diff options
| author | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-12-12 23:20:13 +0000 |
|---|---|---|
| committer | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-12-12 23:20:13 +0000 |
| commit | 7e4fdfa5d4958a54f7129cfedce1fc916d65d08d (patch) | |
| tree | ef611d8b1b82ed57382f4824a4391d76dd469f5c | |
| parent | 5de3df88c9c9237869ba6479aa89079ec13b8b0e (diff) | |
| parent | 5dac894ac0c571a94b973e6685288e01cc49bb93 (diff) | |
Creating tag for the release of asterisk-1.4.0-beta4
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.0-beta4@48429 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | .lastclean | 1 | ||||
| -rw-r--r-- | .version | 1 | ||||
| -rw-r--r-- | ChangeLog | 2673 | ||||
| -rwxr-xr-x | build_tools/prep_tarball | 2 |
4 files changed, 1 insertions, 2676 deletions
diff --git a/.lastclean b/.lastclean deleted file mode 100644 index 6f4247a62..000000000 --- a/.lastclean +++ /dev/null @@ -1 +0,0 @@ -26 diff --git a/.version b/.version deleted file mode 100644 index 4b687e8df..000000000 --- a/.version +++ /dev/null @@ -1 +0,0 @@ -1.4.0-beta4 diff --git a/ChangeLog b/ChangeLog deleted file mode 100644 index 5e49673a8..000000000 --- a/ChangeLog +++ /dev/null @@ -1,2673 +0,0 @@ -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. diff --git a/build_tools/prep_tarball b/build_tools/prep_tarball index 6b2354e5f..5e78349cc 100755 --- a/build_tools/prep_tarball +++ b/build_tools/prep_tarball @@ -5,4 +5,4 @@ # # It will be executed from the top-level directory of the project. -make -C sounds all MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM MENUSELECT_MOH=MOH-FREEPLAY-WAV +make -C sounds all MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM MENUSELECT_MOH=MOH-FREEPLAY-WAV WGET=wget |