitestate-1.4 branch after
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successful testing. Will check if I can solve this with less
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changes in 1.2.
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* configs/sip.conf.sample: Add missing s from another repository.
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(thanks jcmoore!)
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* configs/sip.conf.sample: Updating sip.conf.sample with
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information about T38 not working when chan_local or chan_agent
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is involved in the call. I don't know how big a fix that would be
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to solve, but this is the current state of affairs. (Chan_sip
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currently checks if the other side of the bridge has a SIP tech.
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We could/should implement another check, possibly for udptl_write
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or some flag in the ast_channel structure).
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2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
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* apps/app_voicemail.c: Oops, forgot to release the odbc handle
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* apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006)
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| 6 lines If the recording in the database is too large, it will
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fail to retrieve with an mmap error. Not too sure why this
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doesn't happen when we put it in the database, also, but since
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that doesn't seem to be broken, I'm not going to fix it (at least
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until someone reports it). Solution is to ask for the file in
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smaller chunks. (Bug 8385) ........
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2006-12-04 21:48 +0000 [r48237-48248] Jason Parker <jparker@digium.com>
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* apps/app_voicemail.c: Fix an issue which didn't allow
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unavail/greet/busy/etc messages from being saved into ODBC (and
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probably IMAP).
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* /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell |
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2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert
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change from 8016 - this breaks other stuff... Needs further
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review. Tip: When you've reported a bug about something and
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somebody has put up a patch for it.. It's not a good idea to open
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a completely new bug and say that something is broken because of
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the patch in the other bug - PLEASE mention something in the bug
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where the patch was actually created. ........
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* /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell |
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2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an
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issue where a message isn't saved correctly when using ODBC
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storage and reviewing a message. Issue 8016 - patch by sokhapkin.
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........
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2006-12-04 18:16 +0000 [r48234] Joshua Colp <jcolp@digium.com>
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* /: Blocked revisions 48233 via svnmerge ........ r48233 | file |
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2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the
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generic bridge tells us not to retry, and we have a frame to spit
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out then break the bridge. Props to markit in #asterisk-bugs for
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bringing this up. ........
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2006-12-04 17:54 +0000 [r48228-48230] Jason Parker <jparker@digium.com>
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* configs/voicemail.conf.sample: Add documentation to
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voicemail.conf.sample for ODBC storage. Issue 8499 - patch by
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blitzrage.
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* doc/snmp.txt: Attempt to document some of the dependencies that
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are needed for net-snmp Issue 8499 - initial patch by blitzrage.
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2006-12-03 06:34 +0000 [r48223] Russell Bryant <russell@digium.com>
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* sounds/Makefile: When "fetch" is in use, instead of "wget",
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--continue is not a valid option. (issue #8451)
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2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson <oej@edvina.net>
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* channels/chan_sip.c: - Removing one of two pieces of code to
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handle 481 response on INVITE - Move handling of REFER response
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to handle_response_refer()
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* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
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configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax
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transmission happens - Encapsulate RTP timers in the rtp
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structure so we have one for video and one for audio The video
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one is not used in 1.4, really. Will be used for RTP keepalives
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when we can send something that video phones support in the RTP
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stream. I now this is a big architectual change at this stage for
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1.4, but decided it was needed to avoid future bug reports. -
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Document the RTP NAT keepalive option in sip.conf.sample Issue
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7679 in the bug tracker. Please test.
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2006-12-02 03:50 +0000 [r48195] Russell Bryant <russell@digium.com>
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* include/asterisk/utils.h: Backport the comment containing the
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warning regarding the limitations on the usage of this function.
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It is thread safe, but not technically reentrant.
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2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming <kpfleming@digium.com>
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* apps/app_dial.c, /: Merged revisions 48192 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006)
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| 2 lines if Dial() is going to send music-on-hold to the calling
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party, it has to send PROGRESS first to ensure that the reverse
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audio path has been setup first (BE-106) ........
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2006-12-01 23:16 +0000 [r48190] Russell Bryant <russell@digium.com>
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* Makefile, configure, configure.ac, makeopts.in, sounds/Makefile:
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FreeBSD 6.1 does not include wget by default. However, it has
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fetch which will work just fine for our purposes of downloading
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the sounds packages. So, check for both wget and fetch and the
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configure script and use what was found to download them. If
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neither one was found, and sound packages are selected that must
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be downloaded, the install process will print out an informative
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error message indicating the situation. Also, fix a couple places
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where "make" was hard coded into some output messages by
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replacing them with the $(MAKE) variable. (issue #8451, initial
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patch by pabelanger, with additional modifications by me)
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2006-12-01 20:25 +0000 [r48184-48186] Jason Parker <jparker@digium.com>
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* configs/extensions.conf.sample, /: Merged revisions 48183 via
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svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
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lines Fix a small typo - issue 8848, reported by pabelanger
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........
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2006-12-01 19:38 +0000 [r48179] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
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* main/cli.c: Double-unlock error (reported by blitzrage on IRC)
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2006-12-01 17:41 +0000 [r48177] Olle Johansson <oej@edvina.net>
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* channels/chan_sip.c, configs/sip.conf.sample: - Backport of the
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"limitonpeers" patch from trunk, to fix a lot of issues with
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queues and SIP device states - Remove support for T.38 early
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media, since it's impossible. (Two patches in one - extra friday
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evening offer due to being off line from svn today... :-)
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2006-11-30 21:18 +0000 [r48168] Joshua Colp <jcolp@digium.com>
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* main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not
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do a partial bridge for Google Talk since we need to handle STUN.
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(issue #8448 reported by phsultan)
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2006-11-30 20:51 +0000 [r48166] Olle Johansson <oej@edvina.net>
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* /, channels/chan_sip.c: Issue 8319 - change noncecount before
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using it.
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2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp <jcolp@digium.com>
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* /: Blocked revisions 48161 via svnmerge ........ r48161 | file |
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2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't
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write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel
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driver. (issue #8390 reported by hselasky) ........
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* /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
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lines Only print out debug message if bridged channel is not
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NULL. (issue #8412 reported by jubilex) ........
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* /, res/res_features.c: Merged revisions 48154 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
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lines Do not listen for DTMF on the bridge that comes into
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existence when ParkedCall is executed. This means native bridging
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can now occur for this. (issue #8406 reported by kebl0155)
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........
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* main/cdr.c, /: Merged revisions 48151 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
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lines Print certain CDR messages out at the NOTICE level versus
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WARNING since they can occur when used with the CDR applications
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and are perfectly fine. (issue #8367 reported by dartvader)
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........
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* /: Blocked revisions 48146 via svnmerge ........ r48146 | file |
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2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember
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the pointer to the allocated block of memory so that we can free
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it and not cause a memory leak. (issue #8449 reported by arkadia)
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........
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* /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/branches/1.2
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........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov
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2006) | 2 lines Document 'port' for SIP peers, came up because of
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the current mailing list thread. (issue #8450 reported by
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blitzrage) ........
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2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson <oej@edvina.net>
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* doc/manager.txt: Explain status reports and make codefreeze more
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happy :-)
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* /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by
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GS 487 adapter without CSEQ on separate line in the REGISTER
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request. Imported from 1.2.
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2006-11-29 21:05 +0000 [r48115] Joshua Colp <jcolp@digium.com>
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* apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in
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mm_login. (issue #8420 reported by slimey)
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2006-11-29 19:56 +0000 [r48113] Olle Johansson <oej@edvina.net>
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* configs/sip.conf.sample: Explain the use device status system
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implemented in SIP for subscriptions, queues and manager a bit
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better. Like in 1.2, you will get more detailed information if
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you set a call limit for a device. When the call limit is
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reached, the status system will report a device as busy. For
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queues, setting a call limit per SIP device is propably a
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requirement. In most cases, it will work much better if you only
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use type=peer and not type=friend. We might decide to backport
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the new setting from trunk to apply all call limits to the peer
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part of a friend only.
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2006-11-29 16:50 +0000 [r48107] Joshua Colp <jcolp@digium.com>
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* main/rtp.c, /: Merged revisions 48106 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
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lines If the frame was duplicated before writing out then we need
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to free it. (issue #8429 reported by edguy3) ........
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2006-11-29 08:03 +0000 [r48105] Olle Johansson <oej@edvina.net>
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* configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma.
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2006-11-29 04:26 +0000 [r48101] Joshua Colp <jcolp@digium.com>
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* apps/app_voicemail.c: Don't crash if the mailstream was not
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created.
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2006-11-28 18:26 +0000 [r48095] Jason Parker <jparker@digium.com>
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* Makefile: Export several more variables in top level Makefile.
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Inspired by issue 8438.
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2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp <jcolp@digium.com>
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* channels/chan_phone.c, /: Merged revisions 48087 via svnmerge
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from https://origsvn.digium.com/svn/asterisk/branches/1.2
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........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov
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2006) | 2 lines According to the research I have done we never
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needed to include compiler.h in the first place so let's not!
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(issue #8430 reported by edguy3) ........
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* apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
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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)
|
|
|
|
* main/frame.c: Suppress a compiler warning about the use of a
|
|
potentially uninitialized variable. It couldn't actually happen,
|
|
though.
|
|
|
|
2006-09-22 03:01 +0000 [r43469] Jason Parker <jparker@digium.com>
|
|
|
|
* channels/chan_skinny.c: First shot at unload_module in
|
|
chan_skinny.. More to come.
|
|
|
|
2006-09-21 23:50 +0000 [r43466] Matt O'Gorman <mogorman@digium.com>
|
|
|
|
* include/asterisk/jabber.h, channels/chan_gtalk.c,
|
|
res/res_jabber.c: updates for better compontent support
|
|
|
|
2006-09-21 23:24 +0000 [r43464] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we
|
|
actually documented how the new features in res_odbc actually
|
|
work. (Oops)
|
|
|
|
2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp <jcolp@digium.com>
|
|
|
|
* channels/chan_oss.c: Some more clean up in the load function for
|
|
chan_oss (issue #8002 reported by Mithraen with minor mods by
|
|
moi)
|
|
|
|
* channels/chan_mgcp.c: Clean up chan_mgcp's module load function
|
|
(issue #8001 reported by Mithraen with mods by moi)
|
|
|
|
2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/Makefile, build_tools/strip_nonapi (added): add another
|
|
attempt to strip non-API symbols from the final binary... script
|
|
will need to be extended to work on non-Linux systems
|
|
|
|
2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
|
|
|
|
* apps/app_url.c: Fix documentation to reflect how Url() really
|
|
works
|
|
|
|
* cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates
|
|
|
|
2006-09-21 Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* Asterisk 1.4.0-beta2 released.
|
|
|
|
2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* main/Makefile: remove this change... it requires binutils 2.17
|
|
|
|
2006-09-20 23:19 +0000 [r43396] Jason Parker <jparker@digium.com>
|
|
|
|
* build_tools/make_version: fix minor typo in the way version is
|
|
handled
|
|
|
|
2006-09-20 Kevin P. Fleming <kpfleming@digium.com>
|
|
|
|
* Asterisk 1.4.0-beta1 released.
|