[svn-commits] kpfleming: tag 1.4.0-beta4 r48430 - /tags/1.4.0-beta4/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Dec 12 16:20:49 MST 2006
Author: kpfleming
Date: Tue Dec 12 17:20:48 2006
New Revision: 48430
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48430
Log:
importing files for 1.4.0-beta4 release
Added:
tags/1.4.0-beta4/.lastclean (with props)
tags/1.4.0-beta4/.version (with props)
tags/1.4.0-beta4/ChangeLog (with props)
Added: tags/1.4.0-beta4/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.0-beta4/.lastclean?view=auto&rev=48430
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--- tags/1.4.0-beta4/ChangeLog (added)
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@@ -1,0 +1,2673 @@
+2006-12-12 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.4.0-beta4 released.
+
+2006-12-12 04:13 +0000 [r48401] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * /, channels/chan_sip.c: Don't send Contact on MESSAGE
+
+2006-12-05 20:42 +0000 [r48279] Jason Parker <jparker at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at mail.jeffandtilghman.com>
+
+ * main/cli.c: Double-unlock error (reported by blitzrage on IRC)
+
+2006-12-01 17:41 +0000 [r48177] Olle Johansson <oej at 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 at 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 at edvina.net>
+
+ * /, channels/chan_sip.c: Issue 8319 - change noncecount before
+ using it.
+
+2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma.
+
+2006-11-29 04:26 +0000 [r48101] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at digium.com>
+
+ * main/manager.c: Remove a couple of unused variables (issue #8380,
+ casper)
+
+2006-11-27 15:32 +0000 [r48038] Joshua Colp <jcolp at 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 at edvina.net>
+
+ * channels/chan_sip.c: Change logging message
+
+2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy <murf at 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 at 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 at 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 at 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 at edvina.net>
+
+ * /, channels/chan_sip.c: Don't allocate unused variable.
+
+2006-11-22 21:47 +0000 [r47944] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * main/file.c: don't try to call fclose() if fopen() failed
+
+2006-11-15 20:31 +0000 [r47698] Olle Johansson <oej at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * channels/chan_sip.c: Only produce error message about sip history
+ once
+
+2006-11-13 05:48 +0000 [r47527] Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart
[... 1837 lines stripped ...]
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