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-rw-r--r--ChangeLog2821
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+1.4.0
diff --git a/ChangeLog b/ChangeLog
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+2006-12-23 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.4.0 released.
+
+2006-12-22 22:33 +0000 [r48870-48906] Jason Parker <jparker@digium.com>
+
+ * Makefile, main/stdtime/localtime.c: Minor fixes for Solaris.
+
+ * channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia
+
+2006-12-21 20:26 +0000 [r48783] Joshua Colp <jcolp@digium.com>
+
+ * /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2
+ lines Add new silence sound files to the spec for Redhat. (issue
+ #8652 reported by alvaro_palma_aste) ........
+
+2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: vms doesn't exist on non-IMAP storage
+ builds.
+
+ * apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so
+ it is then passed to the IMAP store file function. (issue #8614
+ reported by punknow)
+
+ * doc/snmp.txt: find is not the same as bind when it comes to
+ documentation. (issue #8626 reported by johann8384)
+
+2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/Makefile: suppress compiler warnings in this module
+ until it can be improved
+
+2006-12-19 21:12 +0000 [r48585] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 48584 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2
+ lines Free localuser structure when we fail to dial (issue #8612
+ reported by rizzo) ........
+
+2006-12-19 21:03 +0000 [r48583] Luigi Rizzo <rizzo@icir.org>
+
+ * apps/app_sms.c: fix a bogus datalen in the frames generated by
+ app_sms (causing noisy output if you listen to the output!) This
+ affects trunk as well, whereas 1.2 is ok.
+
+2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable
+ type for these unixODBC API calls, eliminating warnings on 64-bit
+ platforms that use the 'new' 64-bit types for ODBC API calls
+
+2006-12-19 03:46 +0000 [r48571] Joshua Colp <jcolp@digium.com>
+
+ * Makefile: Use env -i to start a fresh environment when going to
+ build menuselect. This is more portable then using unset. (issue
+ #8543 reported by jtodd)
+
+2006-12-18 17:23 +0000 [r48566] Luigi Rizzo <rizzo@icir.org>
+
+ * include/asterisk/channel.h: unbreak the macro used for
+ incrementing the frame counters. I don't know when the bug was
+ introduced, but with the typical usage c->fin =
+ FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects
+ trunk as well (fix coming).
+
+2006-12-18 17:15 +0000 [r48564] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Put thread into proper list if we abort
+ handling due to an error, and also hold the lock while putting it
+ back into the proper idle list so we don't prematurely get a
+ signal. (issue #8604 reported by arkadia)
+
+2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile,
+ utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile,
+ utils/ael_main.c: remove some now-unnecessary explicit includes
+ of autoconfig.h clean up per-file dependencies during 'make
+ clean'
+
+ * build_tools/prep_tarball: need an additional argument here to
+ make the downloads actually occur
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep
+ these calls from thinking they have multiple arguments
+
+ * codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile,
+ funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast,
+ main, codecs/gsm, pbx, res, channels, codecs, utils, agi,
+ main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr:
+ simplify dependency tracking system, using the compiler's
+ built-in method for generating them, and only doing dependency
+ tracking if developer mode is enabled via the configure script
+
+ * Makefile, include/asterisk.h, main/stdtime/localtime.c: since we
+ really, really have to have autoconfig.h included before all
+ other headers (especially system headers), the Makefile will now
+ force it to happen (this will fix build problems with files like
+ ast_expr2f.c, where we can't control the inclusion order in the
+ file itself)
+
+ * funcs/func_curl.c: instead of initializing the curl library every
+ time the CURL() function is invoked, do it only once per thread
+ (this allows multiple calls to CURL() in the dialplan for a
+ channel to run much more quickly, and also to re-use connections
+ to the server) (thanks to JerJer for frequently complaining about
+ this performance problem)
+
+2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Turn payload_lock into bridge_lock and make it
+ encompass all RTP structure contents that may relate to bridge
+ information, including who we are bridged to.
+
+ * channels/chan_iax2.c: Hold call structure lock in places where a
+ qualify or peer action can destroy it.
+
+ * channels/chan_iax2.c: Lock network retransmission queue in all
+ places that it is used.
+
+2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported
+ from 1.2)
+
+ * channels/chan_sip.c: Update to latest IANA spec
+
+2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Use a wakeup variable so that we don't wait
+ on IO indefinitely if packets need to be retransmitted.
+
+ * main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP
+ structure can change AFTER a bridge has started. This comes from
+ the packet handling of the SIP response when indication that it
+ was answered has been sent. Therefore we need to protect this
+ data with a lock when we read/write. (issue #8232 reported by
+ tgrman)
+
+ * main/rtp.c: Remove direct RTCP bridging. I've come to the
+ conclusion that we should handle this through the core and not
+ just forward it on. Should solve a few bugs.
+
+2006-12-12 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.4.0-beta4 released.
+
+2006-12-12 04:13 +0000 [r48401] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This
+ is the way it should have been done.
+
+2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman <mogorman@digium.com>
+
+ * sounds/Makefile: new sounds package with 100% more silence
+
+ * /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge
+ from https://svn.digium.com/svn/asterisk/branches/1.2 ........
+ r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
+ | 4 lines app_externalivr needs a real silence file, and
+ additional changes to add silence files into core instead of
+ extra patch provided by bug 8177 with minor additions. ........
+
+2006-12-11 21:31 +0000 [r48391] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Return non-existant callerid handling to
+ that which it was before. In 1.4 and trunk callerid can be
+ allocated but not have any contents so we have to use
+ ast_strlen_zero before passing it to the relevant functions.
+ (issue #8567 reported by pabelanger)
+
+2006-12-11 05:37 +0000 [r48382] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * funcs/func_strings.c: STRFTIME() does not actually require an
+ argument (issue 8540)
+
+2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Merge in my latest RTP changes. Break out RTP and
+ RTCP callback functions so they no longer share a common one.
+
+ * apps/app_meetme.c: Use the correct API call to say a device state
+ changed. (Yes, I'm a nub.)
+
+ * apps/app_meetme.c: Don't access the conference structure after it
+ has been freed.
+
+2006-12-11 00:47 +0000 [r48375] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
+ res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
+ apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006)
+ | 5 lines When doing a fork() and exec(), two problems existed
+ (Issue 8086): 1) Ignored signals stayed ignored after the exec().
+ 2) Signals could possibly fire between the fork() and exec(),
+ causing Asterisk signal handlers within the child to execute,
+ which caused nasty race conditions. ........
+
+2006-12-10 03:04 +0000 [r48372] Steve Murphy <murf@digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
+ line This version applies the patch suggested by stevens in bug
+ 7836 (make inbound channel RINGING state consistent with other
+ channels). ........
+
+2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Use locking when accessing the
+ registrations list. This list is not actually used very often, so
+ the likelihood of there being a problem is pretty small, but
+ still possible. For example, if the CLI command to list the
+ registrations was called at the same time that a reload was
+ occurring and the registrations list was getting destroyed and
+ rebuilt, a crash could occur. In passing, go ahead and convert
+ this list to use the linked list macros.
+
+ * /: Blocked revisions 48361 via svnmerge ........ r48361 | russell
+ | 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use
+ locking when accessing the registrations list. This list is not
+ actually used very often, so the likelihood of there being a
+ problem is pretty small, but still possible. For example, if the
+ CLI command to list the registrations was called at the same time
+ that a reload was occurring and the registrations list was
+ getting destroyed and rebuilt, a crash could occur. ........
+
+2006-12-07 18:17 +0000 [r48357] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
+ Dec 2006) | 3 lines Ensure that the file position is not
+ incremented beyond the total number of files available for
+ playback. (issue #8539, ulogic) ........
+
+2006-12-07 15:33 +0000 [r48349] Steve Murphy <murf@digium.com>
+
+ * main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that
+ killed bug 8423 -- OriginateSuccess and OriginateError incomplete
+ channel name. May it rest in peace.
+
+2006-12-06 16:25 +0000 [r48326] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being
+ retransmitted to Asterisk
+
+2006-12-06 16:15 +0000 [r48323] Russell Bryant <russell@digium.com>
+
+ * configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
+ Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
+ in the sample configuration file. (issue #8526, arkadia) ........
+
+2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Don't send Contact on MESSAGE
+
+2006-12-05 20:42 +0000 [r48279] Jason Parker <jparker@digium.com>
+
+ * configure.ac: Fix curl version number testing to be much more
+ friendly to non-bash shells. Issue 8508, patch by me. This
+ *SHOULD* be POSIX compliant now..
+
+2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Merging the invitestate-1.4 branch after
+ successful testing. Will check if I can solve this with less
+ changes in 1.2.
+
+ * configs/sip.conf.sample: Add missing s from another repository.
+ (thanks jcmoore!)
+
+ * configs/sip.conf.sample: Updating sip.conf.sample with
+ information about T38 not working when chan_local or chan_agent
+ is involved in the call. I don't know how big a fix that would be
+ to solve, but this is the current state of affairs. (Chan_sip
+ currently checks if the other side of the bridge has a SIP tech.
+ We could/should implement another check, possibly for udptl_write
+ or some flag in the ast_channel structure).
+
+2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Oops, forgot to release the odbc handle
+
+ * apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006)
+ | 6 lines If the recording in the database is too large, it will
+ fail to retrieve with an mmap error. Not too sure why this
+ doesn't happen when we put it in the database, also, but since
+ that doesn't seem to be broken, I'm not going to fix it (at least
+ until someone reports it). Solution is to ask for the file in
+ smaller chunks. (Bug 8385) ........
+
+2006-12-04 21:48 +0000 [r48237-48248] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Fix an issue which didn't allow
+ unavail/greet/busy/etc messages from being saved into ODBC (and
+ probably IMAP).
+
+ * /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell |
+ 2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert
+ change from 8016 - this breaks other stuff... Needs further
+ review. Tip: When you've reported a bug about something and
+ somebody has put up a patch for it.. It's not a good idea to open
+ a completely new bug and say that something is broken because of
+ the patch in the other bug - PLEASE mention something in the bug
+ where the patch was actually created. ........
+
+ * /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell |
+ 2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an
+ issue where a message isn't saved correctly when using ODBC
+ storage and reviewing a message. Issue 8016 - patch by sokhapkin.
+ ........
+
+2006-12-04 18:16 +0000 [r48234] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 48233 via svnmerge ........ r48233 | file |
+ 2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the
+ generic bridge tells us not to retry, and we have a frame to spit
+ out then break the bridge. Props to markit in #asterisk-bugs for
+ bringing this up. ........
+
+2006-12-04 17:54 +0000 [r48228-48230] Jason Parker <jparker@digium.com>
+
+ * configs/voicemail.conf.sample: Add documentation to
+ voicemail.conf.sample for ODBC storage. Issue 8499 - patch by
+ blitzrage.
+
+ * doc/snmp.txt: Attempt to document some of the dependencies that
+ are needed for net-snmp Issue 8499 - initial patch by blitzrage.
+
+2006-12-03 06:34 +0000 [r48223] Russell Bryant <russell@digium.com>
+
+ * sounds/Makefile: When "fetch" is in use, instead of "wget",
+ --continue is not a valid option. (issue #8451)
+
+2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: - Removing one of two pieces of code to
+ handle 481 response on INVITE - Move handling of REFER response
+ to handle_response_refer()
+
+ * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
+ configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax
+ transmission happens - Encapsulate RTP timers in the rtp
+ structure so we have one for video and one for audio The video
+ one is not used in 1.4, really. Will be used for RTP keepalives
+ when we can send something that video phones support in the RTP
+ stream. I now this is a big architectual change at this stage for
+ 1.4, but decided it was needed to avoid future bug reports. -
+ Document the RTP NAT keepalive option in sip.conf.sample Issue
+ 7679 in the bug tracker. Please test.
+
+2006-12-02 03:50 +0000 [r48195] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/utils.h: Backport the comment containing the
+ warning regarding the limitations on the usage of this function.
+ It is thread safe, but not technically reentrant.
+
+2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 48192 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006)
+ | 2 lines if Dial() is going to send music-on-hold to the calling
+ party, it has to send PROGRESS first to ensure that the reverse
+ audio path has been setup first (BE-106) ........
+
+2006-12-01 23:16 +0000 [r48190] Russell Bryant <russell@digium.com>
+
+ * Makefile, configure, configure.ac, makeopts.in, sounds/Makefile:
+ FreeBSD 6.1 does not include wget by default. However, it has
+ fetch which will work just fine for our purposes of downloading
+ the sounds packages. So, check for both wget and fetch and the
+ configure script and use what was found to download them. If
+ neither one was found, and sound packages are selected that must
+ be downloaded, the install process will print out an informative
+ error message indicating the situation. Also, fix a couple places
+ where "make" was hard coded into some output messages by
+ replacing them with the $(MAKE) variable. (issue #8451, initial
+ patch by pabelanger, with additional modifications by me)
+
+2006-12-01 20:25 +0000 [r48184-48186] Jason Parker <jparker@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 48183 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
+ lines Fix a small typo - issue 8848, reported by pabelanger
+ ........
+
+2006-12-01 19:38 +0000 [r48179] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * main/cli.c: Double-unlock error (reported by blitzrage on IRC)
+
+2006-12-01 17:41 +0000 [r48177] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: - Backport of the
+ "limitonpeers" patch from trunk, to fix a lot of issues with
+ queues and SIP device states - Remove support for T.38 early
+ media, since it's impossible. (Two patches in one - extra friday
+ evening offer due to being off line from svn today... :-)
+
+2006-11-30 21:18 +0000 [r48168] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not
+ do a partial bridge for Google Talk since we need to handle STUN.
+ (issue #8448 reported by phsultan)
+
+2006-11-30 20:51 +0000 [r48166] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Issue 8319 - change noncecount before
+ using it.
+
+2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp <jcolp@digium.com>
+
+ * /: Blocked revisions 48161 via svnmerge ........ r48161 | file |
+ 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't
+ write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel
+ driver. (issue #8390 reported by hselasky) ........
+
+ * /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
+ lines Only print out debug message if bridged channel is not
+ NULL. (issue #8412 reported by jubilex) ........
+
+ * /, res/res_features.c: Merged revisions 48154 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
+ lines Do not listen for DTMF on the bridge that comes into
+ existence when ParkedCall is executed. This means native bridging
+ can now occur for this. (issue #8406 reported by kebl0155)
+ ........
+
+ * main/cdr.c, /: Merged revisions 48151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
+ lines Print certain CDR messages out at the NOTICE level versus
+ WARNING since they can occur when used with the CDR applications
+ and are perfectly fine. (issue #8367 reported by dartvader)
+ ........
+
+ * /: Blocked revisions 48146 via svnmerge ........ r48146 | file |
+ 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember
+ the pointer to the allocated block of memory so that we can free
+ it and not cause a memory leak. (issue #8449 reported by arkadia)
+ ........
+
+ * /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov
+ 2006) | 2 lines Document 'port' for SIP peers, came up because of
+ the current mailing list thread. (issue #8450 reported by
+ blitzrage) ........
+
+2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson <oej@edvina.net>
+
+ * doc/manager.txt: Explain status reports and make codefreeze more
+ happy :-)
+
+ * /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by
+ GS 487 adapter without CSEQ on separate line in the REGISTER
+ request. Imported from 1.2.
+
+2006-11-29 21:05 +0000 [r48115] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in
+ mm_login. (issue #8420 reported by slimey)
+
+2006-11-29 19:56 +0000 [r48113] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Explain the use device status system
+ implemented in SIP for subscriptions, queues and manager a bit
+ better. Like in 1.2, you will get more detailed information if
+ you set a call limit for a device. When the call limit is
+ reached, the status system will report a device as busy. For
+ queues, setting a call limit per SIP device is propably a
+ requirement. In most cases, it will work much better if you only
+ use type=peer and not type=friend. We might decide to backport
+ the new setting from trunk to apply all call limits to the peer
+ part of a friend only.
+
+2006-11-29 16:50 +0000 [r48107] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, /: Merged revisions 48106 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
+ lines If the frame was duplicated before writing out then we need
+ to free it. (issue #8429 reported by edguy3) ........
+
+2006-11-29 08:03 +0000 [r48105] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma.
+
+2006-11-29 04:26 +0000 [r48101] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Don't crash if the mailstream was not
+ created.
+
+2006-11-28 18:26 +0000 [r48095] Jason Parker <jparker@digium.com>
+
+ * Makefile: Export several more variables in top level Makefile.
+ Inspired by issue 8438.
+
+2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_phone.c, /: Merged revisions 48087 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov
+ 2006) | 2 lines According to the research I have done we never
+ needed to include compiler.h in the first place so let's not!
+ (issue #8430 reported by edguy3) ........
+
+ * apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2
+ lines Use the proper function to get the new message count
+ instead of always using the filesystem. (issue #8421 reported by
+ slimey) ........
+
+2006-11-27 17:20 +0000 [r48049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27
+ Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381)
+ ........
+
+2006-11-27 17:17 +0000 [r48046] Russell Bryant <russell@digium.com>
+
+ * main/manager.c: Remove a couple of unused variables (issue #8380,
+ casper)
+
+2006-11-27 15:32 +0000 [r48038] Joshua Colp <jcolp@digium.com>
+
+ * pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2
+ lines Do not reference the freed outgoing structure in the debug
+ message. (issue #8425 reported by arkadia) ........
+
+2006-11-27 06:41 +0000 [r48031] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Change logging message
+
+2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy <murf@digium.com>
+
+ * funcs/func_cdr.c: might as well also document the raw values of
+ the flag vars
+
+ * /, funcs/func_cdr.c: A little bit of func_cdr documentation
+ upgrade-- no bug# involved, although 8221 may have inspired it.
+
+2006-11-25 09:28 +0000 [r48002] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4
+ and future releases, you can disable subscription support totally
+ or per peer in sip.conf with allowsubscribe = yes | no
+
+2006-11-24 17:17 +0000 [r47992] Steve Murphy <murf@digium.com>
+
+ * main/translate.c: bug 8189 posted this fix for main/translate.c
+ for PLC
+
+2006-11-24 15:46 +0000 [r47989] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn_config.c,
+ channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23
+ Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE.
+ beatufied some logs, changed some loglevels. changed the default
+ value of block_on_alarm ........
+
+2006-11-23 11:01 +0000 [r47959] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Don't allocate unused variable.
+
+2006-11-22 21:47 +0000 [r47944] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Video will never reach Packet2Packet bridging and can
+ do more harm then good.
+
+2006-11-21 17:32 +0000 [r47897] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: If we have the non standard G726-32 setting turned on
+ we want to return G726-32 to the SDP, not our AAL2 string. (issue
+ #8330 reported by voipgate)
+
+2006-11-21 15:20 +0000 [r47892] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Apparently Exosip sends a 101 after a 100
+ provisional response. Let's not treat that as early media.
+ (discovered at the AVTF meeting in Paris).
+
+2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Oops, merge missed release of odbc object
+
+ * apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006)
+ | 2 lines Failing to trap -1 error from mmap causes segfault
+ (Issue 8385) ........
+
+2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp <jcolp@digium.com>
+
+ * main/frame.c, /: Merged revisions 47859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2
+ lines Don't forget to byte swap if we are exiting the smoother
+ feed early. (issue #8287 reported by arturs) ........
+
+ * /: Blocked revisions 47855 via svnmerge ........ r47855 | file |
+ 2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines Free
+ history items at the end of use of the temporary SIP pvt
+ structure. (issue #8383 reported by benh) ........
+
+ * main/rtp.c: Only remove/destroy the RTCP I/O item if it exists.
+
+ * .cleancount, apps/app_dial.c, apps/app_directed_pickup.c,
+ include/asterisk/channel.h: Use a separate variable in the
+ channel structure to store the context that the channel was
+ dialed from. (issue #8382 reported by jiddings)
+
+2006-11-20 11:45 +0000 [r47843-47845] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Explain properly how videosupport works.
+ Committ from Asterisk Video Task Force meeting in Paris!
+
+ * /, channels/chan_sip.c: Make sure we destroy scheduled items and
+ not use them ever again after destruction (rizzo)
+
+2006-11-18 17:59 +0000 [r47823] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_sip.c: fix bug 7450 - Parsing fails if From header
+ contains angle brackets (the bug was only in a corner case where
+ the < was right after the opening quote, and the fix is trivial).
+
+2006-11-16 23:19 +0000 [r47781-47782] Jason Parker <jparker@digium.com>
+
+ * apps/app_db.c, apps/app_dial.c: Fix a couple of typos. Initially
+ pointed out by mrobinson.
+
+ * /: Blocked revisions 47780 via svnmerge ........ r47780 | qwell |
+ 2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a
+ couple of typos in applications.. Initially spotted by mrobinson.
+ ........
+
+2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, doc/billing.txt: update documentation regarding IAX2 transfers
+ and CDRs Merged revisions 47776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006)
+ | 2 lines update clearly wrong documentation regarding cdr_custom
+ ........
+
+2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Compare technology using the pointers
+ instead of a straight comparison based on name. (issue #8228
+ reported by dean bath)
+
+ * /: Blocked revisions 47761 via svnmerge ........ r47761 | file |
+ 2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines Look for
+ the header file specifically in all cases, not just the existence
+ of the directory. (issue #8358 reported by mrness) ........
+
+2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, configure.ac: check for pre-1.4 versions of Zaptel and
+ abort the configure script if found with an appropriate error
+ message
+
+2006-11-16 19:24 +0000 [r47755] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD
+ notification optional, in order to avoid a lot of extra database
+ lookups for all those realtime users out there.
+
+2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 47750 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov
+ 2006) | 2 lines Because of the way chan_local is written we
+ should be extra careful and make sure our callback functions have
+ a tech_pvt. (issue #8275 reported by mflorell) ........
+
+ * apps/app_meetme.c: Don't unreference the SLA object if there is
+ no SLA object in the devicestate callback. (issue #8354 reported
+ by loloski)
+
+2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Don't fixup if there's nothing to fixup
+
+ * UPGRADE.txt: Warn users about change in canreinvite
+
+ * channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never
+ authenticated (according to the RFC) - Update docs on
+ canreinvite. "nonat" is the recommended setting for most users
+ with phones behind a NAT.
+
+2006-11-15 22:31 +0000 [r47712] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 47711 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov
+ 2006) | 2 lines Make sure that the pvt structure exists before
+ trying to do fixup on Local channels. (issue #7937 reported by
+ mada123, fix by alamantia with mods by me) ........
+
+2006-11-15 21:56 +0000 [r47709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL
+
+2006-11-15 21:33 +0000 [r47707] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: We need to ensure timelimit stuff is included as
+ well so warnings get played. (issue #8050 reported by KNK)
+
+2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/file.c: don't try to call fclose() if fopen() failed
+
+2006-11-15 20:31 +0000 [r47698] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: - Improve SIP history - Never send reply to
+ ACK (again...)
+
+2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006)
+ | 4 lines ensure that message duration is included in email
+ notifications for forwarded messages (BE-96, fix by me after
+ corydon used his clue-bat on me) ensure that duration in the
+ message metadata is updated if prepending is done during
+ forwarding (related to BE-96) remove prototype for API call that
+ does not exist ........
+
+ * main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15
+ Nov 2006) | 2 lines clear the category's variable tail pointer as
+ well when variables are detached from it ........ r47688 |
+ kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2
+ lines when appending a list of variable to a category, ensure the
+ tail pointer points to the last variable in the list ........
+ r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006)
+ | 2 lines when re-writing the config file, don't repeat the path
+ if it hasn't changed ........
+
+ * main/config.c, /: Merged revisions 47682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006)
+ | 2 lines ouch... don't use printf, use ast_log/ast_verbose
+ ........
+
+2006-11-15 17:46 +0000 [r47672] Luigi Rizzo <rizzo@icir.org>
+
+ * main/cli.c: fix longest match search in find_cli. Trunk already
+ fixed. 1.2 not affected (well, i have no idea, the code is
+ totally different there).
+
+2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Send error message when we can't allocate
+ SIP dialog, possibly due to limitation of file descriptors.
+ (imported from 1.2)
+
+2006-11-15 04:45 +0000 [r47645] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: If NAT detection is turned on or already detected
+ then say NAT is active when setting the remote RTP peer when
+ doing early bridging. (issue #8365 reported by marcelbarbulescu)
+
+2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/term.c: more formatting cleanup, and avoid running off the
+ end of the string
+
+2006-11-15 00:14 +0000 [r47639] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Turn notice about unknown RTCP packet type into a
+ debug message instead.
+
+2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/misdn/isdn_lib.c: silence compiler warning on 64-bit
+ platforms (this variable is an 'int' anyway, comparing it to
+ 'signed long' is not useful)
+
+2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2
+ lines Update copyright information in the ADSI logo blob.
+ ........
+
+ * channels/chan_sip.c: Only keep the video RTP structure around if
+ 1. Video support is enabled and 2. A video codec is enabled on
+ the dialog
+
+ * funcs/func_uri.c: Small documentation clarification for
+ URIENCODE. (issue #8294 reported by salaud)
+
+2006-11-14 18:54 +0000 [r47621] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Conversion of res_odbc API to include ast_
+ prefix did not completely transition app_voicemail when
+ ODBC_STORAGE is used (reported on IRC by caio1982, not in
+ bugtracker)
+
+2006-11-14 16:45 +0000 [r47617] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_amd.c: Use LOG_DEBUG to print out the indication that
+ app_amd is using default settings instead of using LOG_NOTICE.
+ This stops needless logging of this information under normal
+ circumstances. (issue #8361 reported by Seb7)
+
+2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Update documentation to fit the
+ implementation...
+
+ * /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in
+ retransmission system if it's an OPTION packet from peerpoke
+
+2006-11-13 21:28 +0000 [r47584] Joshua Colp <jcolp@digium.com>
+
+ * /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2
+ lines Initialize global pointers for connection and result to
+ NULL. (issue #8356 reported by james) ........
+
+2006-11-13 20:20 +0000 [r47581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006)
+ | 2 lines Having more than 255 old messages caused corruption in
+ the new/old count ........
+
+2006-11-13 19:15 +0000 [r47576] Steve Murphy <murf@digium.com>
+
+ * main/config.c: This solves bug 8342, whereby a crash occurs under
+ certain circumstances while reading a config file with comments--
+ a call to CB_ADD shouldn't happen if withcomments is zero
+
+2006-11-13 19:11 +0000 [r47573] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * main/cli.c, channels/chan_sip.c: Re-enable old deprecated
+ commands
+
+2006-11-13 19:10 +0000 [r47572] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: - Don't reply to INVITE already replied
+ to when we get BYE - Declare errmsg as int. Oops.
+
+2006-11-13 18:18 +0000 [r47564] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing
+ the messed if, but we all forgot to update the regressions. Until
+ now.
+
+2006-11-13 17:13 +0000 [r47553] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being
+ found... just confuses users
+
+2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_sms.c: Merged revisions 47549 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2
+ lines When sending an SMS with a user data header properly set
+ the UDH flag in the first byte. (issue #8347 reported by
+ hoffmeis) ........
+
+ * main/cli.c: Free full command string upon unregistering of CLI
+ command. Backported from revision 47536 from rizzo.
+
+2006-11-13 16:00 +0000 [r47540] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Only produce error message about sip history
+ once
+
+2006-11-13 05:48 +0000 [r47527] Russell Bryant <russell@digium.com>
+
+ * configure, acinclude.m4: AC_PROG_SED is included in autoconf
+ 2.60, but apparently it is not included in 2.59. So, to maintain
+ compatability with 2.59 since it is a small change, copy this
+ macro into acinclude.m4 and rename it to AST_PROG_SED. (issue
+ #8345)
+
+2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * res/res_odbc.c, /: Merged revisions 47525 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006)
+ | 2 lines If the execute fails a second time, make sure that we
+ don't pass back a stale handle ........
+
+ * channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006)
+ | 2 lines Don't play dialtone if the seizing the channel fails
+ (Bug 7754) ........
+
+2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks
+ DEA!!!)
+
+ * channels/chan_sip.c: Part of issue 8078 - parse even if udptl is
+ UDPTL in sdp...
+
+ * channels/chan_sip.c: - Don't destroy SIP dialog because of a
+ failed T.38 re-invite. Wait for a bye. Final response to a
+ re-invite does not mean that the session dies, only that the
+ re-invite fails. - Keep RTP active during processing of T.38
+ re-invite. If the re-invite fails, RTP needs to remain as before
+ the re-invite. Issue 8338 - darren1713. Please test.
+
+ * channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp
+ -Add some comments to t.38 code
+
+2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) |
+ 4 lines Only do the check to determine whether the channel
+ calling this function is an IAX2 channel when getting the IP
+ address using the special argument, CURRENTCHANNEL. (issue #8341,
+ jcovert) ........
+
+ * Makefile: Add the target "menuconfig" as an alias for the
+ "menuselect" target. This is just a favor to users so that if you
+ accidentally type "make menuconfig" instead of "make menuselect",
+ it still works. (inspired by a comment on IRC from wangster
+ calling me an "especially devious asterisk developer" for having
+ it be menuselect instead of menuconfig. :) )
+
+ * main/term.c: Tweak the formatting of this new function to better
+ conform to coding guidelines.
+
+2006-11-11 02:04 +0000 [r47490] Matt O'Gorman <mogorman@digium.com>
+
+ * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo
+ safe output!
+
+2006-11-10 22:23 +0000 [r47480] Matt Frederickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Make sure we don't use 32 bits when we only
+ need one bit.
+
+2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: ...and make sure that the dialog is
+ destroyed, even if we don't get any answer on the bye... This is
+ the channel that remains dead after the SIP transfer
+
+ * channels/chan_sip.c: Add debug output while trying to trace bug
+ in bug report
+
+ * channels/chan_sip.c: Make sure we destroy dialog...
+
+ * /, channels/chan_sip.c: Small cleanup of handle_request_invite()
+ - imported from 1.2 with changes
+
+2006-11-10 19:47 +0000 [r47462] Matt Frederickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Fix for #7321. Be able to explicitly hide
+ callerid name for switches that bork on it.
+
+2006-11-10 18:56 +0000 [r47454] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart
+ SDP (alphaque)
+
+2006-11-10 17:13 +0000 [r47444] Luigi Rizzo <rizzo@icir.org>
+
+ * build_tools/prep_moduledeps: grep -m is not available on BSD, so
+ use head -1 instead
+
+2006-11-10 16:53 +0000 [r47437] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_chanspy.c: Only split up extension and context if a
+ value exists. (issue #8332 reported by loloski)
+
+2006-11-10 16:51 +0000 [r47436] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c,
+ channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_iax2.c: Discussion of these CLI changes resulted in
+ more consistency (Bug 8236)
+
+2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c: if adding a queue member is LOG_NOTICE, then
+ removing them should be LOG_NOTICE, not LOG_DEBUG
+
+ * apps/app_queue.c: reflect addition/removal of dynamic queue
+ members in queue_log, so that people using dialplan replacement
+ for AgentCallbackLogin can still track login/logout (issue #7736,
+ reported/patched by whoiswes but this commit was written by me
+ and covers all three paths for AQM/RQM)
+
+2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Rip out half implementation of 491 response
+ support, since it wasn't implemented properly and caused memory
+ leaks in the case of us getting 491's, which Asterisk actually
+ sends... Since it is a bit too complicated to fix this, I'll rip
+ it out of 1.4 and put it on the to-do-list for future releases.
+ Now, we handle this as congestion, which it really is. Issue
+ #8331
+
+ * channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD.
+ Thanks fenlander!
+
+2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_h323.c: Fix building of chan_h323 by completeing
+ some structure definitions. (issue #8327 reported by Mithraen)
+
+ * apps/app_voicemail.c: Do conversion in a more easier to read and
+ working way for \r, \n, and \t. (issue #8324 reported by
+ johnlange)
+
+2006-11-09 21:26 +0000 [r47391] Russell Bryant <russell@digium.com>
+
+ * apps/app_voicemail.c, channels/chan_zap.c,
+ build_tools/prep_moduledeps: Work around an issue that caused
+ menuselect to display a bogus description for app_voicemail and
+ chan_zap. These modules use some preprocessor directives to
+ determine what it will report to Asterisk as its description.
+ However, the way we extract this information from the source
+ files for menuselect is not smart enough to figure this out.
+ (issue #8326, #8328)
+
+2006-11-09 16:53 +0000 [r47380] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_phone.c, /: Merged revisions 47379 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov
+ 2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and
+ higher as, well, it's apparently going to be removed. This should
+ make all you FC6 fans happy as your Asterisk will now build
+ without any mods. ........
+
+2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant <russell@digium.com>
+
+ * main/cli.c: fix tab completion for "core debug channel" and "core
+ no debug channel"
+
+ * main/cli.c: Fix "core show channel". Also, fix tab completion for
+ both "core show channel" and "core show channels".
+
+ * main/cli.c: Fix "core debug channel <whatever>". I guess someone
+ needs to go through and audit every CLI command that changed
+ number of arguments ...
+
+ * main/asterisk.c: revert the previous change, which actually
+ modified the deprecated command, "show profile". Now, actually
+ apply the change to "core show profile".
+
+ * main/asterisk.c: Fix argument parsing for the "core show profile"
+ CLI command (fixed by rizzo in his branch, team/rizzo/astobj2)
+
+ * main/cli.c: Fix another CLI command, "core show uptime" ...
+ (issue #8323, reported by johnlange, fixed by myself)
+
+ * main/asterisk.c: fix "core show version" to reflect the new
+ number of arguments for this CLI command (issue #8316, kshumard)
+
+2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy <murf@digium.com>
+
+ * main/channel.c: This update fixes 7531
+
+ * channels/chan_skinny.c: Committed in behalf of 8190.
+
+2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c: the battle over CLI command formats has broken
+ stuff...
+
+ * channels/chan_sip.c: add simple fix for SDP to report proper
+ sample rate for G.722 media sessions
+
+2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant <russell@digium.com>
+
+ * utils/streamplayer.c: I occasionally get email from users that
+ are trying to figure out what this does, or due to some
+ misunderstanding as to what it is supposed to do, can't get it to
+ work. So, I have added some text here to hopefully explain what
+ this application does and does not do.
+
+ * channels/chan_gtalk.c: Make this module build again
+
+ * configure, configure.ac, acinclude.m4: Copy the macros from
+ libtool.m4 to our own acinclude.m4 such that libtool is no longer
+ required to be installed to be able to generated the configure
+ script.
+
+2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo)
+
+2006-11-07 23:46 +0000 [r47303] Steve Murphy <murf@digium.com>
+
+ * channels/chan_oss.c, main/channel.c, channels/chan_phone.c,
+ channels/chan_misdn.c, channels/chan_skinny.c,
+ channels/chan_features.c, channels/chan_h323.c,
+ channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c,
+ include/asterisk/stringfields.h, apps/app_voicemail.c,
+ main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c,
+ channels/chan_zap.c, channels/chan_sip.c, res/res_features.c,
+ channels/chan_agent.c, main/utils.c, include/asterisk/channel.h,
+ channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to
+ solve the problem in bug 7506. It's a lot of rework to solve a
+ fairly small problem... such is life.
+
+2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c: Make MOH work as it did before in
+ chan_local, without this then it can go funky when transfers and
+ MOH are involved. (issue #7671 reported by jmls)
+
+2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/musiconhold.conf.sample: clean up sample config, and make
+ native file playback the more obvious default choice
+
+2006-11-07 18:38 +0000 [r47275] Matt O'Gorman <mogorman@digium.com>
+
+ * apps/app_voicemail.c: large overhaul to voicemail imap support.
+ Allows support for more imap servers, also a better
+ implementation of several parts of the original work. patch
+ provided by 8033 with major upgrades.
+
+2006-11-07 17:30 +0000 [r47268] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of
+ continue.
+
+2006-11-07 13:13 +0000 [r47250] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Fixing the attack shield so it doesn't
+ produce attacks... Issue 8265 - never reply to an ACK
+
+2006-11-07 01:25 +0000 [r47239] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06
+ Nov 2006) | 5 lines If random order is enabled for files mode
+ music on hold, set a random initial position, instead of always
+ starting at the first file, and doing the random operation only
+ when switching to the next file. (bug reported by John Lange on
+ the asterisk-dev mailing list) ........
+
+2006-11-04 18:32 +0000 [r47199] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and
+ transfer from "john" Thank you!
+
+2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant <russell@digium.com>
+
+ * main/cli.c: Fix another bug in "core set debug" ...
+
+ * main/asterisk.c, main/cli.c: Really fix the "core set debug" and
+ "core set verbose" CLI commands.
+
+ * main/cli.c: fix the "atleast" option to the "core set verbose"
+ and "core set debug" CLI commands
+
+2006-11-03 23:17 +0000 [r47176] Steve Murphy <murf@digium.com>
+
+ * channels/chan_sip.c: This fix introduced via bug 8233
+
+2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo <rizzo@icir.org>
+
+ * bootstrap.sh: align bootstrap.sh with the version in trunk (needs
+ to be blocked as it is already in trunk)
+
+ * configure.ac: add proper environment vars to detect modules on
+ freebsd. (already applied to trunk so it needs to be blocked
+ there)
+
+2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c,
+ channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More
+ changes making the CLI more consistent with "category verb
+ arguments" (continuation of issue 8236)
+
+ * main/config.c, main/cli.c, main/channel.c, main/manager.c,
+ channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c,
+ main/http.c, main/file.c, main/logger.c, main/image.c,
+ res/res_indications.c, main/asterisk.c, res/res_odbc.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
+ channels/chan_local.c, main/frame.c, channels/chan_sip.c,
+ res/res_features.c, channels/chan_agent.c, res/res_crypto.c,
+ res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c:
+ Reverse change of "show" to "list" and make several other
+ commands more consistent with "category verb arguments"
+
+2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Move check for codec translation to
+ sip_call() instead of in add_sdp. No one bothers with the result
+ of add_sdp anyway... Yet...
+
+ * channels/chan_sip.c: Disable code for T38 over TCP and RTP since
+ there's no trace of actual functionality for it :-)
+
+2006-11-02 17:49 +0000 [r46965] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02
+ Nov 2006) | 3 lines ignore files in a music on hold directory
+ that begin with '.' (issue #8249, cboie) ........
+
+2006-11-02 17:17 +0000 [r46963] Nadi Sarrar <ns@beronet.com>
+
+ * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix
+
+2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: don't send INVITE when we have determined
+ that we can't offer any audio formats due to lack of transcoding
+ support (or incorrect configuration)
+
+2006-11-02 16:06 +0000 [r46930] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2
+ lines Repeat after me oej: I will at least make sure my code
+ compiles before I commit it. ........
+
+2006-11-02 15:24 +0000 [r46901] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2)
+
+2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant <russell@digium.com>
+
+ * /, main/callerid.c: Add the missing call to free described in
+ issue #8268. Also, add a bunch of missing calls to free in
+ callerid_feed_jp().
+
+ * main/say.c: fix saying one hundred and two hundred in hebrew
+ (issue #7810, eldadran)
+
+ * Makefile, configure, codecs/gsm/Makefile, configure.ac,
+ build_tools/strip_nonapi, makeopts.in: Fixes for
+ cross-compilation on mips (issue #8058, ywalther, with some
+ modifications)
+
+ * aclocal.m4, build_tools/menuselect-deps.in, configure,
+ build_tools/embed_modules.xml, configure.ac: Add a check in the
+ configure script to determine whether ld is GNU ld or not. This
+ is needed because module embedding only works for gnu ld. GNU ld
+ is now listed as a dependency for all of the module embedding
+ options in menuselect. (issue #8143)
+
+2006-11-01 20:35 +0000 [r46822] Matt O'Gorman <mogorman@digium.com>
+
+ * channels/chan_gtalk.c: bind address support from bug 8164
+
+2006-11-01 19:49 +0000 [r46802] Steve Murphy <murf@digium.com>
+
+ * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
+ accept longer strings or mass confusion and a lot of lost time is
+ the result
+
+2006-11-01 18:39 +0000 [r46780] Joshua Colp <jcolp@digium.com>
+
+ * main/Makefile: Force poll() emulation for Darwin to always be on.
+ It's too broken to consider being used. This resolves the console
+ issue OSX users have been seeing. I would have liked to autoconf
+ this but I haven't been able to come up with a test case that
+ works. Que sera.
+
+2006-11-01 18:26 +0000 [r46778] Russell Bryant <russell@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 46776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) |
+ 9 lines soxmix and Asterisk expect different file extensions for
+ certain formats. This was already handled for the wav49 format.
+ However, it was not handled for ulaw and alaw. I fixed this in
+ such a way that using the alternate extensions for ulaw and alaw
+ will only happen if we know we're calling soxmix, and not a
+ custom script defined using the MONITOR_EXEC variable. The wav49
+ processing was left alone so that external scripts will see no
+ behavior change. (issue #7550, reported by mnicholson, proposed
+ patch by junky, committed fix is a bit different) ........
+
+2006-11-01 18:21 +0000 [r46775] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: It's another round of chan_iax2 fixes!
+ Should hopefully fix the deadlock issues people have been
+ reporting. IAXtel now has qualify turned on for 800 peers and it
+ is handling it fine.
+
+2006-11-01 17:48 +0000 [r46760] Steve Murphy <murf@digium.com>
+
+ * main/config.c: Cleanups suggested by Russell.
+
+2006-11-01 16:39 +0000 [r46744] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: Prevent an infinite loop when config
+ processing gets to a jitterbuffer option
+
+2006-10-31 22:02 +0000 [r46716] Jason Parker <jparker@digium.com>
+
+ * main/translate.c: Fix "core show translation" output. Issue
+ #8243, patch by Damin.
+
+2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/translate.h, main/translate.c: add an API so
+ that translators can activate/deactivate themselves when needed
+
+ * include/asterisk/translate.h, main/translate.c: revert changes
+ that were the wrong way to address this... proper fix coming
+
+ * main/translate.c: let's set the seen flag early enough to
+ actually make a difference...
+
+ * include/asterisk/translate.h, main/translate.c: don't re-do setup
+ operations for translators that can dynamically register
+ themselves
+
+2006-10-31 15:49 +0000 [r46663] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * /: Blocked revisions 46662 via svnmerge ........ r46662 |
+ tilghman | 2006-10-31 09:46:04 -0600 (Tue, 31 Oct 2006) | 3 lines
+ Move thread-unsafe initializer to the module loading code; add
+ the corresponding function to the module unload to fix a memory
+ leak. ........
+
+2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson <oej@edvina.net>
+
+ * main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue
+ #8089 - Fix the ENUM support (picking one record by number).
+ Thanks otmar!
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport
+ when we're supposed to support ;rport. Issue #7473.
+
+ * /, channels/chan_sip.c: If peer fails ACL check, fail peer at
+ REGISTER
+
+ * channels/chan_sip.c: Fix T38 too. Thanks, tgrman !
+
+2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant <russell@digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the
+ boot process to ensure it starts after stuff like MySQL (issue
+ #8253, Alric)
+
+ * /, main/utils.c: Merged revisions 46560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) |
+ 3 lines When handling the case where the hostname is just an IPV4
+ numeric address, be sure to set the address type. (issue #8247,
+ alexr) ........
+
+ * /, res/res_agi.c: Merged revisions 46557 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) |
+ 3 lines fix some copy/paste bugs in the checking of arguments for
+ the "control stream file" AGI command (issue #8255, mnicholson)
+ ........
+
+ * main/translate.c: Add a small tweak to the code that checks to
+ see whether destination formats are translatable based on the
+ source format. If we have already determined that there is no
+ translation path in one direction, don't bother checking the
+ other direction.
+
+2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/translate.c: when unregistering a translator, don't rebuild
+ the translation matrix unless needed when filtering formats out
+ of an offer, ensure we check for translation ability in both
+ directions
+
+ * include/asterisk/linkedlists.h: ensure that items removed from a
+ list are always unlinked from the list (next pointer set to NULL)
+
+2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp <jcolp@digium.com>
+
+ * configure, configure.ac: Don't explicitly link in crypt as it is
+ not used on some platforms.
+
+ * channels/chan_iax2.c: We need to lock the pvt structure during
+ retransmission as another worker thread may be doing something as
+ well.
+
+2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson <oej@edvina.net>
+
+ * main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h,
+ include/asterisk/doxyref.h, channels/chan_sip.c,
+ main/ast_expr2f.c, include/asterisk/module.h,
+ formats/format_ogg_vorbis.c, main/app.c,
+ include/asterisk/channel.h, include/asterisk/lock.h,
+ include/asterisk/frame.h: Issue #8246 - Doxygen fixes from
+ kshumard. An extra big thankyou is given to everyone that
+ contributes to doxygen! THANK YOU!
+
+ * main/rtp.c, /: Bind RTCP to the same IP as RTP
+
+ * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302
+ redirects (imported from 1.2)
+
+ * /, channels/chan_sip.c: Issue #7608 - Notifications sent with
+ wrong content-type (imported from 1.2, modified)
+
+ * channels/chan_sip.c, CHANGES: Backport of patch for #7828 that
+ was reported for trunk, but obviously exists in 1.4 too.
+
+ * channels/chan_sip.c: Restoring the old logic, since working
+ around it and fixing it seemed too complicated. - The
+ SIP_OUTGOING flag indicates the direction of the last transaction
+ in the dialog. - The initreq stores the last request in the
+ dialog, the request that opened the latest transaction. Please
+ now retry all the 1.4 bug reports with mixed to/from headers,
+ tags etc in ACK, BYE, CANCEL. Thanks!
+
+ * channels/chan_sip.c: Accepting a message twice may be
+ misinterpreted...
+
+ * channels/chan_sip.c: - 183 is not reliable message... - Error
+ should not have SDP
+
+2006-10-28 16:37 +0000 [r46377] Joshua Colp <jcolp@digium.com>
+
+ * utils/Makefile: Don't build muted on OpenBSD, it is not
+ supported.
+
+2006-10-27 19:03 +0000 [r46370] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: move the copy of the default settings to the
+ global settings back out of process_zap, so that they aren't
+ overwritten when process_zap is called multiple times
+
+2006-10-27 18:29 +0000 [r46367] Olle Johansson <oej@edvina.net>
+
+ * contrib/asterisk-ng-doxygen: Put some doxygen pressure on
+ Christian :-)
+
+2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, res/res_agi.c, apps/app_externalivr.c,
+ res/res_musiconhold.c: We should always be using _exit() after a
+ fork() or vfork() instead of exit(). This is because exit() does
+ some extra cleanup which in some implementations of vfork(), for
+ example, can actually modify the state of the parent process,
+ causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
+
+ * /: Blocked revisions 46361 via svnmerge ........ r46361 | russell
+ | 2006-10-27 12:36:07 -0500 (Fri, 27 Oct 2006) | 5 lines We
+ should always be using _exit() after a fork() or vfork() instead
+ of exit(). This is because exit() does some extra cleanup which
+ in some implementations of vfork(), for example, can actually
+ modify the state of the parent process, causing very weird bugs
+ or crashes. (issue #7971, Nick Gavrikov) ........
+
+ * channels/chan_zap.c: Instead of iterating all of the options once
+ to look for jitterbuffer options, and then again for everything
+ else, move the processing of jitterbuffer options into the main
+ loop so that there are no erroneous messages about ignoring
+ unknown options. (issue #8226)
+
+2006-10-27 10:03 +0000 [r46351-46353] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
+ Merged revisions 46350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) |
+ 1 line fixed a bug which caused chan_misdn to try to allocate 2
+ times the same channel on high load, which then caused
+ instability of mISDN. removed a useless function from isdn_lib.c
+ ........
+
+ * channels/misdn_config.c: fixed not compile issue, which was just
+ introduced
+
+ * channels/misdn_config.c, channels/chan_misdn.c, /,
+ channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
+ Merged revisions 46176 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) |
+ 1 line added nttimeout option to configure wether we disconnect
+ calls on NT timeouts or not during an overlapdial session
+ ........
+
+2006-10-26 17:57 +0000 [r46335-46340] Jason Parker <jparker@digium.com>
+
+ * /, contrib/scripts/astgenkey.8: Merged revisions 46337 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2
+ lines oops - somebody forgot to change this - long ago, probably.
+ ........
+
+ * CHANGES: grammar check
+
+2006-10-26 16:38 +0000 [r46331] Olle Johansson <oej@edvina.net>
+
+ * CHANGES: Corrections to changes (Multiparking is not included)
+
+2006-10-26 16:31 +0000 [r46329] Russell Bryant <russell@digium.com>
+
+ * main/translate.c: - If the source has no audio or no video
+ portion, do not call powerof() to get the format index. - Don't
+ run through the audio and video loops if there is no audio or
+ video portion of the source If 0 is passed to powerof, it will
+ return -1. This value of -1 was then being used as an array index
+ in these loops, which caused a crash on some systems. Other than
+ this issue, this code works as we expected it to. If a format is
+ not in the source, and we have to translation path to it, it is
+ not offered in the list of acceptable destination formats. (fixes
+ issue #8231)
+
+2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming <kpfleming@digium.com>
+
+ * CHANGES: update to reflect G.722 addition
+
+2006-10-26 04:18 +0000 [r46298] Russell Bryant <russell@digium.com>
+
+ * doc/backtrace.txt: update backtrace documentation to reflect
+ changes in 1.4 (issue #8230, kshumard)
+
+2006-10-26 01:37 +0000 [r46287] Mark Spencer <markster@digium.com>
+
+ * main/config.c, main/manager.c: Fix config comment code
+ preservation code (thanks murf!)
+
+2006-10-25 20:14 +0000 [r46276] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Old todo note - Don't add Contact header on
+ BYE and Cancel
+
+2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant <russell@digium.com>
+
+ * configure.ac: fix error output when checking for openh323 to
+ refer to openh323 instead of pwlib (issue #8222, misaksen)
+
+2006-10-25 19:16 +0000 [r46252] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Somewhat ugly code to try to fix issue
+ #7608. Since the problem was not very well defined, the fix is a
+ bit fuzzy too... Thanks to Luigi for accidentally spotting the
+ possible problem!
+
+2006-10-25 19:08 +0000 [r46249] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c: update warning message to include "agi" option
+ (issue #8225, jmls)
+
+2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile: use 1.4.3 extra sounds with corrected silence
+ files
+
+ * sounds/sounds.xml, sounds/Makefile: add support for prebuilt
+ G.722 prompts and music on hold files
+
+2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: show settings doesn't produce a list of
+ similar objects, it should stay a "show"
+
+2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c,
+ channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c,
+ pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c,
+ main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c,
+ cdr/cdr_custom.c, channels/chan_mgcp.c,
+ apps/app_parkandannounce.c, apps/app_voicemail.c,
+ channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c,
+ res/res_adsi.c, main/utils.c, apps/app_ices.c,
+ pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c,
+ apps/app_getcpeid.c: apparently developers are still not aware
+ that they should be use ast_copy_string instead of strncpy... fix
+ up many more users, and fix some bugs in the process
+
+2006-10-25 04:58 +0000 [r46165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * main/pbx.c: WaitExten truncates decimals of times to wait,
+ instead of accepting them (Bug 8208)
+
+2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c,
+ channels/chan_h323.c, channels/chan_iax2.c,
+ include/asterisk/frame.h: add passthrough and file format support
+ for G.722 16KHz audio (issue #5084, original patch by andrew,
+ updated by mithraen)
+
+ * channels/chan_sip.c, main/translate.c: code zone experiment:
+ don't offer formats in the outbound INVITE that aren't either
+ passthrough or translatable
+
+ * main/translate.c: if multiple translators are registered for the
+ same source/dest combination, ensure that the lowest-cost one is
+ always inserted earlier in the list
+
+2006-10-24 20:30 +0000 [r46142] Mark Spencer <markster@digium.com>
+
+ * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628,
+ #8147)
+
+2006-10-24 19:29 +0000 [r46130] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: We need to initialize our scheduler pthread
+ condition... yes.
+
+2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo <rizzo@icir.org>
+
+ * main/http.c: merge 45152 don't leak descriptors in http.c
+
+ * channels/chan_sip.c: merge 45966 refer_to_domain potentially
+ containing options
+
+ * channels/chan_sip.c: merge 46026 improper checks on get_header()
+ return values
+
+ * channels/chan_sip.c: merge 46045 prevent NULL args to
+ ast_strdupa() in chan_sip.c
+
+2006-10-24 05:23 +0000 [r46093] Russell Bryant <russell@digium.com>
+
+ * Makefile: Restore the ability to remove the firmware directory
+ without causing the installation to fail (issue #8111)
+
+2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/translate.c: ensure that the translation matrix is properly
+ lock-protected every place it is used
+
+ * include/asterisk/translate.h, main/translate.c: add an API call
+ to allow channel drivers to determine which media formats are
+ compatible (passthrough or transcode) with the format an existing
+ channel is already using
+
+ * doc/imapstorage.txt: simplify and correct voicemail IMAP storage
+ build instructions
+
+2006-10-24 03:01 +0000 [r46078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * main/channel.c: Pass through a frame if we don't know what it is,
+ rather than trying to pass a NULL, which will segfault a channel
+ driver (Bug 8149)
+
+2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant <russell@digium.com>
+
+ * utils/muted.c, utils/ael_main.c: In muted.c, check the return
+ value of strdup. In ael_main.c, check the return value of calloc.
+ (issue #8157) In passing fix a few minor bugs in ael_main.c. The
+ last argument to strncpy() was a hard-coded 100, where it should
+ have been 99. I changed this to use sizeof() - 1.
+
+ * apps/app_meetme.c: Fix the descriptions of some of the
+ MeetMeAdmin options (issue #8098, mflorell)
+
+ * res/res_jabber.c: don't crash when an incoming message has no
+ "from" (issue #8205, jmls)
+
+2006-10-23 00:27 +0000 [r45928] Joshua Colp <jcolp@digium.com>
+
+ * /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2
+ lines Don't leak memory mmmk? ........
+
+2006-10-22 21:44 +0000 [r45916] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21
+ Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and
+ couldn't be initialized it would cause a segfault after 'reload'.
+ Reported by Drew/Matt thx. ........
+
+2006-10-21 18:49 +0000 [r45818] Russell Bryant <russell@digium.com>
+
+ * res/res_monitor.c: Add a couple missing unregistrations of
+ manager actions and remove duplicate unregistrations of
+ applications. (issue #8194, jmls)
+
+2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp <jcolp@digium.com>
+
+ * main/loader.c: Don't use promotion on Darwin because it doesn't
+ seem to work quite right in all cases, this should solve the
+ unresolved symbol issue people have been seeing.
+
+ * Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get
+ installed in the proper location (reported on asterisk-dev
+ mailing list)
+
+2006-10-20 07:44 +0000 [r45741] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Let's understand SIP: - REFER can create
+ dialog, Asterisk does not support it yet - NOTIFY can create
+ dialog in Asterisk's implementation (voicemail) even though we
+ don't support the server side of it. In this case, the standard
+ is a side issue ;-) - Added extened functionality for unsupported
+ methods (PING, PUBLISH) so we don't create PVT's for those
+ either. Russellb needs to judge what to do with this in 1.2, but
+ I think the current implementation n 1.2 is a bug since we're
+ sending bad replies to NOTIFY and REFER outside of dialogs
+
+2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp <jcolp@digium.com>
+
+ * res/res_jabber.c: Let's remember to unregister JabberStatus too
+ (issue #8184 reported by jmls)
+
+ * /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct
+ 2006) | 2 lines Respect language selection when seeing if the
+ file exists (issue #8178 reported by mnicholson) ........
+
+ * channels/chan_sip.c: If the jitterbuffer is forced on then we
+ can't partially bridge (reported by wangster on #asterisk-dev)
+
+2006-10-19 00:59 +0000 [r45622] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Don't leak the actual thread-specific
+ sip_pvt struct
+
+2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: don't leak memory when a chan_sip thread is
+ destroyed that has a thread-local temp_pvt allocated
+
+2006-10-18 21:03 +0000 [r45595] Joshua Colp <jcolp@digium.com>
+
+ * main/asterisk.c: Don't modify things if we are using vfork as
+ this is very bad and may cause unexpected behavior (issue #7970
+ reported by Nick Gavrikov)
+
+2006-10-18 11:54 +0000 [r45517] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: remove duplicate declarations
+
+2006-10-18 04:09 +0000 [r45464] Luigi Rizzo <rizzo@icir.org>
+
+ * main/http.c: merge from trunk: move ast_variables_destroy() to a
+ better place in handle_uri() to avoid leaking memory on non
+ existing files.
+
+2006-10-18 03:02 +0000 [r45452] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Don't segfault if you're using a channel driver that
+ doesn't turn RTCP on
+
+2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Don't attempt to access private data members of
+ the pthread_mutex_t object, because this does not work on all
+ linux systems. Instead, just access the reentrancy field in the
+ ast_mutex_info struct when DEBUG_THREADS is enabled. If
+ DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
+ DEBUG_THREADS on as well. (issue #8139, me)
+
+ * configs/sip_notify.conf.sample: update entry to reboot a snom
+ phone (issue #7850, pnlarsson)
+
+2006-10-17 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.4.0-beta3 released.
+
+2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/stringfields.h, main/ast_expr2.c,
+ main/channel.c, channels/chan_sip.c, channels/chan_iax2.c:
+ optimize the 'quick response' code a bit more... no more malloc()
+ or memset() for each response expand stringfields API a bit to
+ allow reusing the stringfield pool on a structure when needed,
+ and remove some unnecessary code when the structure was being
+ freed
+
+2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Don't create a "real" pvt structure for
+ requests that shouldn't be able to create one. Instead use a
+ temporary pvt and fill it with enough information so we can send
+ a reply.
+
+2006-10-17 17:39 +0000 [r45329] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Adding information about Marks
+ direct-RTP hack to the docs...
+
+2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming <kpfleming@digium.com>
+
+ * LICENSE: provide licensing language for IAXy firmware file
+
+2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, apps/app_directed_pickup.c: Backport of new
+ directed pickup (BE-85).
+
+2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson <oej@edvina.net>
+
+ * CREDITS: Adding Inotel to credits for SIP transfers. Thanks for
+ your support!
+
+ * channels/chan_sip.c: Don't destroy dialog for unexpected REFER
+ response...
+
+2006-10-14 04:38 +0000 [r45143] Steve Murphy <murf@digium.com>
+
+ * funcs/func_rand.c: update the doc string for both AEL and
+ extensions.conf users.
+
+2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/acl.c don't drop the entire permit/deny list when an attempt
+ is made to add an invalid entry (BE-92)
+
+2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp <jcolp@digium.com>
+
+ * res/res_speech.c: Clear the quiet flag too since we are
+ restarting a recognition again (reported on -dev by Stephan
+ Edelman)
+
+ * res/res_speech.c: Check return value from engine in case of
+ failure (ie: out of licenses) (reported on -dev mailing list)
+
+2006-10-13 20:52 +0000 [r45103] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-vtest17 (added),
+ pbx/ael/ael-test/ael-vtest17/extensions.ael (added),
+ pbx/ael/ael-test/ael-vtest17 (added),
+ pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in
+ this release via these changes
+
+2006-10-13 19:19 +0000 [r45088] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: avoiding warning, fixing potential bug
+
+2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp <jcolp@digium.com>
+
+ * codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c,
+ codecs/lpc10/decode.c, codecs/lpc10/dcbias.c,
+ codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c,
+ codecs/lpc10/difmag.c, codecs/lpc10/hp100.c,
+ codecs/lpc10/synths.c, codecs/lpc10/preemp.c,
+ codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c,
+ codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c,
+ codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c,
+ codecs/lpc10/lpcini.c, codecs/lpc10/random.c,
+ codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c,
+ codecs/lpc10/placea.c, codecs/lpc10/tbdm.c,
+ codecs/lpc10/analys.c, codecs/lpc10/onset.c,
+ codecs/lpc10/energy.c, codecs/lpc10/deemp.c,
+ codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c,
+ codecs/lpc10/median.c, codecs/lpc10/encode.c,
+ codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c,
+ codecs/lpc10/invert.c: And file said... let the compiler warnings
+ STOP!
+
+ * apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136
+ reported by mnicholson)
+
+ * apps/app_playback.c: Move say.conf existence check to do_say
+ function since it is called from multiple places (issue #8144
+ reported by kshumard)
+
+2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: when sending a call to a peer, use the proper socket if
+ we have multiple bindings (reported on asterisk-dev)
+
+2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Complete merging in RPID screen changes
+ (issue #8101 reported by hristo, patch by oej in revision 44757)
+
+ * main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add
+ the background refresh item back into the scheduler if enabled
+ since it is deleted during reload. (issue #8142 reported by
+ p_lindheimer)
+
+2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/utils.c: use a configure script test for PMTU discovery
+ control instead of just assuming it's available on Linux
+
+2006-10-13 14:45 +0000 [r44994-45026] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some
+ echocandisable issues when bridged. this caused a kernel panic
+ sometimes.. also some minor formatting fixes
+
+ * channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause
+ got a wrong isdn cause at RELEASE_COMPLETE
+
+2006-10-12 22:07 +0000 [r44992] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_sip.c: merge formatting and minor code
+ simplifications from trunk
+
+2006-10-12 20:34 +0000 [r44982] Matt O'Gorman <mogorman@digium.com>
+
+ * channels/chan_gtalk.c: fix for bug 7764.
+
+2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: we can only send one 'a=ptime' attribute per
+ media session, not one for each format
+
+ * main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c,
+ main/utils.c: ensure that IAX2 and SIP sockets allow UDP
+ fragmentation when running on Linux (thanks to Brian Candler on
+ the asterisk-dev list for the tip)
+
+2006-10-12 16:56 +0000 [r44945] Russell Bryant <russell@digium.com>
+
+ * main/manager.c: fix a silly typo in a comment that I saw while
+ reading the commit list
+
+2006-10-12 16:08 +0000 [r44942] Joshua Colp <jcolp@digium.com>
+
+ * Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue
+ #8135 reported by ssokol)
+
+2006-10-12 12:55 +0000 [r44921] Nadi Sarrar <ns@beronet.com>
+
+ * main/manager.c: append_event must be called while holding the
+ session lock
+
+2006-10-12 10:24 +0000 [r44911] Russell Bryant <russell@digium.com>
+
+ * res/res_jabber.c: change some debug output to use LOG_DEBUG
+ instead of verbose output
+
+2006-10-11 16:57 +0000 [r44888] Jason Parker <jparker@digium.com>
+
+ * main/db1-ast/Makefile: These are already set by the parent
+ Makefile.. There is no need to have this here (it doesn't
+ actually work anyways).
+
+2006-10-11 09:18 +0000 [r44854] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: removed warning because of missing
+ prototype declaration
+
+2006-10-10 19:23 +0000 [r44830] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Do not set default/global values in the
+ variable declaration, set it in reload_config()
+
+2006-10-10 17:21 +0000 [r44819] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Move some stuff around so that a NOTIFY
+ dialog won't hang around until the end of the world under certain
+ circumstances
+
+2006-10-10 16:44 +0000 [r44809] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * main/channel.c, funcs/func_channel.c, include/asterisk/channel.h:
+ CHANNEL() function sometime mix parameter and value
+
+2006-10-10 16:42 +0000 [r44808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * funcs/func_logic.c: Lost of a bit of logic when this was
+ simplified between 1.2 and 1.4 (Bug 8117)
+
+2006-10-10 16:30 +0000 [r44806] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Bail out if we have no refer structure and
+ we get a refer response
+
+2006-10-10 16:21 +0000 [r44805] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_sip.c: more merge from trunk (comments and change a
+ static function name)
+
+2006-10-10 15:23 +0000 [r44788] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only set DTMF information if an RTP
+ structure exists
+
+2006-10-10 13:50 +0000 [r44786] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added
+ support of dynamically enabling hdlc on bchannels
+
+2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_sip.c: whitespace changes related to previous
+ commit
+
+ * channels/chan_sip.c: merge a few code simplifications that have
+ gone into trunk during last week, to reduce differences between
+ the two branches and make porting fixes easier.
+
+2006-10-09 16:12 +0000 [r44764] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c: Fix a problem where phones that go
+ "missing" never got unregistered. Issue #8067, reported by pj,
+ patch by Anthony LaMantia (with minor whitespace modifications)
+
+2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid
+ the deadlock
+
+ * channels/chan_iax2.c: Properly avoid a collision with iax2_hangup
+ (issue #8115 reported by vazir)
+
+2006-10-08 14:14 +0000 [r44746] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_sip.c: do not dereference p if we
+ know it is NULL
+
+2006-10-07 14:39 +0000 [r44684] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx, channels/chan_h323.c,
+ channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate
+ caller's transfer capability too
+
+2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_sip.c: put common code in a
+ function to avoid repetitions.
+
+ * channels/chan_sip.c: remove hardwired usage of 5060, use
+ DEFAULT_SIP_PORT instead
+
+ * channels/chan_sip.c: option_debug checking
+ before printing to debug channel.
+
+ * channels/chan_sip.c: backport simplifications on sip_register,
+ usage of ast_set2_flag(), and fixes to the handling of failed
+ module loading.
+
+ * channels/chan_sip.c: improve and document function
+ get_in_brackets(), introducing a helper function
+ find_closing_quote() of more general use.
+
+2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/linkedlists.h: ensure that mutex locks inside
+ list heads are initialized properly on platforms that require
+ constructor initialization (issue #8029, patch from timrobbins)
+
+ * CHANGES: remove Jingle as per mog
+
+2006-10-06 21:08 +0000 [r44628] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Remove the seqno check for RFC2833, the handler is
+ smart enough to not need it.
+
+2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming <kpfleming@digium.com>
+
+ * CHANGES: various cleanups
+
+2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: When the sequence number rolls over then reset the
+ recorded sequence number for DTMF (issue #8106 reported by
+ bungalow)
+
+ * main/file.c: Even more frames to treat as though the remote side
+ disappeared (issue #8097 reported by eldadran)
+
+2006-10-06 15:59 +0000 [r44567] Luigi Rizzo <rizzo@icir.org>
+
+ * main/manager.c, main/http.c: make sure sockets are blocking when
+ they should be blocking.
+
+2006-10-06 12:53 +0000 [r44559-44563] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed segfault which happens during
+ hold/transfer action
+
+ * channels/chan_misdn.c: if INFORMATION Message come with keypad
+ instead of called party number, we just use the keypad as called
+ party number.
+
+ * channels/misdn/isdn_lib.c, channels/misdn_config.c,
+ channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
+ added the option 'reject_cause' to make it possible to set
+ the RELEASE_COMPLETE - cause on the 3. incoming PMP channel,
+ which is automatically rejected because chan_misdn does not
+ support that kind of callwaiting. Therefore chan_misdn supports
+ now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc
+ now gets the info if the requested channel is incoming or
+ outgoing to make the 3. channel possible
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c: fixed the hold/retrieve/transfer issues,
+ removed a useless bc field, added setting of frame.delivery fields,
+ some minor code cleanups
+
+2006-10-05 19:57 +0000 [r44502] Joshua Colp <jcolp@digium.com>
+
+ * main/file.c: Treat busy control frames as hangup in the file streaming
+ core (issue #8097 reported by eldadran)
+
+2006-10-05 18:21 +0000 [r44488] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang.
+ Many thanks to Doug!
+
+2006-10-05 18:01 +0000 [r44486] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite
+ hanging by a thread if the other side is already setup with T.38
+
+2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/app.c: don't segfault when an argument without a close
+ parenthesis is found stop parsing as soon as that situation
+ occurs
+
+2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy <murf@digium.com>
+
+ * CHANGES: I put the accumulated changes from the commit logs and
+ inspection, into CHANGES. Hope everyone approves!
+
+ * configs/muted.conf.sample, utils/muted.c: Hang on a minute, the
+ install process sticks muted.conf in /etc/asterisk, so that's
+ where muted should look for it, right?
+
+2006-10-05 02:40 +0000 [r44450] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Don't totally bail out if T.38 was
+ negotiated
+
+2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: fix Polycom presence notification again
+
+2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo <rizzo@icir.org>
+
+ * utils/Makefile: as far as i can tell astman only uses newt...
+
+ * Makefile: put linker flags in ASTLDFLAGS where they belong
+
+2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE
+ requests add workaround for new Polycom firmware SUBSCRIBE
+ requests (bug is known to exist in 2.0.1 firmware)
+
+ * include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually
+ work
+
+2006-10-04 19:57 +0000 [r44380] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c,
+ pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
+ pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
+ pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
+ pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6,
+ pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8,
+ pbx/ael/ael-test/ael-test16/extensions.ael (added),
+ pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y,
+ pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
+ pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
+ pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the
+ problems reported in bug 8090
+
+2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_oss.c, main/cdr.c, channels/chan_phone.c,
+ main/manager.c, pbx/pbx_spool.c, res/res_smdi.c,
+ channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
+ channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c,
+ main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c,
+ include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c,
+ channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c,
+ main/devicestate.c, main/utils.c, res/res_musiconhold.c,
+ channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update
+ thread creation code a bit reduce standard thread stack size
+ slightly to allow the pthreads library to allocate the stack+data
+ and not overflow a power-of-2 allocation in the kernel and waste
+ memory/address space add a new stack size for 'background'
+ threads (those that don't handle PBX calls) when LOW_MEMORY is
+ defined
+
+2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy <murf@digium.com>
+
+ * configs/muted.conf.sample: I've been meaning to add some
+ explanation about muted... here it is
+
+ * configs/manager.conf.sample: CLI reverbification update to this
+ config file
+
+ * apps/app_macro.c: In response to bug 7776, a Warning has been
+ added to the doc string for Macro().
+
+2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/asterisk.c, main/loader.c, main/term.c, Makefile,
+ include/asterisk.h: ensure that local include files are always
+ used avoid a duplicate function name (term_init())
+
+2006-10-03 22:35 +0000 [r44312] Matt O'Gorman <mogorman@digium.com>
+
+ * channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing
+ client without resource.
+
+2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c: fix a logic error in my previous fix to the queue
+ reload code
+
+2006-10-03 18:42 +0000 [r44286] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx: Change default presentation indicator
+ to "user provided not screened" if octet 3a missed in
+ CallingPartyNumber IE
+
+2006-10-03 18:35 +0000 [r44284] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Use VideoSupport instead so it is considered
+ a valid XML attribute name. (issue #8075 reported by renemendoza)
+
+2006-10-03 18:30 +0000 [r44283] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx: Fix preparation of type and
+ presentation of calling number
+
+2006-10-03 00:01 +0000 [r44240] Matt O'Gorman <mogorman@digium.com>
+
+ * doc/jingle.txt, channels/chan_jingle.c (removed),
+ include/asterisk/jabber.h, configs/jingle.conf.sample (removed),
+ res/res_jabber.c: updated res_jabber for even better component
+ support, soon will be jep-0100 compliant. also removed
+ chan_jingle and infromed info from jingle.txt, chan_gtalk still
+ works and should be used in this version.
+
+2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Change the fd on the I/O context in case it
+ changed during the reload, which is indeed possible. (issue #7943
+ reported by eclubb)
+
+ * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
+ instead of hardcoding the path for the error message (issue #7942
+ reported by eclubb)
+
+2006-10-02 18:52 +0000 [r44186] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * configs/users.conf.sample, pbx/pbx_config.c: Missed part of
+ userconf functionality for chan_h323
+
+2006-10-02 17:25 +0000 [r44169] Joshua Colp <jcolp@digium.com>
+
+ * main/io.c: Shrink when current_ioc is unused. It is set to -1 when
+ unused, not 0. (issue #7941 reported by eclubb)
+
+2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * doc/realtime.txt: Typo fix
+
+ * channels/chan_h323.c: Optimization of oh323_indicate(): less
+ locks - less problems, plus single exit point
+
+2006-10-02 02:38 +0000 [r44146] Mark Spencer <markster@digium.com>
+
+ * channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when
+ you're not talking about a channel :)
+
+2006-10-01 19:32 +0000 [r44135] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/chan_h323.c: Do not simulate any audio tones if we got
+ PROGRESS message
+
+2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant <russell@digium.com>
+
+ * Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to
+ be empty. The cause is that since ASTDATADIR is explicitly
+ exported using "export ASTDATADIR" at the top of the Makefile,
+ make no longer considers the variable "undefined", so the
+ Makefile can't use ?= to set ASTDATADIR if not yet set. (issue
+ #8063, reported by akohlsmith, fixed by me)
+
+ * configs/queues.conf.sample: Fix the name of the "eventmemberstatus"
+ option in the sample queues.conf (issue #8065, adamg)
+
+2006-10-01 15:01 +0000 [r44109] Luigi Rizzo <rizzo@icir.org>
+
+ * channels/chan_sip.c: sync with trunk - move variable declarations
+ to the beginning of a block.
+
+2006-09-30 19:20 +0000 [r44090] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * main/rtp.c: Allow one-way RTP streams (device->Asterisk)
+
+2006-09-30 16:28 +0000 [r44080] Luigi Rizzo <rizzo@icir.org>
+
+ * codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent
+ build problems: - with AST_DEVMODE, building codecs/lpc10 fails
+ because of lots of warnings, and the configure step in editline
+ fails as well. Fix this by removing the -Werror in these steps. -
+ on FreeBSD (but probably on other platforms as well), the final
+ link of asterisk fails because AST_LIBS was not exported to the
+ subdirs Makefiles. Add a proper fix in the top-level Makefile (a
+ possible alternative way is to add "export AST_LIBS" near the
+ beginning of the file). With this fix, i believe that some of the
+ platform-specific conditionals in main/Makefile are redundant
+ (because they should be already dealt with in the top level
+ Makefile) but i don't have a platform to check. Merging to head
+ will happen in a moment.
+
+2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment
+ of previous fix: Issue #7928 - Don't send both 404 and 503. Fix
+ by phsultan with a small fix by me, myself or I. Thanks,
+ Philippe! (This was caused by my changes to the transaction
+ handling)
+
+ * channels/chan_sip.c: Found some buggy SIP clients (phones Planet
+ VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
+ sends ACK not on OK message only (when remote party answers) but
+ on RINGING message too, so when we send 200 OK message, we get
+ unidentified ACK message (because INVITE acknowledged on RINGING
+ message already), so 200 OK retransmits within its retransmission
+ interval then call gets dropped. If someone else knows how to
+ provide workaround for such cases, please, fix it in correct way.
+ Thanks to ssh from #asteriskru for provide access to his box to
+ study and fix this case.
+
+2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming <kpfleming@digium.com>
+
+ * agi, utils: ignore temporary files made by the Makefiles during a
+ build
+
+ * codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile,
+ codecs/Makefile, utils/Makefile, configure,
+ build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac,
+ Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile,
+ pbx/Makefile, res/Makefile, channels/Makefile: fix a few build
+ system bugs, and convert Makefiles to be compatible with GNU make
+ 3.80
+
+2006-09-29 22:35 +0000 [r44053] Jason Parker <jparker@digium.com>
+
+ * main/asterisk.c, main/cli.c: Fix a bug with the removal of
+ 'atleast' argument to 'core verbose' and 'core debug'. Add that
+ argument back in.
+
+2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more
+ carefully when no CallingNumber IE available
+
+ * channels/h323/ast_h323.cxx: Fake display name by called number on
+ incoming calls (until passing connected number/connected name is
+ not implemented)
+
+ * channels/h323/ast_h323.cxx: Ported code refers to H.450 - add
+ includes
+
+ * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly
+ pass TON/PRESENTATION information - original
+ H323Connection::SendSignalSetup() destroys Q.931 fields.
+
+2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/Makefile: yet another place where we were not using the
+ correct CFLAGS by default
+
+ * main/Makefile: missed one conversion to ASTCFLAGS
+
+2006-09-29 18:30 +0000 [r44009] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx, channels/chan_h323.c,
+ channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass
+ TON/PRESENTATION information too
+
+2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile,
+ main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
+ Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse
+ CFLAGS and LDFLAGS for build of Asterisk components, because they
+ are also then used for non-Asterisk components (like menuselect);
+ use our own variables instead
+
+ * configure, configure.ac: support --without-curl in configure
+ script
+
+ * Makefile.rules: another cross-compile fix
+
+ * Makefile: a couple more environment settings that can't leak into
+ the menuselect build
+
+ * main/cli.c: proper fix for ast_group_t change
+
+ * include/asterisk/lock.h: eliminate compiler warning when
+ DEBUG_CHANNEL_LOCKS is enabled and users of this header file
+ don't also include channel.h
+
+2006-09-28 20:11 +0000 [r43944] Jason Parker <jparker@digium.com>
+
+ * apps/app_queue.c: Fix incorrect argument order for member names,
+ on persisted members. Issue 8047, patch by jmls.
+
+2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_playback.c, res/res_monitor.c,
+ include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c,
+ channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c,
+ main/udptl.c, main/frame.c, funcs/func_timeout.c,
+ channels/chan_sip.c, apps/app_festival.c,
+ channels/iax2-provision.c, apps/app_alarmreceiver.c,
+ res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c:
+ Put in missing \ns on the end of ast_logs (issue #7936 reported
+ by wojtekka)
+
+2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c: fix buggy (and overly complex) loop used during reload
+ of app_queue for static member list updating
+
+2006-09-28 17:34 +0000 [r43918] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx: Extend call establishment timeout
+
+2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Make sure the pvt exists before accessing
+ it again as it may have gone away (issue #7562 reported by Seb7
+ and issue #7939 reported by sorg)
+
+ * main/cli.c: Warning be gone!
+
+2006-09-28 16:41 +0000 [r43899] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_queue.c: app_queue is comparing the device names incorrectly
+ while checking their statuses. It's internal list of interfaces
+ includes the dial string, while the argument passed to this
+ function does not have the dial string (/n for a local channel).
+ This causes it to ignore the device state changes because it
+ thinks it belongs to none of its members. (#8040 reported and
+ patch by tim_ringenbach)
+
+2006-09-28 16:17 +0000 [r43893] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Stop the stream after waitstream returns so that our
+ formats get restored. (issue #7370 reported by kryptolus)
+
+2006-09-28 15:56 +0000 [r43877] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx: Fix compiler warning
+
+2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 -
+ tim_ringenbach reported and patched)
+
+ * apps/app_queue.c: Autopause not working for queue members. (#8042
+ - jmls reported and patch)
+
+2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force
+ remote side to start media on outgoing PROGRESS message
+
+ * include/asterisk/compiler.h: Put attribute tag at correct place
+
+2006-09-28 11:03 +0000 [r43852] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c: fixed a bug which led to chan_list zombies,
+ when the call could not be properly established in misdn_call.
+ also removed the ACK_HDLC stuff which is not really needed.
+
+2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx: Do not open transmit channel until
+ TCS is received
+
+ * main/file.c: Don't warn on HOLD/UNHOLD control frames
+
+ * main/file.c: Don't treat unknown control frames as voice
+
+2006-09-27 20:21 +0000 [r43816] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Avoid inability to lock directory log message by
+ creating the directory ahead of time. (Issue 7631)
+
+2006-09-27 19:44 +0000 [r43801-43803] Jason Parker <jparker@digium.com>
+
+ * apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS
+ not being set under certain circumstances. Fix a minor issue, to
+ make it use the filenames that were parsed, instead of the entire
+ argument string. Fix Background() to return -1 like Playback(),
+ if no args are specified.
+
+2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Compensate for out of order packets better if RFC2833
+ compensation is turned on.
+
+ * channels/chan_iax2.c: Get rid of two functions from a time now
+ past (we THINK these are from pre-recursive lock time) that may
+ be contributing to two open issues on the bug tracker (7562/7939)
+ and that has the potential to just make bad things happen if the
+ timing is right.
+
+2006-09-27 16:55 +0000 [r43779] Russell Bryant <russell@digium.com>
+
+ * main/channel.c,res/res_features.c: Fix a problem that occurred if
+ a user entered a digit
+ that matched a bridge feature that was configured using multiple
+ digits, and the digit that was pressed timed out in the feature
+ digit timeout period. For example, if blind transfer is
+ configured as '##', and a user presses just '#'. In this
+ situation, the call would lock up and no longer pass any frames.
+ (issue #7977 reported by festr, and issue #7982 reported by
+ michaels and valuable input provided by mneuhauser and kuj. Fixed
+ by me, with testing help and peer review from Joshua Colp). There
+ are a couple of issues involved in this fix: 1) When
+ ast_generic_bridge determines that there has been a timeout, it
+ returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets
+ this result, it calls ast_generic_bridge over again with the same
+ timestamp for the next event. This results in an endless loop of
+ nothing until the call is terminated. This is resolved by simply
+ changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it
+ sees a timeout. 2) I also changed ast_channel_bridge such that if
+ in the process of calculating the time until the next event, it
+ knows a timeout has already occured, to immediately return
+ AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
+ anyway. 3) In the process of testing the previous two changes, I
+ ran into a problem in res_features where ast_channel_bridge would
+ return because it determined that there was a timeout. However,
+ ast_bridge_call in res_features would then determine by its own
+ calculation that there was still 1 ms before the timeout really
+ occurs. It would then proceed, and since the bridge broke out and
+ did *not* return a frame, it interpreted this as the call was
+ over and hung up the channels. The reason for this was because
+ ast_bridge_call in res_features and ast_channel_bridge in
+ channel.c were using different times for their calculations.
+ channel.c uses the start_time on the bridge config, which is the
+ time that the feature digit was recieved. However, res_features
+ had another time, 'start', which was set right before calling
+ ast_channel_bridge. 'start' will always be slightly after
+ start_time in the bridge config, and sometimes enough to round up
+ to one ms. This is fixed by making ast_bridge_call use the same
+ time as ast_channel_bridge for the timeout calculation. ........
+
+2006-09-27 16:24 +0000 [r43775] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/Makefile: removed the chan_misdn
+ versioning, since Asterisk has it's own
+
+2006-09-27 16:23 +0000 [r43774] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Make rfc2833compensate a global option.
+
+2006-09-27 04:35 +0000 [r43756] Russell Bryant <russell@digium.com>
+
+ * apps/app_voicemail.c: Backport revision 43754 from the trunk,
+ which removes an unused buffer from mm_login to close bug 8038,
+ as well as addresses some formatting and coding guidelines issues
+ in passing. Originally, I did not commit this to 1.4 since it is
+ not necessarily fixing a bug. However, since the IMAP storage
+ code is brand new, I decided it would be better to make the
+ change here as well, in case someone has to work on this code to
+ address issues in the very near future. I don't want to make
+ unnecessary merge problems going to the trunk.
+
+2006-09-27 02:32 +0000 [r43739] Steve Murphy <murf@digium.com>
+
+ * configs/extensions.ael.sample: This change to extensions.ael was
+ to fix bug 8031; the install scripts are causing it to be copied
+ to /etc/asterisk/extensions.ael, and because it is a fairly
+ direct conversion of the original extensions.conf, the macro and
+ context names clash with the existing extensions.conf. So, I put
+ an ael- in front of all macros and contexts, and checked every
+ goto and macro call. Also, this file compiles under aelparse.
+
+2006-09-26 20:56 +0000 [r43710] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Back in revision 4798, this message was changed from
+ using ast_cli() to directly calling write(). During this change,
+ checking if this was a remote console was removed. This caused
+ this message about using "exit" or "quit" to exit an Asterisk
+ console to come up in times where it did not make sense. This
+ change restores the check to see if this is a remote console
+ before printing the message. (fixes BE-65)
+
+2006-09-26 20:47 +0000 [r43707] Joshua Colp <jcolp@digium.com>
+
+ * .cleancount, main/cli.c, channels/chan_sip.c,
+ include/asterisk/channel.h: Use proper type to represent the group variable
+ (issue #8025 reported by makoto)
+
+2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Add missing newline character in the warning
+ message about deprecated TOS values in configuration.
+
+ * apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain
+ mailbox definitions, don't introduce a length limit on the
+ definition by using a 256 byte temporary storage buffer. Instead,
+ make the temporary buffer just as big as it needs to be to hold
+ the entire mailbox definition. (fixes BE-68)
+
+2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c: Strip options off the argument passed for
+ devicestate in chan_local. (issue #8034 reported by pcardozo)
+
+ * apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight
+ overhaul of the whisper support. 1. We need to duplicate the
+ frame from ast_translate 2. We need to ensure we always have
+ signed linear coming in for signed linear combining. 3. We need
+ to ensure we are always feeding signed linear out. 4. Properly
+ store and restore write format when beeping on the channel we are
+ whispering on. 5. Properly discontinue the stream on the channel
+ for the beep. (issue #8019 reported by timkelly1980)
+
+2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile: update to use 1.4.3 core sounds, with corrected
+ beep/beeperr/tt-monkeys files
+
+2006-09-26 18:08 +0000 [r43650-43674] Jason Parker <jparker@digium.com>
+
+ * doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by
+ Dan Austin. Maximum values were incorrect, which is why this is
+ being put in 1.4
+
+ * channels/chan_skinny.c: Add proper codec support to chan_skinny.
+ Works with at least ulaw, alaw, and g729a. This is technically a
+ "new feature", but there are justifications for it. I found a bug
+ with the recent rtp packetization changes, which caused the media
+ setup to fail under certain circumstances, particularly when
+ using allow=all, or having no allow= statements (globally or on
+ the device). I could have either removed the rtp packetization
+ features, or I could add proper codec support (which, without, I
+ think most people would consider to be a bug anyways).
+
+2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Should have moved these lines up in the
+ merge, instead of removing them
+
+ * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1)
+ delete=yes was ignored 2) maxmessages was ignored
+
+2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h,
+ channels/h323/cisco-h225.asn: Fix ASN1 description of
+ non-standard Cisco extensions
+
+ * channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport
+ changes of trunk: 1) r43540: Avoid possible deadlock on channel
+ destruction 2) r43590: Disable fastStart if requested by remote
+ side
+
+2006-09-25 15:23 +0000 [r43616] Jason Parker <jparker@digium.com>
+
+ * sounds/Makefile: One more fix for sounds installation - this time
+ for portability. Reported to asterisk-dev mailing list.
+
+2006-09-25 14:52 +0000 [r43605] Steve Murphy <murf@digium.com>
+
+ * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from
+ crashing if trying to play an OGG moh file.
+
+2006-09-25 06:15 +0000 [r43582] Paul Cadach <paul@odt.east.telecom.kz>
+
+ * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h,
+ channels/chan_h323.c: Merged revisions 43472,43495 from trunk
+
+2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant <russell@digium.com>
+
+ * channels/iax2-provision.c: Fix a CLI command registration issue
+ where an erroneous message claiming that "iax2 show provisioning"
+ was already registered. This was because this command was
+ registering itself as both the command, as well as the command it
+ is deprecating. (issue #8022, reported by bjweeks, fixed by
+ myself)
+
+ * channels/chan_iax2.c:Check to see if the channel that is activating the
+ IAXPEER function is actually an IAX2 channel before proceeding to
+ process it to avoid crashing. (issue #8017, reported by admott,
+ fixed by myself)
+
+2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: don't output the 'build complete' message when the
+ target being run is already going to do an installation
+
+2006-09-22 22:12 +0000 [r43518] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c: Allow chan_skinny.so to be unloaded
+ properly. Remove reload support, since it doesn't
+ actually...work.
+
+2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_ael.c: This commits a change to return
+ MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all
+ goes well for bug 8004
+
+ * pbx/pbx_ael.c: If the extensions.ael file not found, or
+ unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004.
+
+2006-09-22 17:25 +0000 [r43492] Jason Parker <jparker@digium.com>
+
+ * main/cli.c: Make sure we explicitly set the CLI command to not be
+ deprecated, if it isn't.
+
+2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile: use rebuilt extra sounds
+
+ * main/channel.c: all the Linux systems I have don't use
+ '__m_count' for this field, so I don't know where this came
+ from...
+
+2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/threadstorage.h: backport the compatability fix
+ to use attribute_malloc instaed of __attribute__ ((malloc))
+
+ * channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN
+ could not be configured (issue #8006, Mithraen)
+
+ * 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.