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diff --git a/.lastclean b/.lastclean new file mode 100644 index 000000000..8f92bfdd4 --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +35 diff --git a/.version b/.version new file mode 100644 index 000000000..5c976b900 --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.6.0-beta4 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..e0af21a4a --- /dev/null +++ b/ChangeLog @@ -0,0 +1,36716 @@ +2008-02-20 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0-beta4 released. + +2008-02-20 22:34 +0000 [r103957] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 103956 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103956 | mmichelson | 2008-02-20 16:32:22 -0600 (Wed, 20 Feb + 2008) | 8 lines Clear up confusion when viewing the + QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the + user's perspective, the queue does exist, we shouldn't tell them + we couldn't find the queue. Instead since it is a dead queue, + report a 0 waiting count This issue was brought up on IRC by jmls + ........ + +2008-02-20 22:29 +0000 [r103954-103955] Joshua Colp <jcolp@digium.com> + + * channels/chan_h323.c: Try to do Packet2Packet bridging with + chan_h323 if reinviting isn't enabled. (closes issue #11901) + Reported by: pj + + * channels/chan_zap.c, /: Merged revisions 103953 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103953 | file | 2008-02-20 18:06:59 -0400 (Wed, 20 Feb 2008) | 6 + lines Don't wait for additional digits when overlap dialing is + enabled if the setup message contains the sending_complete + information element. (closes issue #11785) Reported by: klaus3000 + Patches: sending_complete_overlap_asterisk-1.4.17.patch.txt + uploaded by klaus3000 (license 65) ........ + +2008-02-20 21:41 +0000 [r103908] Mark Michelson <mmichelson@digium.com> + + * channels/chan_local.c, /: Merged revisions 103904 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r103904 | mmichelson | 2008-02-20 15:40:08 -0600 (Wed, + 20 Feb 2008) | 6 lines Fix a crash if the channel becomes NULL + while attempting to lock it. (closes issue #12039) Reported by: + danpwi ........ + +2008-02-20 21:36 +0000 [r103903] Jason Parker <jparker@digium.com> + + * include/asterisk/dsp.h, main/dsp.c: Largely refactor DSP tone + detection routines. Separate fax detection from digit detected. + Added CED (called) tone detection for fax (previously, only CNG + (calling) was supported). Separate DTMF/MF code paths where + appropriate. Allow detection of arbitary tones. (closes issue + #11796) Reported by: dimas Patches: v6-dsp-faxtones.patch + uploaded by dimas (license 88) Tested by: dimas, IgorG, Cache + +2008-02-20 21:08 +0000 [r103902] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Fix a crash due to the wrong variable being + used when building a directory string. (closes issue #12027) + Reported by: jaroth Patches: forward.patch uploaded by jaroth + (license 50) Tested by: jaroth + +2008-02-20 18:29 +0000 [r103846-103847] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/sched.h: Add some documentation fixups + + * /, main/stdtime/localtime.c: Merged revisions 103845 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r103845 | tilghman | 2008-02-20 11:53:00 -0600 (Wed, 20 + Feb 2008) | 7 lines Compat fix for Solaris (closes issue #12022) + Reported by: asgaroth Patches: 20080219__bug12022.diff.txt + uploaded by Corydon76 (license 14) Tested by: asgaroth ........ + +2008-02-20 15:21 +0000 [r103844] Mark Michelson <mmichelson@digium.com> + + * res/res_monitor.c: Fix another spot where a hard-coded '|' hadn't + been converted to ',' (closes issue #12034) Reported by: kowalma + +2008-02-20 03:52 +0000 [r103838-103842] Joshua Colp <jcolp@digium.com> + + * main/audiohook.c: *mumble* + + * main/audiohook.c: file not found. + + * main/audiohook.c: Minor test... + +2008-02-20 00:49 +0000 [r103833] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: When using IMAP storage, if the folder you + attempt to save to does not exist, create it first. (closes issue + #12032) Reported by: jaroth Patches: createfolder.patch uploaded + by jaroth (license 50) Tested by: jaroth + +2008-02-19 22:35 +0000 [r103831-103832] Jason Parker <jparker@digium.com> + + * main/channel.c: Make sure to mask out non-audio first as well + + * main/channel.c: Maybe we should set the value before we test it? + Fixes an issue people have been seeing (unreported?) with file + playback not working. + +2008-02-19 21:54 +0000 [r103824-103828] Joshua Colp <jcolp@digium.com> + + * main/loader.c: Add a log message that appears when you try to + unload a module that isn't loaded. (closes issue #12033) Reported + by: jamesgolovich Patches: asterisk-loader.diff.txt uploaded by + jamesgolovich (license 176) + + * main/file.c: Only output a log message saying the format does not + exist if it actually does not exist, not if the file itself could + not be opened. (closes issue #11828) Reported by: IgorG Patches: + readfile.v1.diff uploaded by IgorG (license 20) + + * /, channels/h323/ast_h323.cxx: Merged revisions 103823 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103823 | file | 2008-02-19 16:28:08 -0400 (Tue, 19 Feb 2008) | 6 + lines Send CallerID Name in setup message. (closes issue #11241) + Reported by: tusar Patches: h323id_as_callerid_name.patch + uploaded by tusar (license 344) ........ + +2008-02-19 20:06 +0000 [r103822] Russell Bryant <russell@digium.com> + + * channels/chan_local.c, /: Merged revisions 103821 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19 + Feb 2008) | 8 lines Account for the fact that the "other" channel + can disappear while the local pvt is not locked. (fixes a problem + introduced in rev 100581) (closes issue #12012) Reported by: + stevedavies Patch by me ........ + +2008-02-19 19:27 +0000 [r103819-103820] Joshua Colp <jcolp@digium.com> + + * apps/app_authenticate.c: len already contains the position we + want to examine, if we move one left again we'll actually + probably be looking at a digit. (issue #12030) Reported by: + alligosh + + * apps/app_channelredirect.c, UPGRADE.txt, CHANGES: Add + CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan + application. This will either be set to NOCHANNEL if the given + channel was not found or SUCCESS if it worked. (closes issue + #11553) Reported by: johan Patches: + UPGRADE.txt.channelredirect.patch uploaded by johan (license 334) + CHANGES.channelredirect.patch uploaded by johan (license 334) + app_channelredirect-20080219.patch uploaded by johan (license + 334) + +2008-02-19 18:14 +0000 [r103818] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_zap.c: (closes issue #11864) Reported by: julianjm + Patches: chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm + (license 99) Patch fixes problem of device state incorrectly + reporting idle before PBX answers incoming call on FXO channel. + Device status is updated now during new channel creation. + +2008-02-19 17:33 +0000 [r103808-103813] Joshua Colp <jcolp@digium.com> + + * /, configure, configure.ac: Merged revisions 103812 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r103812 | file | 2008-02-19 13:31:32 -0400 (Tue, 19 Feb + 2008) | 4 lines Don't look for launchd when cross compiling. + (closes issue #12029) Reported by: ovi ........ + + * /: Blocked revisions 103807 via svnmerge ........ r103807 | file + | 2008-02-19 11:01:42 -0400 (Tue, 19 Feb 2008) | 2 lines Fix + building of chan_sip. ........ + +2008-02-19 00:59 +0000 [r103805] Tilghman Lesher <tlesher@digium.com> + + * main/say.c: Change verbosity into debug for Hebrew (and various + whitespace fixes) (Closes issue #12011) + +2008-02-18 23:58 +0000 [r103798-103802] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 103801 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103801 | file | 2008-02-18 19:56:48 -0400 (Mon, 18 Feb 2008) | + 10 lines Ensure that emulated DTMFs do not get interrupted by + another begin frame. (closes issue #11740) Reported by: gserra + Patches: v1-11740.patch uploaded by dimas (license 88) (closes + issue #11955) Reported by: tsearle (closes issue #10530) Reported + by: xmarksthespot ........ + + * main/channel.c, main/frame.c, channels/chan_sip.c, + include/asterisk/channel.h, include/asterisk/frame.h: Add a + non-invasive API for application level manipulation of T38 on a + channel. This uses control frames (so they can even pass across + IAX2) to negotiate T38 and provided a way of getting the current + status of T38 using queryoption. This should by no means cause + any issues and if it does I will take responsibility for it. + (closes issue #11873) Reported by: dimas Patches: + v4-t38-api.patch uploaded by dimas (license 88) + + * main/frame.c: Add some missing control frames. + +2008-02-18 22:33 +0000 [r103796] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, /: Merged revisions 103795 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103795 | qwell | 2008-02-18 16:28:56 -0600 (Mon, 18 Feb 2008) | + 1 line Fix previous commit so that we actually disable + echocanbridged if echocancel is off. ........ + +2008-02-18 21:57 +0000 [r103794] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Commit chan_zap portion of #11964: add the + ability to get ORIG_CALLED_NUM + +2008-02-18 21:30 +0000 [r103791] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, /: Merged revisions 103790 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103790 | qwell | 2008-02-18 15:23:32 -0600 (Mon, 18 Feb 2008) | + 4 lines Correct a message when echocancelwhenbridged is on, but + echocancel is not. Closes issue #12019 ........ + +2008-02-18 20:58 +0000 [r103788] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Make sure EC is enabled when SS7 call comes + in. Also add support for multiple DPCs per linkset. #11779 + +2008-02-18 20:53 +0000 [r103787] Mark Michelson <mmichelson@digium.com> + + * /, main/app.c: Merged revisions 103786 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb + 2008) | 10 lines There was an invalid assumption when calculating + the duration of a file that the filestream in question was + created properly. Unfortunately this led to a segfault in the + situation where an unknown format was specified in voicemail.conf + and a voicemail was recorded. Now, we first check to be sure that + the stream was written correctly or else assume a zero duration. + (closes issue #12021) Reported by: jakep Tested by: putnopvut + ........ + +2008-02-18 19:47 +0000 [r103783] Michiel van Baak <michiel@vanbaak.info> + + * main/asterisk.c: make the output of 'core show settings' a bit + nicer. (closes issue #12020) Reported by: seanbright Patches: + asterisk.c.patch uploaded by seanbright (license 71) + +2008-02-18 17:45 +0000 [r103781] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c, main/rtp.c: Merged revisions 103780 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008) + | 9 lines When a SIP channel is being auto-destroyed, it's + possible for it to still be in bridge code. When that happens, we + crash. Delay the RTP destruction until the bridge is ended. + (closes issue #11960) Reported by: norman Patches: + 20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14) + Tested by: norman ........ + +2008-02-18 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0-beta3 released. + +2008-02-18 17:12 +0000 [r103772] Olle Johansson <oej@edvina.net> + + * main/channel.c, channels/chan_sip.c: Make sure we can set up + calls without audio (text+video). And ... it works! + +2008-02-18 16:40 +0000 [r103771] Mark Michelson <mmichelson@digium.com> + + * channels/chan_zap.c, /: Merged revisions 103770 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103770 | mmichelson | 2008-02-18 10:37:31 -0600 (Mon, 18 Feb + 2008) | 10 lines Fix a linked list corruption that under the + right circumstances could lead to a looped list, meaning it will + traverse forever. (closes issue #11818) Reported by: michael-fig + Patches: 11818.patch uploaded by putnopvut (license 60) Tested + by: michael-fig ........ + +2008-02-18 16:13 +0000 [r103764-103769] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 103768 via svnmerge ........ r103768 | file + | 2008-02-18 12:11:51 -0400 (Mon, 18 Feb 2008) | 4 lines Backport + fix from issue #9325. (closes issue #11980) Reported by: rbrunka + ........ + + * apps/app_channelredirect.c, main/pbx.c, include/asterisk/pbx.h: + Add an API call (ast_async_parseable_goto) which parses a goto + string and does an async goto instead of an explicit goto. + (closes issue #11753) Reported by: johan + + * /, channels/chan_sip.c: Merged revisions 103763 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103763 | file | 2008-02-18 11:33:14 -0400 (Mon, 18 Feb 2008) | 2 + lines Don't care if the extension given doesn't exist for + subscription based MWI. ........ + +2008-02-18 10:10 +0000 [r103755] Olle Johansson <oej@edvina.net> + + * CHANGES, channels/chan_iax2.c: - No space in manager event names, + please - Add new event to CHANGES + +2008-02-18 04:43 +0000 [r103754] Tilghman Lesher <tlesher@digium.com> + + * build_tools/cflags.xml, main/channel.c, main/pbx.c, + funcs/func_channel.c, include/asterisk/channel.h, CHANGES, + main/cli.c: Context tracing for channels (closes issue #11268) + Reported by: moy Patches: + chantrace-datastored-encapsulated-rev94934.patch uploaded by moy + (license 222) + +2008-02-16 21:22 +0000 [r103750] Michiel van Baak <michiel@vanbaak.info> + + * channels/chan_skinny.c: move two ast_log calls to ast_debug. Now + monitoring chan_skinny port with nagios or zabbix wont generate + noise on the console. @ok tilghman + +2008-02-15 23:32 +0000 [r103742] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 103741 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r103741 | russell | 2008-02-15 17:31:39 -0600 (Fri, 15 + Feb 2008) | 8 lines Fix a crash in chan_iax2 due to a race + condition (closes issue #11780) Reported by: guillecabeza + Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license + 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license + 380) ........ + +2008-02-15 23:20 +0000 [r103740] Mark Michelson <mmichelson@digium.com> + + * CHANGES: Document GotoIfTime change from svn revision 103738 + +2008-02-15 23:14 +0000 [r103739] Russell Bryant <russell@digium.com> + + * include/asterisk/aes.h: Fix a regression in Asterisk 1.6 related + to the use of AES encryption. 1024 was used instead of 128 when + using AES from OpenSSL. Many thanks to d1mas for figuring this + one out! (closes issue #11946) Reported by: bbhoss Patches: + v1-11946.patch uploaded by dimas (license 88) + +2008-02-15 23:07 +0000 [r103737-103738] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c: Add proper "false" case behavior to GotoIfTime + (closes issue #11719) Reported by: kshumard Patches: + gotoiftime.twobranches.patch uploaded by kshumard (license 92) + Tested by: kshumard + + * apps/app_voicemail.c: Fix redeclaration of variables when using + IMAP storage (closes issue #11988) Reported by: jaroth Patches: + variable_cleanup.patch uploaded by jaroth (license 50) + +2008-02-15 19:50 +0000 [r103727-103729] Russell Bryant <russell@digium.com> + + * /, main/loader.c: Merged revisions 103728 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103728 | russell | 2008-02-15 13:50:11 -0600 (Fri, 15 Feb 2008) + | 4 lines In the case that you try to directly reload a module + has returned AST_MODULE_LOAD_DECLINE, log a message indicating + that the module is not fully initialized and must be initialized + using "module load". ........ + + * /, main/loader.c: Merged revisions 103726 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103726 | russell | 2008-02-15 12:33:29 -0600 (Fri, 15 Feb 2008) + | 6 lines Don't attempt to execute the reload callback for a + module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash + that was reported against chan_console in trunk. (closes issue + #11953, reported by junky, fixed by me) ........ + +2008-02-15 17:32 +0000 [r103725] Mark Michelson <mmichelson@digium.com> + + * doc/tex/imapstorage.tex, /, configure, configure.ac: Merged + revisions 103722 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb + 2008) | 8 lines Final round of changes for configure script logic + for IMAP Now if a directory is specified, then we will search + that directory for a source installation of the IMAP toolkit. If + none is found, then we will use that directory as the basis for + detecting a package installation of the IMAP c-client. If that + check fails, then configure will fail. ........ + +2008-02-15 17:29 +0000 [r103723] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, channels/chan_sip.c, res/res_phoneprov.c, + include/asterisk/extconf.h, channels/misdn/isdn_msg_parser.c, + apps/app_queue.c, channels/misdn/isdn_lib.c, main/config.c, + main/channel.c, res/res_config_curl.c, channels/misdn/isdn_lib.h, + main/ast_expr2f.c, channels/misdn/ie.c, + channels/misdn/chan_misdn_config.h, channels/misdn/portinfo.c, + include/asterisk/strings.h, res/res_config_ldap.c, + include/asterisk/time.h: Fix up some doxygen issues. (closes + issue #11996) Patches: bug_11996_doxygen.diff uploaded by snuffy + (license 35) + +2008-02-15 15:45 +0000 [r103716] Tilghman Lesher <tlesher@digium.com> + + * utils/conf2ael.c: Remove extraneous copy (closes issue #12002) + Reported by: junky Patches: conf2ael.diff uploaded by junky + (license 177) + +2008-02-15 15:11 +0000 [r103699-103715] Mark Michelson <mmichelson@digium.com> + + * configure, configure.ac: Merging of changes from 1.4 revision + 103713. + + * /: Blocked revisions 103713 via svnmerge ........ r103713 | + mmichelson | 2008-02-15 09:05:49 -0600 (Fri, 15 Feb 2008) | 8 + lines Fix a bit of wrong logic in the configure script that + caused problems when trying to configure without IMAP. Patch + suggestion from phsultan, but I modified it slightly. (closes + issue #12003) Reported by: pj Tested by: putnopvut ........ + + * doc/tex/imapstorage.tex, configure, configure.ac: Same changes as + made to 1.4 in revision 103710 + + * /: Blocked revisions 103709 via svnmerge ........ r103709 | + mmichelson | 2008-02-14 18:50:49 -0600 (Thu, 14 Feb 2008) | 6 + lines I apparently misunderstood one of the requirements of this + configure change. Now, if a source directory is specified with + the --with-imap option, and a valid source installation is not + detected there, then configure will fail and will not check for a + package installation. ........ + + * doc/tex/imapstorage.tex: Trunk version of 1.4's imap + documentation updates + + * /: Blocked revisions 103703 via svnmerge ........ r103703 | + mmichelson | 2008-02-14 17:49:24 -0600 (Thu, 14 Feb 2008) | 3 + lines Make a small clarification in the documentation ........ + + * /: Blocked revisions 103701 via svnmerge ........ r103701 | + mmichelson | 2008-02-14 17:44:17 -0600 (Thu, 14 Feb 2008) | 3 + lines Update documentation regarding configuration of IMAP + ........ + + * configure, configure.ac: See commit message for svn revision + 103698. This behavior is the same as what is described there. The + difference is that trunk already had the --with-imap=system + option, but it only checked the include path for headers in the + imap directory and not also the c-client directory. + + * /: Blocked revisions 103698 via svnmerge ........ r103698 | + mmichelson | 2008-02-14 17:30:17 -0600 (Thu, 14 Feb 2008) | 13 + lines Change to the configure logic regarding IMAP. Prior to this + commit, if you wished to configure Asterisk with IMAP support, + you would use the --with-imap configure switch in one of the + following two ways: --with-imap=/some/directory would look in the + directory specified for a UW IMAP source installation --with-imap + would assume that you had imap-2004g installed in .. relative to + the Asterisk source With this set of changes the two above + options still work the same, but there are two new behaviors, + too. --with-imap=system will assume that you have -libc-client.so + where you store your shared objects and will attempt to find + c-client headers in your include path either in the imap or + c-client directory. If either of the two original methods of + specifying the imap option should fail, then the check for + --with-imap =system will be performed in addition. It is only + after this "system" check that failure can happen. ........ + +2008-02-14 21:21 +0000 [r103694] Jason Parker <jparker@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac: Modify + ldap autoconf function, so that an incorrect ldap library is not + found on Solaris (it is incompatible). Also removes second check + for awk, which causes Solaris to find an incompatible version of + awk. (closes issue #11829) Reported by: snuffy Patches: + bug-11829.diff uploaded by snuffy (license 35) + +2008-02-14 21:04 +0000 [r103687-103691] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 103690 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r103690 | mmichelson | 2008-02-14 15:03:02 -0600 (Thu, + 14 Feb 2008) | 3 lines Fix build for non-IMAP builds ........ + + * /, apps/app_voicemail.c: Merged revisions 103688 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r103688 | mmichelson | 2008-02-14 14:55:48 -0600 (Thu, + 14 Feb 2008) | 9 lines Fix the new message count if delete=yes + when using IMAP storage. (closes issue #11406) Reported by: + jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license + 50) Tested by: jaroth ........ + + * configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Change + the queue holdtime announcement to happen at any interval (not + just greater than two minutes). Remove the saying of less-than + for holdtime announcements since it can lead to awkward holdtime + announcements. Using '1' as a queue-round-seconds value is no + longer valid. (closes issue #9736) Reported by: caio1982 Patches: + queue_announce5.diff uploaded by caio1982 (license 22) Tested by: + caio1982, putnopvut + +2008-02-14 19:52 +0000 [r103685] Jason Parker <jparker@digium.com> + + * /, funcs/func_cdr.c: Merged revisions 103683 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103683 | qwell | 2008-02-14 13:51:10 -0600 (Thu, 14 Feb 2008) | + 5 lines Document the 'l' option to the CDR() function. (Thanks + voipgate for pointing out the option, and Leif for providing text + for it.) Closes issue #11695. ........ + +2008-02-14 19:47 +0000 [r103682] Jeff Peeler <jpeeler@digium.com> + + * apps/app_externalivr.c: a few syntax changes and safer code + +2008-02-14 18:39 +0000 [r103677] Jason Parker <jparker@digium.com> + + * channels/chan_iax2.c: Add periodic jitter stats to CLI and + manager. (closes issue #8188) Reported by: stevedavies Patches: + jblogging-trunk.patch uploaded by stevedavies + jblogging-trunk_wmgrevent.patch uploaded by johann8384 + updated_jbloggin-trunk_mgrevent.patch uploaded by johann8384 + (license 190) (with additional changes by me) Tested by: + stevedavies, johann8384 + +2008-02-14 10:19 +0000 [r103668] Olle Johansson <oej@edvina.net> + + * res/res_agi.c, apps/app_externalivr.c: Formatting fixes + +2008-02-13 21:04 +0000 [r103662] Jeff Peeler <jpeeler@digium.com> + + * apps/app_externalivr.c: (closes issue #11825) Reported by: + ctooley Patches: additional_eivr_commands.patch uploaded by + ctooley (license 136) Tested by: ctooley + +2008-02-13 15:47 +0000 [r103658] Mark Michelson <mmichelson@digium.com> + + * UPGRADE.txt, res/res_musiconhold.c: 1. Deprecate SetMusicOnHold + and WaitMusicOnHold. 2. Add a duration parameter to MusicOnHold + (closes issue #11904) Reported by: dimas Patches: v2-moh.patch + uploaded by dimas (license 88) Tested by: dimas + +2008-02-13 06:35 +0000 [r103608] Tilghman Lesher <tlesher@digium.com> + + * /: Blocked revisions 103607 via svnmerge ........ r103607 | + tilghman | 2008-02-13 00:25:03 -0600 (Wed, 13 Feb 2008) | 7 lines + We aren't talking to ourselves; we're talking to someone else. + (closes issue #11771) Reported by: msetim Patches: + ami_agent_talkingto-1.4.diff uploaded by caio1982 (license 22) + Tested by: caio1982, msetim ........ + +2008-02-13 00:55 +0000 [r103559] Mark Michelson <mmichelson@digium.com> + + * main/event.c: Fix a small logic error in ast_event_iterator_next. + The previous logic allowed for the iterator to indicate there was + more data than there really was, causing the iterator read beyond + the end of the event structure. This led to invalid memory reads + and potential crashes. + +2008-02-13 00:31 +0000 [r103557] Tilghman Lesher <tlesher@digium.com> + + * /: Blocked revisions 103556 via svnmerge ........ r103556 | + tilghman | 2008-02-12 18:26:57 -0600 (Tue, 12 Feb 2008) | 7 lines + Refuse to load app_voicemail if res_adsi is not loaded (which is + a symbol dependency) (closes issue #11760) Reported by: + non-poster Patches: 20080114__bug11760.diff.txt uploaded by + Corydon76 (license 14) Tested by: Corydon76, non-poster, + jamesgolovich ........ + +2008-02-12 22:26 +0000 [r103447-103506] Jason Parker <jparker@digium.com> + + * /: Blocked revisions 103503 via svnmerge ........ r103503 | qwell + | 2008-02-12 16:22:54 -0600 (Tue, 12 Feb 2008) | 1 line Remove + condition that was impossible. ........ + + * main/manager.c: Even more sane permissions. This should be + handled via a umask, like in many other places. + + * main/manager.c: Use slight more sane permissions + +2008-02-12 15:39 +0000 [r103387-103388] Russell Bryant <russell@digium.com> + + * main/asterisk.c: Remove development version notice. + + * main/manager.c: Fix build on *BSD. These permissions constants + are not available there. + +2008-02-12 15:13 +0000 [r103386] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 103385 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4 + lines Even if no CallerID name or number has been provided by the + remote party still use the configured sip.conf ones. (closes + issue #11977) Reported by: pj ........ + +2008-02-12 14:08 +0000 [r103341] Philippe Sultan <philippe.sultan@gmail.com> + + * include/asterisk/jabber.h, res/res_jabber.c: Use an ast_flags + structure in aji_client and aji_buddy rather than an integer. + Modify calls to various ast_*_flag macros accordingly. + +2008-02-12 00:24 +0000 [r103331] Jeff Peeler <jpeeler@digium.com> + + * main/manager.c, include/asterisk/config.h, CHANGES, + main/config.c: Requested changes from Pari, reviewed by Russell. + Added ability to retrieve list of categories in a config file. + Added ability to retrieve the content of a particular category. + Added ability to empty a context. Created new action to create a + new file. Updated delete action to allow deletion by line number + with respect to category. Added new action insert to add new + variable to category at specified line. Updated action newcat to + allow new category to be inserted in file above another existing + category. + +2008-02-11 22:10 +0000 [r103317-103325] Joshua Colp <jcolp@digium.com> + + * /, apps/app_meetme.c: Merged revisions 103324 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103324 | file | 2008-02-11 18:09:07 -0400 (Mon, 11 Feb 2008) | 4 + lines If entering a conference with the 'w' option ensure that we + can't listen or speak until the marked user appears. (closes + issue #11835) Reported by: alanmcmillan ........ + + * res/res_agi.c: Remove ast_module_user usage from res_agi. This is + taken care of in the core. + + * main/pbx.c, main/manager.c, main/translate.c, main/logger.c, + main/app.c, main/utils.c, main/indications.c, main/asterisk.c, + main/rtp.c: Just some minor coding style cleanup... + + * main/pbx.c: Fix Manager Redirect while in an AGI. (closes issue + #10661) Reported by: junky + +2008-02-11 17:09 +0000 [r103316] Kevin P. Fleming <kpfleming@digium.com> + + * /, configs/zapata.conf.sample: Merged revisions 103315 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103315 | kpfleming | 2008-02-11 11:05:22 -0600 (Mon, 11 Feb + 2008) | 2 lines improve 2BCT documentation a bit (thanks Jared) + ........ + +2008-02-11 16:17 +0000 [r103313-103314] Joshua Colp <jcolp@digium.com> + + * main/channel.c, channels/chan_iax2.c: Add support for allowing a + native bridge to happen when the L option is enabled. The RTP + bridging could already handle this, it just needed to be enabled + in the main bridging code. (issue #10647) Reported by: samdell3 + + * channels/chan_skinny.c: Change chan_skinny to use debug messages + as appropriate. (closes issue #11967) Reported by: mvanbaak + Patches: 2008021000-skinnydebug.diff.txt uploaded by mvanbaak + (license 7) + +2008-02-11 06:05 +0000 [r103306] James Golovich <james@gnuinter.net> + + * channels/chan_sip.c: Don't wipe out transport and fd in chan_sip + on reload (issue #11930) + +2008-02-11 03:03 +0000 [r103282-103284] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fix improper indentation. Thanks again to + snuffy for pointing it out. + + * apps/app_queue.c: Add a couple of comments to clarify the + unreffing of queues. Thanks to snuffy for the idea. + + * main/event.c: Fix a problem regarding network vs. host byte order + in the event API. ast_event_iterator_get_ie_type should return + the ie type in host byte order. Furthermore, ast_event_get_ie_raw + should already have its ie type argument in host byte order since + it could be called externally (and it in fact is called in this + way by ast_event_get_cached). + +2008-02-09 11:27 +0000 [r103249] Michiel van Baak <michiel@vanbaak.info> + + * apps/app_dial.c, apps/app_dictate.c, apps/app_echo.c, + apps/app_authenticate.c, apps/app_disa.c, apps/app_chanisavail.c, + apps/app_exec.c, apps/app_db.c, apps/app_controlplayback.c, + apps/app_channelredirect.c, apps/app_directed_pickup.c, + apps/app_dumpchan.c, apps/app_amd.c, apps/app_externalivr.c, + apps/app_directory.c, apps/app_chanspy.c, apps/app_cdr.c: + whitespace fixes only. + +2008-02-09 06:33 +0000 [r103198] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 103197 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r103197 | tilghman | 2008-02-09 00:23:49 -0600 (Sat, 09 + Feb 2008) | 4 lines Commit fix for being unable to send voicemail + from VoiceMailMain Reported by: William F Acker (via the -users + mailing list) Patch by: Corydon76 (license 14) ........ + +2008-02-08 21:26 +0000 [r103171] Russell Bryant <russell@digium.com> + + * main/udptl.c, main/pbx.c, channels/chan_sip.c, + channels/chan_iax2.c, res/res_jabber.c, apps/app_playback.c, + main/rtp.c, channels/chan_usbradio.c, main/cdr.c, + channels/chan_skinny.c, apps/app_minivm.c, res/res_agi.c, + pbx/pbx_ael.c, pbx/pbx_dundi.c, funcs/func_devstate.c, + apps/app_rpt.c, main/asterisk.c, channels/chan_mgcp.c, + apps/app_voicemail.c: Merge changes from + team/mvanbaak/cli-command-audit (closes issue #8925) About a year + ago, as Leif Madsen and Jim van Meggelen were going over the CLI + commands in Asterisk 1.4 for the next version of their book, they + documented a lot of inconsistencies. This set of changes + addresses all of these issues and has been reviewed by Leif. + While this does introduce even more changes to the CLI command + structure, it makes everything consistent, which is the most + important thing. Thanks to all that helped with this one! + +2008-02-08 18:58 +0000 [r103071-103122] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Forgot that AST_LIST_REMOVE_CURRENT takes + different arguments in trunk than 1.4. + + * /, apps/app_queue.c: Merged revisions 103120 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r103120 | mmichelson | 2008-02-08 12:48:17 -0600 (Fri, 08 Feb + 2008) | 10 lines Prevent a potential three-thread deadlock. Also + added a comment block to explicitly state the locking order + necessary inside app_queue. (closes issue #11862) Reported by: + flujan Patches: 11862.patch uploaded by putnopvut (license 60) + Tested by: flujan ........ + + * /, channels/chan_iax2.c: Merged revisions 103070 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r103070 | mmichelson | 2008-02-08 12:00:38 -0600 (Fri, + 08 Feb 2008) | 6 lines Yield the thread and return -1 if the + ioctl fails for Zaptel timing device. (closes issue #11891) + Reported by: tzafrir ........ + +2008-02-08 16:49 +0000 [r103044] Russell Bryant <russell@digium.com> + + * UPGRADE-1.2.txt (added), UPGRADE-1.4.txt (added), UPGRADE.txt: At + the request of ManxPower, include the UPGRADE.txt from 1.2 and + 1.4, as well. This way, if people need to go back and review what + was deprecated in previous major releases, it is readily + available to them. Thanks for the suggestion! + +2008-02-08 15:31 +0000 [r102969-103018] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Fix a network byte order issue and ensure + when creating an outgoing dialog that the socket always contains + information such as type and port. (closes issue #11916) Reported + by: mnnojd + + * /, channels/chan_iax2.c: Merged revisions 102968 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r102968 | file | 2008-02-08 11:08:20 -0400 (Fri, 08 Feb + 2008) | 4 lines Make sure the presence of dbsecret is factored + into user scoring. (closes issue #11952) Reported by: bbhoss + ........ + +2008-02-07 21:37 +0000 [r102933] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c: This is a combination new feature/bug fix for + app_chanspy. New feature: Add the 'e' option, which takes as an + argument a list of interfaces separated by colons. This way, you + will only be able to spy on this limited list of interfaces. Bug + fix: change some pointer checks to ast_strlen_zero so that spying + would work properly even if no channel was specified as the first + argument to chanspy. (closes issue #10072) Reported by: + xmarksthespot Patches: + bugfix+newfeature10072patchtotrunkrev102726.diff uploaded by + xmarksthespot (license 16) Tested by: xmarksthespot, mvanbaak + +2008-02-07 21:08 +0000 [r102906-102908] Michiel van Baak <michiel@vanbaak.info> + + * apps/app_adsiprog.c: whitespace fixes only + + * apps/app_alarmreceiver.c: There she goes! First commit from me to + trunk \o/ Make app_alarmreceiver honor code guidelines and fix + whitespace errors. No functional changes. + +2008-02-07 20:02 +0000 [r102859] Jason Parker <jparker@digium.com> + + * /, main/features.c: Merged revisions 102858 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r102858 | qwell | 2008-02-07 13:53:55 -0600 (Thu, 07 Feb 2008) | + 7 lines Specify which digit string was matched in debug message. + (closes issue #11949) Reported by: dimas Patches: + v1-feature-debug.patch uploaded by dimas (license 88) ........ + +2008-02-07 16:47 +0000 [r102808] Kevin P. Fleming <kpfleming@digium.com> + + * /, configs/zapata.conf.sample: Merged revisions 102807 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r102807 | kpfleming | 2008-02-07 10:41:55 -0600 (Thu, 07 Feb + 2008) | 2 lines document usage of 'transfer' configuration option + for ISDN PRI switch-side transfers ........ + +2008-02-06 20:12 +0000 [r102777] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Add the channel's unique id to the AgentCalled + manager event to make it more consistent with other manager + events. + +2008-02-06 18:01 +0000 [r102726] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 102725 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r102725 | file | 2008-02-06 13:59:23 -0400 (Wed, 06 Feb 2008) | 2 + lines Only consider a T.38-only INVITE compatible if we have both + a joint capability between us and them and if they provided T.38. + ........ + +2008-02-06 16:23 +0000 [r102700] Terry Wilson <twilson@digium.com> + + * funcs/func_realtime.c: Add REALTIME_STORE and REALTIME_DESTROY + dialplan functions provided by sergee. I just added the ability + to set multiple fields at once after discussions with Tilghman + and Russell. Currently limited to 30 fields. (closes issue + #11887) Reported by: sergee Patches: + rt-func-store-destroy-multivalue.diff uploaded by otherwiseguy + (license 396) Tested by: sergee, otherwiseguy + +2008-02-06 15:46 +0000 [r102654] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 102653 via svnmerge ........ r102653 | file + | 2008-02-06 11:43:38 -0400 (Wed, 06 Feb 2008) | 4 lines Add + missing header file and ASTERISK_FILE_VERSION usage. (closes + issue #11936) Reported by: snuffy ........ + +2008-02-06 15:20 +0000 [r102652] Russell Bryant <russell@digium.com> + + * /, configs/features.conf.sample: Merged revisions 102651 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r102651 | russell | 2008-02-06 09:19:41 -0600 (Wed, 06 Feb 2008) + | 3 lines Clarify setting DYNAMIC_FEATURES so that it gets + inherited by outbound channels. (due to a discussion between me + and a user via email) ........ + +2008-02-06 03:05 +0000 [r102602] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 102576 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r102576 | tilghman | 2008-02-05 18:26:02 -0600 (Tue, 05 + Feb 2008) | 4 lines Move around some defines to unbreak ODBC + storage. (closes issue #11932) Reported by: snuffy ........ + +2008-02-06 00:08 +0000 [r102501-102550] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Remove an extra debug message I left in + + * channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, + apps/app_privacy.c, apps/app_alarmreceiver.c, res/res_jabber.c, + apps/app_followme.c, main/loader.c, channels/chan_usbradio.c, + main/tcptls.c, res/res_agi.c, apps/app_minivm.c, + apps/app_dumpchan.c, main/logger.c, apps/app_zapras.c, + main/astmm.c: Get rid of any remaining ast_verbose calls in the + code in favor of ast_verb (closes issue #11934) Reported by: + mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by + mvanbaak (license 7) + + * apps/app_voicemail.c: Change verbose messages to use the ast_verb + macro. (closes issue #11931) Reported by: snuffy Patches: + bug-11931.diff uploaded by snuffy (license 35) + +2008-02-05 20:51 +0000 [r102500] Jason Parker <jparker@digium.com> + + * main/pbx.c: Change where priority of a goto is adjusted. + Partially reverts 102272. Closes issue #11929 (credit to file for + fix suggestion - we still <3 you) + +2008-02-05 20:03 +0000 [r102454] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_mgcp.c: Merged revisions 102453 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r102453 | mmichelson | 2008-02-05 14:02:44 -0600 (Tue, + 05 Feb 2008) | 8 lines Clear the DTMF buffer on hangup. (closes + issue #11919) Reported by: eferro Patches: + mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337) + Tested by: eferro ........ + +2008-02-05 19:58 +0000 [r102379-102452] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Yeah yeah, I broke building on trunk. Shoot + me. + + * /, channels/chan_sip.c: Merged revisions 102450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r102450 | file | 2008-02-05 15:52:30 -0400 (Tue, 05 Feb 2008) | 3 + lines If a REGISTER attempt comes in that is a retransmission of + a previous REGISTER do not create a new nonce value. (issue + #BE-381) ........ + + * /, res/res_clioriginate.c: Merged revisions 102378 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r102378 | file | 2008-02-05 11:09:29 -0400 (Tue, 05 Feb + 2008) | 4 lines Perform dialing asynchronously when using the + originate CLI command so the CLI does not appear to block. + (closes issue #11927) Reported by: bbhoss ........ + +2008-02-04 21:15 +0000 [r102329] Tilghman Lesher <tlesher@digium.com> + + * utils/muted.c, /, configure, include/asterisk/autoconfig.h.in, + configure.ac, main/asterisk.c: Merged revisions 102323 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r102323 | tilghman | 2008-02-04 15:06:09 -0600 (Mon, 04 Feb 2008) + | 7 lines Cross-platform fix: OS X now deprecates the use of the + daemon(3) API. (closes issue #11908) Reported by: oej Patches: + 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14) + Tested by: Corydon76 ........ + +2008-02-04 18:39 +0000 [r102297] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c: Add line numbers to warning/error messages + (and pretty up some existing ones). (closes issue #11894) + Reported by: jmls Patches: chan_zap.patch uploaded by jmls + (license 141) + +2008-02-04 15:16 +0000 [r102272] Joshua Colp <jcolp@digium.com> + + * main/pbx.c: Update handling of asyncgoto so it properly works on + channels that are currently executing a PBX. (closes issue + #11914) Reported by: arnd (closes issue #11753) Reported by: + johan + +2008-02-04 14:37 +0000 [r102262] Jason Parker <jparker@digium.com> + + * configs/extensions.ael.sample, configs/extensions.lua.sample: + Change examples to use G here also. Closes issue #11875 + +2008-02-04 05:32 +0000 [r102190-102238] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_strings.c: Merged revisions 102214 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r102214 | tilghman | 2008-02-03 23:10:02 -0600 (Sun, 03 + Feb 2008) | 6 lines Missing braces. (closes issue #11912) + Reported by: dimas Patches: sprintf.patch uploaded by dimas + (license 88) ........ + + * main/manager.c: CoreSettings and CoreStatus are missing the + terminating "\r\n". Also, some miscellaneous spacing and + initialization issues. (closes issue #11909) Reported by: srt + Patches: patch-11909-2.diff uploaded by srt (license 378) Tested + by: srt + +2008-02-03 16:46 +0000 [r102091-102143] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 102142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r102142 | oej | 2008-02-03 17:38:12 +0100 (Sön, 03 Feb 2008) | 8 + lines Use the same CSEQ on CANCEL as on INVITE (according to RFC + 3261) (closes issue #9492) Reported by: kryptolus Patches: + bug9492.txt uploaded by oej (license 306) Tested by: oej ........ + + * /, channels/chan_sip.c: Merged revisions 102090 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8 + lines Handle ACK and CANCEL in an invite transaction - even if we + get INFO transactions during the actual call setup. (closes issue + #10567) Reported by: jacksch Tested by: oej Patch by: oej + inspired by suggestions from neutrino88 in the bug tracker + ........ + +2008-02-03 06:43 +0000 [r102064] Russell Bryant <russell@digium.com> + + * configure, configure.ac: Change the version number in the + configure script from 1.4 to 1.6 + +2008-02-02 06:10 +0000 [r101990-102037] Russell Bryant <russell@digium.com> + + * include/asterisk/event.h: The documentation page has to be in its + own comment block to work, apparently. Fix it up! + + * /, channels/chan_sip.c: Merged revisions 101989 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008) + | 5 lines Change the SDP_SAMPLE_RATE macro. It turns out that + even though G.722 is 16 kHz, it is supposed to specified as 8 kHz + in the RTP, and RTP timestamps are supposed to be calculated + based on 8 kHz. (Apparently this is due to a bug in a spec, but + people follow it anyway, because it's the spec ...) ........ + +2008-02-01 22:12 +0000 [r101873-101943] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 101942 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r101942 | tilghman | 2008-02-01 15:54:28 -0600 (Fri, 01 + Feb 2008) | 8 lines Fix the VM_DUR variable for forwarded + voicemail, and fixed several other bugs while I'm in the area. + (closes issue #11615) Reported by: jamessan Patches: + 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14) + Tested by: Corydon76, jamessan ........ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 101894 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101894 | tilghman | 2008-02-01 13:36:12 -0600 (Fri, 01 Feb 2008) + | 2 lines Change detection of getifaddrs to use + AST_C_COMPILE_CHECK, backported from trunk (as suggested by + kpfleming) ........ + + * res/res_config_curl.c: Fix multi, when using the LIKE query. + (closes issue #11889) Reported by: jmls Patches: + res_config_curl.patch uploaded by jmls (license 141) Tested by: + jmls + +2008-02-01 18:24 +0000 [r101869] Jason Parker <jparker@digium.com> + + * apps/app_authenticate.c: Comparison, not set :) Thanks mvanbaak. + +2008-02-01 18:08 +0000 [r101824] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c, configs/res_odbc.conf.sample: Clarify the pooling + functionality by changing the config file keyword + +2008-02-01 17:44 +0000 [r101823] Jason Parker <jparker@digium.com> + + * /, apps/app_authenticate.c: Move an feof() call to before the + fgets(). This would have exited the loop early if you had an + authentication file with no newline at the end. + +2008-02-01 17:28 +0000 [r101819-101821] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 101820 via svnmerge ........ r101820 | + russell | 2008-02-01 11:27:02 -0600 (Fri, 01 Feb 2008) | 1 line + off by one error ........ + + * /, apps/app_authenticate.c: Merged revisions 101818 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r101818 | russell | 2008-02-01 11:23:47 -0600 (Fri, 01 + Feb 2008) | 4 lines Don't overwrite the last character of a line + if it's not a newline. This would happen if the last line in the + file doesn't have a newline. (pointed out by Qwell) ........ + +2008-02-01 16:01 +0000 [r101773] Tilghman Lesher <tlesher@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + main/acl.c: Merged revisions 101772 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101772 | tilghman | 2008-02-01 09:55:58 -0600 (Fri, 01 Feb 2008) + | 2 lines Compatibility fix for OpenWRT (reported by Brian + Capouch via the mailing list) ........ + +2008-02-01 06:27 +0000 [r101694-101746] Russell Bryant <russell@digium.com> + + * apps/app_authenticate.c: simplify some code, tweak formatting, + and reduce indentation + + * apps/app_authenticate.c: reduce a level of indentation + + * apps/app_channelredirect.c: Get rid of a goto where there was no + extra cleanup happening at the exit point + + * /, channels/chan_iax2.c: Merged revisions 101693 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r101693 | russell | 2008-01-31 18:32:49 -0600 (Thu, 31 + Jan 2008) | 8 lines Add some more sanity checking on IAX2 dial + strings for the case that no peer or hostname was provided, which + is the one part of the dial string that is absolutely required. + If it's not there, bail out. (closes issue #11897) Reported by + sokhapkin Patch by me ........ + +2008-02-01 00:08 +0000 [r101650] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_amd.c: Merged revisions 101649 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101649 | mmichelson | 2008-01-31 18:06:37 -0600 (Thu, 31 Jan + 2008) | 9 lines From bugtracker: "fix totalAnalysisTime to handle + periods of no channel activity" (closes issue #9256) Reported by: + cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt + uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81, + rjain ........ + +2008-01-31 23:14 +0000 [r101611] Russell Bryant <russell@digium.com> + + * /, main/translate.c, main/file.c: Merged revisions 101601 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101601 | russell | 2008-01-31 17:10:06 -0600 (Thu, 31 Jan 2008) + | 12 lines Fix a couple of places where ast_frfree() was not + called on a frame that came from a translator. This showed itself + by g729 decoders not getting released. Since the flag inside the + translator frame never got unset by freeing the frame to indicate + it was no longer in use, the translators never got destroyed, and + thus the g729 licenses were not released. (closes issue #11892) + Reported by: xrg Patches: 11892.diff uploaded by russell (license + 2) Tested by: xrg, russell ........ + +2008-01-31 22:12 +0000 [r101578-101580] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Forgot an ! + + * apps/app_queue.c: A change I made to accommodate the "linear" + strategy in trunk caused queue strategies to not be loaded from + realtime queues. This commit fixes that. Thanks to jmls for + pointing this problem out to me on IRC. This also contains some + changes to S_OR where it should be used. Thanks to Qwell for + pointing these out. + +2008-01-31 21:33 +0000 [r101577] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Fix a simple deadlock that was introduced + _right_ before this code got merged into trunk. (closes issue + #11895, reported by pj, patched by me) + +2008-01-31 21:31 +0000 [r101532-101576] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Handle the case of a NULL state_interface when + checking a realtime member. Thanks to jmls for finding this + issue. + + * /, res/res_monitor.c: Merged revisions 101531 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101531 | mmichelson | 2008-01-31 15:00:24 -0600 (Thu, 31 Jan + 2008) | 10 lines 1. Prevent the addition of an extra '/' to the + beginning of an absolute pathname. 2. If ast_monitor_change_fname + is called and the new filename is the same as the old, then exit + early and don't set the filename_changed field in the monitor + structure. Setting it in this case was causing ast_monitor_stop + to erroneously delete them. (closes issue #11741) Reported by: + garlew Tested by: putnopvut ........ + +2008-01-31 19:54 +0000 [r101483] Jason Parker <jparker@digium.com> + + * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions + 101482 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101482 | qwell | 2008-01-31 13:52:49 -0600 (Thu, 31 Jan 2008) | + 4 lines Solaris compat fixes for struct in_addr funkiness. Issue + #11885, patch by snuffy. ........ + +2008-01-31 19:43 +0000 [r101481] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 101480 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101480 | murf | 2008-01-31 12:30:37 -0700 (Thu, 31 Jan 2008) | 1 + line closes issue #11845; that's the one where there's a 1004 + byte cdr leak with every AMI Redirect to a zap channel ........ + +2008-01-31 19:20 +0000 [r101416-101449] Russell Bryant <russell@digium.com> + + * /, channels/chan_agent.c: Merged revisions 101433 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r101433 | russell | 2008-01-31 13:17:05 -0600 (Thu, 31 + Jan 2008) | 2 lines Add more missing locking of the agents list + ... ........ + + * /, channels/chan_agent.c: Merged revisions 101413-101414 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101413 | russell | 2008-01-31 13:04:52 -0600 (Thu, 31 Jan 2008) + | 2 lines Add missing locking to the find_agent() function. + ........ r101414 | russell | 2008-01-31 13:07:46 -0600 (Thu, 31 + Jan 2008) | 3 lines Move the locking from find_agent() into the + agent dialplan function handler to ensure that the agent doesn't + disappear while we're looking at it. ........ + +2008-01-31 15:36 +0000 [r101393] Joshua Colp <jcolp@digium.com> + + * funcs/func_realtime.c: Add missing braces. (closes issue #11886) + Reported by: sergee Patches: func_realtime_fix-r101392.diff + uploaded by sergee (license 138) + +2008-01-31 05:28 +0000 [r101373] Russell Bryant <russell@digium.com> + + * CHANGES: remove entry that is no longer in the tree + +2008-01-30 23:10 +0000 [r101344] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: The deprecation of "username" in favor of + "defaultuser" for SIP peers unfortunately broke realtime + configurations which still used the "username" field. This was + taken care of properly when reading from realtime but was not + handled properly when updating a realtime peer. This change also + adds a deprecation NOTICE CLI message that will print if using + the deprecated "username" field. (closes issue #11880) Reported + by: cabal95 Patches: 11880.patch uploaded by putnopvut (license + 60) Tested by: cabal95 + +2008-01-30 20:08 +0000 [r101322] Olle Johansson <oej@edvina.net> + + * configs/cli.conf.sample: Clarify configuration file that can be + misunderstood + +2008-01-30 19:03 +0000 [r101296] Jason Parker <jparker@digium.com> + + * apps/app_controlplayback.c: Allow disabling the default + ffwd/rewind keys in the ControlPlayback application. This is done + in a backward compat way. If the "default" key for ffwd/rew is + used for another option (such as stop), the "default" is removed. + (closes issue #11754) Reported by: johan Patches: + app_controlplayback.c.option3.patch uploaded by johan (license + 334) Tested by: johan, qwell + +2008-01-30 17:12 +0000 [r101271] Olle Johansson <oej@edvina.net> + + * configs/rtppage.conf.sample (removed), apps/app_rtppage.c + (removed): Removing applications that wasn't ready for svn trunk, + as trunk now has pre-release status. + +2008-01-30 16:54 +0000 [r101269] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Make the VoicemailUsersList AMI command + consistent with other manager list functions. (closes issue + #11874) Reported by: srt Patches: voicemail_ami-11847.patch + uploaded by srt (license 378) + +2008-01-30 16:39 +0000 [r101267-101268] Olle Johansson <oej@edvina.net> + + * include/asterisk/rtp.h, main/rtp.c: - doxygen fixes - change + function to void because it always returned the same value and no + one read it. + + * main/rtp.c: Formatting fixes + +2008-01-30 15:42 +0000 [r101224] Mark Michelson <mmichelson@digium.com> + + * apps/app_rtppage.c: Get trunk to compile + +2008-01-30 15:42 +0000 [r101223] Joshua Colp <jcolp@digium.com> + + * /, main/slinfactory.c: Merged revisions 101222 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101222 | file | 2008-01-30 11:41:04 -0400 (Wed, 30 Jan 2008) | 4 + lines Fix an issue where if a frame of higher sample size + preceeded a frame of lower sample size and ast_slinfactory_read + was called with a sample size of the combined values or higher a + crash would happen. (closes issue #11878) Reported by: stuarth + ........ + +2008-01-30 15:36 +0000 [r101221] Olle Johansson <oej@edvina.net> + + * CHANGES: Update CHANGES with rtppage + +2008-01-30 15:35 +0000 [r101220] Jason Parker <jparker@digium.com> + + * /, configs/extensions.conf.sample: Merged revisions 101219 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11875) ........ r101219 | qwell | 2008-01-30 09:34:37 + -0600 (Wed, 30 Jan 2008) | 5 lines Change default config to use + descending channel order of groups, rather than ascending. Fixes + a potential source of confusion in glare-type situations. Issue + 11875, reported by JimVanM. ........ + +2008-01-30 15:30 +0000 [r101218] Olle Johansson <oej@edvina.net> + + * configs/rtppage.conf.sample (added), apps/app_rtppage.c (added): + Add rtppage() application to do multicast or unicast RTP paging + to SIP phones. (closes issue #11797) Reported by: macbrody + Patches: app_rtppage-20080130.c uploaded by macbrody (license + 352) + +2008-01-30 15:27 +0000 [r101217] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 101216 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan + 2008) | 5 lines Fix a logic error with regards to autofill. Prior + to this change, it was possible for a caller to go out of turn if + autofill were enabled and callers ahead in the queue were + attempting to call a member. This change fixes this. ........ + +2008-01-30 12:48 +0000 [r101196] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: simplify this code and eliminate the return + value cast that is no longer necessary + +2008-01-30 11:27 +0000 [r101153-101154] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, include/asterisk/channel.h: Constifying the + interface to get pvt_ids in the bridge, based on suggestion from + (const char *) Kevin. Thanks! + + * /, channels/chan_sip.c: Merged revisions 101152 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101152 | oej | 2008-01-30 12:20:31 +0100 (Ons, 30 Jan 2008) | 7 + lines Stop musiconhold on attended transfer. (closes issue + #11872) Reported by: gareth Patches: svn-101018.patch uploaded by + gareth (license 208) ........ + +2008-01-30 00:58 +0000 [r101126] Jason Parker <jparker@digium.com> + + * CHANGES: Fix a typo + +2008-01-30 00:04 +0000 [r101082] Russell Bryant <russell@digium.com> + + * CHANGES, apps/app_speech_utils.c: Add the 'n' option to + SpeechBackground, which has the application not answer the + channel if it has not already been answered. (closes SPD-51) + +2008-01-29 23:59 +0000 [r101081] Dwayne M. Hubbard <dhubbard@digium.com> + + * /, build_tools/make_version: Merged revisions 101080 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r101080 | dhubbard | 2008-01-29 17:50:42 -0600 (Tue, 29 + Jan 2008) | 1 line updated build_tools to handle the autotag + directory structure changes; changes related to BE-353. Patch by + The Russell and reviewed by The Me. ........ + +2008-01-29 23:02 +0000 [r101036] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 101035 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r101035 | mmichelson | 2008-01-29 17:02:03 -0600 (Tue, 29 Jan + 2008) | 3 lines Remove a memory leak from updating realtime + queues ........ + +2008-01-29 22:04 +0000 [r101018] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_curl.c: Oops, a sizeof error + +2008-01-29 19:41 +0000 [r100974] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 100973 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100973 | mmichelson | 2008-01-29 13:39:00 -0600 (Tue, 29 Jan + 2008) | 6 lines Fixing an erroneous return value returned when + attempting to pause or unpause a queue member fails. Fixes + BE-366, thanks to John Bigelow for writing the patch. ........ + +2008-01-29 17:58 +0000 [r100935] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 100934 via svnmerge ........ r100934 | file + | 2008-01-29 13:57:05 -0400 (Tue, 29 Jan 2008) | 4 lines Don't + forget to record the channel so we know whether it is bridged or + not later. (closes issue #11811) Reported by: slavon ........ + +2008-01-29 17:44 +0000 [r100933] Russell Bryant <russell@digium.com> + + * /, main/Makefile: Merged revisions 100932 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100932 | russell | 2008-01-29 11:43:41 -0600 (Tue, 29 Jan 2008) + | 4 lines Fix the last couple of issues related to building from + a path that contains spaces. (closes issue #11834) ........ + +2008-01-29 17:42 +0000 [r100931] Jason Parker <jparker@digium.com> + + * /, channels/misdn_config.c: Merged revisions 100930 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r100930 | qwell | 2008-01-29 11:41:43 -0600 (Tue, 29 Jan + 2008) | 6 lines Initialize an array to 0s if config option not + specified. (closes issue #11860) Patches: + misdn_get_config.v1.diff uploaded by IgorG (license 20) ........ + +2008-01-29 17:22 +0000 [r100900-100928] Russell Bryant <russell@digium.com> + + * Makefile, /: Merged revisions 100922 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100922 | russell | 2008-01-29 11:21:33 -0600 (Tue, 29 Jan 2008) + | 3 lines Use GNU make magic instead of shell magic to escape + spaces in the working directory. (related to issue #11834) + ........ + + * Makefile, /: Merged revisions 100882 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100882 | russell | 2008-01-29 11:06:43 -0600 (Tue, 29 Jan 2008) + | 6 lines Fix building Asterisk when the working path has spaces + in it. (closes issue #11834) Reported by: spendergrass Patched + by: me ........ + +2008-01-29 16:14 +0000 [r100843] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, /: Merged revisions 100835 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100835 | qwell | 2008-01-29 10:10:00 -0600 (Tue, 29 Jan 2008) | + 5 lines Allow zap groups above 30 to work properly. (closes issue + #11590) Reported by: tbsky ........ + +2008-01-29 15:30 +0000 [r100833] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Make externip work as documented. If no port + is specified it will use the value of bindport instead of always + being 5060. (closes issue #11858) Reported by: hmodes + +2008-01-29 10:50 +0000 [r100794-100795] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /: Merged revisions 100793 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r100793 | crichter | 2008-01-29 11:36:19 +0100 (Di, 29 + Jan 2008) | 1 line fixed potential segfault in misdn show + channels CLI command ........ + + * channels/chan_misdn.c, /: Merged revisions 96199 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r96199 | crichter | 2008-01-03 13:12:27 +0100 (Do, 03 + Jan 2008) | 1 line make sure frame is completely clean, before we + send it to asterisk as DTMF. If we don't make it clean, it + happens that one way audio occurs.. ........ + +2008-01-29 09:18 +0000 [r100741-100767] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 100740 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100740 | oej | 2008-01-29 09:26:48 +0100 (Tis, 29 Jan 2008) | 8 + lines (closes issue #11736) Reported by: MVF Patches: + bug11736-2.diff uploaded by oej (license 306) Tested by: oej, + MVF, revolution (russellb: This was the showstopper for the + release.) ........ + + * channels/chan_sip.c: Removing code that wasn't supposed to be + there at all, only at an experimental stage before I found + another solution. Thanks Kevin, for reminding me. + +2008-01-28 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0-beta2 released. + +2008-01-28 21:11 +0000 [r100679] Jason Parker <jparker@digium.com> + + * build_tools/menuselect-deps.in, configs/vpb.conf.sample (added), + doc/tex/channelvariables.tex, makeopts.in: Reintroduce more + chan_vpb stuff that was removed in r100421 and r100422 + +2008-01-28 21:07 +0000 [r100678] Mark Michelson <mmichelson@digium.com> + + * channels/chan_vpb.cc (added), configure, + include/asterisk/autoconfig.h.in, configure.ac, + channels/Makefile: Re-inserting chan_vpb into trunk. + +2008-01-28 21:05 +0000 [r100677] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 100675 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100675 | tilghman | 2008-01-28 15:02:02 -0600 (Mon, 28 Jan 2008) + | 2 lines WaitExten didn't handle AbsoluteTimeout properly (went + to 't' instead of 'T') ........ + +2008-01-28 21:02 +0000 [r100676] Jason Parker <jparker@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 100672 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11795) ........ r100672 | qwell | 2008-01-28 14:42:43 + -0600 (Mon, 28 Jan 2008) | 7 lines When using ODBC_STORAGE, make + sure we put greeting files into the database like we do with the + others. Issue #11795 Reported by: dimas Patches: vmgreet.patch + uploaded by dimas (license 88) ........ + +2008-01-28 20:58 +0000 [r100674] Mark Michelson <mmichelson@digium.com> + + * /: Blocked revisions 100673 via svnmerge ........ r100673 | + mmichelson | 2008-01-28 14:55:56 -0600 (Mon, 28 Jan 2008) | 3 + lines Undoing the deprecation of chan_vpb. It is alive and well. + ........ + +2008-01-28 20:40 +0000 [r100632-100671] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Fix up some T38 state change issues. (closes + issue #11630) Reported by: dimas Patches: v2-sip-t38state.patch + uploaded by dimas (license 88) + + * channels/chan_sip.c: Fix up two scheduling issues. In one + instance a scheduled item was not deleted when it should have + been and in the other it was scheduled again when it shouldn't + have been. + +2008-01-28 18:41 +0000 [r100630-100631] Russell Bryant <russell@digium.com> + + * main/features.c: Merge rev 100626 from Asterisk 1.4. The svnmerge + of this commit was a NoOp, since res_features doesn't exist in + trunk. Thanks to qwell for pointing it out! + + * /, channels/chan_sip.c: Merged revisions 100629 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008) + | 5 lines For some reason, the use of this strdupa() is leading + to memory corruption on freebsd sparc64. This trivial workaround + fixes it. (closes issue #10300, closes issue #11857, reported by + mattias04 and Home-of-the-Brave) ........ + +2008-01-28 18:27 +0000 [r100628] Tilghman Lesher <tlesher@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/logger.c: Normalize the detection for execinfo, so that + Linux (glibc) and other platforms with libexecinfo will generate + inline stack backtraces correctly. + +2008-01-28 18:27 +0000 [r100627] Russell Bryant <russell@digium.com> + + * /: Merged revisions 100626 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100626 | russell | 2008-01-28 12:26:31 -0600 (Mon, 28 Jan 2008) + | 7 lines Fix a crash in ast_masq_park_call() (issue #11342) + Reported by: DEA Patches: res_features-park.txt uploaded by DEA + (license 3) ........ + +2008-01-28 18:24 +0000 [r100625] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, /: Merged revisions 100624 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100624 | qwell | 2008-01-28 12:23:09 -0600 (Mon, 28 Jan 2008) | + 1 line Correct a comment which made little/no sense. ........ + +2008-01-28 17:21 +0000 [r100565-100582] Russell Bryant <russell@digium.com> + + * main/channel.c, channels/chan_local.c, /, + include/asterisk/channel.h: Merged revisions 100581 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 + Jan 2008) | 9 lines Make some deadlock related fixes. These bugs + were discovered and reported internally at Digium by Steve Pitts. + - Fix up chan_local to ensure that the channel lock is held + before the local pvt lock. - Don't hold the channel lock when + executing the timing function, as it can cause a deadlock when + using chan_local. This actually changes the code back to be how + it was before the change for issue #10765. But, I added some + other locking that I think will prevent the problem reported + there, as well. ........ + + * main/pbx.c: Clean up some formatting, and simplify a bit of code + using ast_str + +2008-01-28 13:57 +0000 [r100549] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Don't do a network byte order conversion + when setting the socket's port variable to that of bindaddr's. It + is already in the correct network byte order. (closes issue + #11800) Reported by: hmodes + +2008-01-28 04:43 +0000 [r100514-100533] Russell Bryant <russell@digium.com> + + * main/channel.c: Make a couple more uses of ARRAY_LEN, and convert + some spaces to tabs + + * main/channel.c: - Simplify a line with ARRAY_LEN() - Make a few + little formatting changes + + * main/channel.c: These readlocks always fail for me on my mac, and + I saw it happen again today on another mac. We ignore the return + value of locking operations almost everywhere in Asterisk. So, + ignore these, as well, so Asterisk will actually work on systems + where this is occurring while I look into what the issue is. + +2008-01-27 23:14 +0000 [r100488-100497] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c, include/asterisk/sched.h, + channels/chan_iax2.c: With the switch to the ast_sched_replace* + API in trunk, we lose the correction that was just merged from + 1.4, so this is a changeover to those APIs to use the macro + versions, so that we properly detect errors from ast_sched_del, + instead of simply ignoring the return values. + + * main/cdr.c, channels/chan_misdn.c, main/dnsmgr.c, /, + channels/chan_sip.c, channels/chan_h323.c, + include/asterisk/sched.h, main/file.c, pbx/pbx_dundi.c, + channels/chan_iax2.c, main/rtp.c, channels/chan_mgcp.c: Merged + revisions 100465 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008) + | 11 lines When deleting a task from the scheduler, ignoring the + return value could possibly cause memory to be accessed after it + is freed, which causes all sorts of random memory corruption. + Instead, if a deletion fails, wait a bit and try again (noting + that another thread could change our taskid value). (closes issue + #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt + uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan, + stuarth` ........ + +2008-01-25 22:54 +0000 [r100421-100422] Jason Parker <jparker@digium.com> + + * doc/tex/channelvariables.tex: Get rid of that last little bit. + + * build_tools/menuselect-deps.in, configs/vpb.conf.sample + (removed), makeopts.in: Remove more remnants of chan_vpb + +2008-01-25 22:39 +0000 [r100419-100420] Mark Michelson <mmichelson@digium.com> + + * channels/chan_vpb.cc (removed), configure, + include/asterisk/autoconfig.h.in, configure.ac, + channels/Makefile, .cleancount: Removing chan_vpb from the tree + + * /: Blocked revisions 100418 via svnmerge ........ r100418 | + mmichelson | 2008-01-25 16:32:41 -0600 (Fri, 25 Jan 2008) | 4 + lines Deprecating chan_vpb. It is now preferred that users of + Voicetronix products use chan_zap in combination with their + zaptel drivers. ........ + +2008-01-25 21:26 +0000 [r100379] Jason Parker <jparker@digium.com> + + * /, channels/chan_sip.c: Merged revisions 100378 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100378 | qwell | 2008-01-25 15:24:49 -0600 (Fri, 25 Jan 2008) | + 2 lines This would have never been true, since we're passing + (sizeof(req.data) - 1) as the len to recvfrom(). ........ + +2008-01-25 20:51 +0000 [r100361] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_rpt.c: correct a real problem and silence an annoying + compiler warning + +2008-01-25 14:53 +0000 [r100344] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Insure that we are not going to pass a NULL + pointer to add_to_interfaces. (closes issue #11840) Reported by: + junky + +2008-01-25 02:52 +0000 [r100325] Joshua Colp <jcolp@digium.com> + + * main/dial.c, include/asterisk/dial.h: Add an API call that steals + the answered channel so that a destruction of the dialing + structure does not hang it up. + +2008-01-24 22:58 +0000 [r100307] Tilghman Lesher <tlesher@digium.com> + + * Makefile, build_tools/make_defaults_h: Use the set ASTDBDIR as + the default, too + +2008-01-24 22:36 +0000 [r100305-100306] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/app.h: ummm... might be good if this macro + argument was actually used :-) + + * include/asterisk/app.h: add the ability to define a structure + type for argument parsing when it would be useful to be able to + pass it between functions + +2008-01-24 22:02 +0000 [r100266] James Golovich <james@gnuinter.net> + + * channels/chan_sip.c: Fix simple whitespace issue + +2008-01-24 22:01 +0000 [r100265] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/app.h, /: Merged revisions 100264 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r100264 | kpfleming | 2008-01-24 15:57:41 -0600 (Thu, 24 + Jan 2008) | 2 lines make these macros not assume that the only + other field in the structure is 'argc'... this is true when + someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable + to define your own structure as long as it has the right fields + ........ + +2008-01-24 20:32 +0000 [r100245] Joshua Colp <jcolp@digium.com> + + * main/features.c: Minor cosmetic change... + +2008-01-24 18:35 +0000 [r100224] James Golovich <james@gnuinter.net> + + * main/astmm.c: Increase the size of filenames stored when astmm is + used. If the path length was long they would be truncated and + grouped together with whatever matches + +2008-01-24 17:47 +0000 [r100206] Joshua Colp <jcolp@digium.com> + + * configs/rtp.conf.sample, CHANGES, main/rtp.c: Merge in strictrtp + branch. This adds a strictrtp option to rtp.conf which drops + packets that do not come from the remote party. (closes issue + #8952) Reported by: amorsen + +2008-01-24 17:24 +0000 [r100169] Russell Bryant <russell@digium.com> + + * /, main/asterisk.c: Merged revisions 100164 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100164 | russell | 2008-01-24 11:22:09 -0600 (Thu, 24 Jan 2008) + | 2 lines Update main Asterisk copyright info to 2008 ........ + +2008-01-24 16:47 +0000 [r100121-100139] Jason Parker <jparker@digium.com> + + * /, res/res_phoneprov.c, main/acl.c: Merged revisions 100138 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r100138 | qwell | 2008-01-24 10:41:29 -0600 (Thu, 24 Jan 2008) | + 6 lines Fix compilation on Solaris. (closes issue #11832) + Patches: bug-11832.diff uploaded by snuffy (license 35) ........ + + * channels/chan_sip.c, main/features.c: Move chan_local dependency + into places (only one) that previously depended on res_features, + and used local channels + +2008-01-24 15:54 +0000 [r100076-100112] Joshua Colp <jcolp@digium.com> + + * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c, + channels/chan_mgcp.c: Remove dependency on res_features from some + channel drivers. It is now part of the core and no longer exists + as a module. + + * main/channel.c: Some more cosmetic changes. + + * main/channel.c: Add some spacing. + + * main/dial.c: Test hopefully over. + + * main/dial.c: Testing something... + +2008-01-24 00:04 +0000 [r100057] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: fix flag bit definitions to make code from + issue #11049 actually work; along the way, clarify comments and + add some dummy flag definitions for other multi-bit flags to + hopefully stop this from happening in the future (closes issue + #11049) + +2008-01-23 23:09 +0000 [r100039] Jason Parker <jparker@digium.com> + + * res/res_features.c (removed), main/Makefile, main/features.c + (added), include/asterisk/_private.h, CHANGES, .cleancount, + main/asterisk.c, main/loader.c, include/asterisk/features.h: Move + code from res_features into (new file) main/features.c + +2008-01-23 22:00 +0000 [r100021] Russell Bryant <russell@digium.com> + + * CREDITS: Add Sergey Tamkovich to CREDITS. Thank you for your + contributions! + +2008-01-23 21:11 +0000 [r99979-99980] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 99978 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99978 | oej | 2008-01-23 22:07:16 +0100 (Ons, 23 Jan 2008) | 7 + lines Second attempt. Don't change invitestate when receiving 18x + messages in CANCEL state. (issue #11736) Reported by: MVF Patch + by oej. ........ + + * /, channels/chan_sip.c: Merged revisions 99977 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9 + lines Make sure we don't cancel destruction on calls in CANCEL + state, even if we get 183 while waiting for answer on our CANCEL. + (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by + oej (license 306) Tested by: MVF ........ + +2008-01-23 20:26 +0000 [r99976] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_externalivr.c: Merged revisions 99975 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r99975 | mmichelson | 2008-01-23 14:25:00 -0600 (Wed, 23 + Jan 2008) | 3 lines Fixing a typo. ........ + +2008-01-23 17:48 +0000 [r99922-99924] Russell Bryant <russell@digium.com> + + * /, apps/app_chanspy.c: Merged revisions 99923 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) | + 8 lines ChanSpy issues a beep when it starts at the beginning of + a list of channels to potentially spy on. However, if there were + no matching channels, it would beep at you over and over, which + is pretty annoying. Now, it will only beep once in the case that + there are no channels to spy on, but it will still beep again + once it reaches the beginning of the channel list again. (closes + issue #11738, patched by me) ........ + + * main/tcptls.c: Fix tcptls build when openssl isn't installed + (closes issue #11813) Reported by: tzafrir Patches: + asterisk-tcptls.diff.txt uploaded by jamesgolovich (license 176) + +2008-01-23 17:27 +0000 [r99920] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c: since echo canceler parameters in Zaptel are + now signed integers, allow them during parsing + +2008-01-23 16:21 +0000 [r99879] Mark Michelson <mmichelson@digium.com> + + * /: Blocked revisions 99878 via svnmerge ........ r99878 | + mmichelson | 2008-01-23 10:18:04 -0600 (Wed, 23 Jan 2008) | 4 + lines These flag tests were illogical. They were testing sip_peer + flags on a sip_pvt. Thanks to Russell for helping to get this odd + problem figured out. ........ + +2008-01-23 15:23 +0000 [r99860] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_h323.c: Progress messages don't work (closes issue + #10497) Reported by: pj Patches: h323-announces-r99483.diff + uploaded by sergee (license 138) Tested by: pj + +2008-01-23 10:18 +0000 [r99839] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: - Add a few comments to sip_xmit - Make sure + that we are aware of a pending INVITE even if we're using TCP + +2008-01-23 05:29 +0000 [r99696-99818] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Coding guidelines fixups + + * /, apps/app_voicemail.c: Merged revisions 99777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008) + | 8 lines When we reset the password via an external command, we + should also reset the password stored in the in-memory list, too + (otherwise it doesn't really take effect). (closes issue #11809) + Reported by: davetroy Patches: fix_externpass.diff uploaded by + davetroy (license 384) ........ + + * /, res/res_odbc.c: Merged revisions 99775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99775 | tilghman | 2008-01-22 22:20:15 -0600 (Tue, 22 Jan 2008) + | 2 lines Oops, should have checked for a NULL obj, here, too + ........ + + * res/res_config_ldap.c: Coding guidelines cleanup + + * /, main/acl.c: Merged revisions 99718 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99718 | tilghman | 2008-01-22 18:56:06 -0600 (Tue, 22 Jan 2008) + | 2 lines Just confirmed that all current platforms need this + header file ........ + + * /: Oops + + * /, build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, doc/ldap.txt (added), + configure.ac, configs/res_ldap.conf.sample (added), + res/res_config_ldap.c (added), CHANGES, makeopts.in, + contrib/scripts/asterisk.ldap-schema (added), + contrib/scripts/asterisk.ldif (added): Add res_config_ldap for + realtime LDAP engine. (closes issue #5768) Reported by: mguesdon + Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon + (license 121) res_ldap.conf.sample uploaded by suretec (license + 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70) + asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested + by: oej, mguesdon, suretec, cthorner + +2008-01-22 21:09 +0000 [r99647-99653] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 99652 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4 + lines Thanks to Russell's education I realize that BUFSIZ has + changed since I learned the C language over 20 years ago... + Resetting chan_sip to the size of BUFSIZ that I expected in my + old head to avoid too heavy memory allocations on some systems. + ........ + + * doc/tex/channelvariables.tex, CHANGES: Documentation updates for + BRIDGEPVTCALLID + +2008-01-22 20:42 +0000 [r99646] Tilghman Lesher <tlesher@digium.com> + + * /, main/acl.c: Merged revisions 99643 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99643 | tilghman | 2008-01-22 14:34:55 -0600 (Tue, 22 Jan 2008) + | 2 lines Fix the defines for OS X (and Solaris, too) ........ + +2008-01-22 20:41 +0000 [r99645] Russell Bryant <russell@digium.com> + + * main/asterisk.c: Make sure the command is not just present but is + also configured to be executed + +2008-01-22 20:35 +0000 [r99644] Olle Johansson <oej@edvina.net> + + * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h: + Add a generic function to set the bridged call PVT unique id + string as a channel variable BRIDGEPVTCALLID This is important + for call tracing in log files and CDRs, so that the SIP callID + can be traced along servers. The CHANNEL dialplan function won't + work here, since the outbound channel is gone when we need the + Call-ID. Other channel drivers may now implement the same + function :-), but this patch only supports chan_sip.so. Inspired + by (issue #11816) Reported by: ctooley Patch by oej + +2008-01-22 20:33 +0000 [r99642] Russell Bryant <russell@digium.com> + + * configs/cli.conf.sample (added), CHANGES, main/asterisk.c: Change + the Asterisk CLI startup commands feature to read commands to run + from cli.conf after a discussion on the -dev list. + +2008-01-22 17:46 +0000 [r99595-99596] Olle Johansson <oej@edvina.net> + + * channels/chan_local.c, /, res/res_features.c, + channels/chan_agent.c, apps/app_followme.c: Merged revisions + 99594 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3 + lines Add more dependencies on chan_local and add a note to the + description of chan_local so that people don't disable it in + menuselect just to clean up. ........ + + * apps/app_dial.c, /: Merged revisions 99592 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5 + lines Add dependency on chan_local to app_dial. Dial still runs + without chan_local, but will be missing forwarding functionality. + ........ + +2008-01-22 17:15 +0000 [r99559] Tilghman Lesher <tlesher@digium.com> + + * /, main/acl.c: Merged revisions 99540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008) + | 7 lines Ensure that we can get an address even when we don't + have a default route. (closes issue #9225) Reported by: junky + Patches: 20080122__bug9225.diff.txt uploaded by Corydon76 + (license 14) Tested by: oej, loloski, sergee ........ + +2008-01-22 16:55 +0000 [r99542] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Point out a bug in some debug counter + handling + +2008-01-22 15:25 +0000 [r99464-99521] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Add authentication options to the SIP + dialstring. Documentation follows separately (issue #11587) + Reported by: sobomax Patches: chan_sip.c-trunk.diff uploaded by + sobomax (license 359) + + * configs/sip.conf.sample: Documentation updates + + * doc/siptls.txt: Small fixes + + * main/tcptls.c, channels/chan_zap.c, main/abstract_jb.c, + include/asterisk/tcptls.h: Doxygen updates + +2008-01-21 23:56 +0000 [r99427] Mark Michelson <mmichelson@digium.com> + + * channels/chan_local.c, /: Merged revisions 99426 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21 + Jan 2008) | 12 lines Fixing an issue wherein monitoring local + channels was not possible. During a channel masquerade, the + monitors on the two channels involved are swapped. In 99% of the + cases this results in the desired effect. However, if monitoring + a local channel, this caused the monitor which was on the local + channel to get moved onto a channel which is immediately hung up + after the masquerade has completed. By swapping the monitors + prior to the masquerade, we avoid the problem by tricking the + masquerade into placing the monitor back onto the channel where + we want it. During the investigation of the issue, the channel's + monitor was the only thing that was swapped in such a manner + which did not make sense to have done. All other variable + swapping made sense. ........ + +2008-01-21 23:25 +0000 [r99424] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c: Fix distinctive ring detection. Reported by: + milazzo Patches: drings.diff uploaded by milazzo (license 383) + Closes issue #11799 + +2008-01-21 22:32 +0000 [r99406] Mark Michelson <mmichelson@digium.com> + + * configs/queues.conf.sample, apps/app_queue.c: Adding the + QUEUENAME variable to the variables set using the setqueuevar + option in queues.conf. Suggestion comes from Shaun2222 on IRC. + +2008-01-21 21:11 +0000 [r99382-99384] Olle Johansson <oej@edvina.net> + + * channels/chan_console.c: Remove compiler warning for + uninitialized variable + + * channels/chan_sip.c: Doxygen updates. The TCP/TLS code was + committed without any doxygen obviously. Tss tss. + + * channels/chan_sip.c: Updating doxygen + +2008-01-21 18:15 +0000 [r99350] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/res_odbc.h, /, res/res_odbc.c, + configs/res_odbc.conf.sample: Merged revisions 99341 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21 + Jan 2008) | 8 lines Permit the user to specify number of seconds + that a connection may remain idle, which fixes a crash on + reconnect with the MyODBC driver. (closes issue #11798) Reported + by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt + uploaded by Corydon76 (license 14) Tested by: mvanbaak ........ + +2008-01-21 16:02 +0000 [r99302] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 99301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4 + lines Bump the buffer size for Via headers up to 512. There are + some exceptionally large Via headers out there. (closes issue + #11783) Reported by: ofirroval ........ + +2008-01-21 07:02 +0000 [r99280] Olle Johansson <oej@edvina.net> + + * CREDITS: Update + +2008-01-21 03:54 +0000 [r99265] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Change over to using ast_debug so these + debug messages don't always show up. + +2008-01-20 07:28 +0000 [r99166-99248] Russell Bryant <russell@digium.com> + + * channels/chan_console.c: Add a "console active" CLI command, + which lets you find out which console device is currently active + for the Asterisk CLI, or to set it. Also, knock multiple device + support off of the to-do list. + + * configs/console.conf.sample: correct the name of a CLI command + for getting available device names + + * configs/console.conf.sample, channels/chan_console.c: Merge + changes from team/russell/console_devices - Add support for + multiple devices. All devices are configured in console.conf. - + Add "console list devices" CLI command to show configured + devices. Also, changed the old "list devices" to be "list + available", which queries PortAudio for all audio devices that + are available for use. + + * /, main/slinfactory.c: Merged revisions 99187 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) | + 4 lines Fix a couple of memory leaks with frame handling. + Specifically, ast_frame_free() needed to be called on the frame + that came from the translator to signed linear. ........ + + * README: Add Cygwin as an "other" platform that is supported + + * README: Various README updates + +2008-01-18 22:58 +0000 [r99128] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 99127 via svnmerge ........ r99127 | file | + 2008-01-18 18:57:15 -0400 (Fri, 18 Jan 2008) | 2 lines Remove the + __ in front of the unused variable. This causes some compilers to + freak out. ........ + +2008-01-18 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0-beta1 released. + +2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell@digium.com> + + * CREDITS, include/asterisk/http.h, main/tcptls.c (added), + main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), + main/Makefile, main/http.c, include/asterisk/tcptls.h (added), + configs/sip.conf.sample, CHANGES: Merge changes from + team/group/sip-tcptls This set of changes introduces TCP and TLS + support for chan_sip. There are various new options in + configs/sip.conf.sample that are used to enable these features. + Also, there is a document, doc/siptls.txt that describes some + things in more detail. This code was implemented by Brett Bryant + and James Golovich. It was reviewed by Joshua Colp and myself. A + number of other people participated in the testing of this code, + but since it was done outside of the bug tracker, I do not have + their names. If you were one of them, thanks a lot for the help! + (closes issue #4903, but with completely different code that what + exists there.) + + * main/frame.c, /, include/asterisk/translate.h: Merged revisions + 99081 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) | + 9 lines Revert adding the packed attribute, as it really doesn't + make sense why that would do any good. Fix the real bug, which is + to do the check to see if the frame came from a translator at the + beginning of ast_frame_free(), instead of at the end. This + ensures that it always gets checked, even if none of the parts of + the frame are malloc'd, and also ensures that we aren't looking + at free'd memory in the case that it is a malloc'd frame. (closes + issue #11792, reported by explidous, patched by me) ........ + + * /, include/asterisk/translate.h: Merged revisions 99079 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) | + 4 lines Since we're relying on the offset between the frame and + the beginning of the translator pvt struct, set the packed + attribute to make sure we get to the right place. (potential fix + for issue #11792) ........ + +2008-01-18 16:58 +0000 [r99026] Terry Wilson <twilson@digium.com> + + * res/res_features.c: This should at least temporarily fix a + problem where the 't' Dial option is incorrectly passed to the + transferee when built-in attended transfers are used. There is + still a problem with 'T', but better to fix some problems than no + problems while we work on it. (closes issue #7904) Reported by: + k-egg Patches: transfer-fix-trunk-r97657.diff uploaded by sergee + (license 138) Tested by: sergee, otherwiseguy + +2008-01-18 06:58 +0000 [r99015-99018] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_odbc.c: Convert func_odbc to use SQLExecDirect for + speed (closes issue #10723) Reported by: mnicholson Patches: + func-odbc-direct-execute1.diff uploaded by mnicholson (license + 96) Tested by: Corydon76, mnicholson, falves11 + + * res/res_odbc.c: Permit username and password to be NULL (which + enables pass-through from the layer above). Reported by: lurcher + Patch by: tilghman (Closes issue #11739) + + * funcs/func_cut.c: Reset default CUT delimiter back to '-' + +2008-01-17 23:28 +0000 [r99006-99011] Russell Bryant <russell@digium.com> + + * channels/chan_console.c: Make the output of "console list + devices" a bit prettier. + + * channels/chan_console.c: List which devices are inputs and + outputs in "console list devices" + + * main/channel.c: Add AST_FORMAT_SLINEAR16 to the list for + ast_best_codec() + + * main/frame.c, /, channels/chan_iax2.c, include/asterisk/frame.h: + Merged revisions 99004 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | + 10 lines Have IAX2 optimize the codec translation path just like + chan_sip does it. If the caller's codec is in our codec list, + move it to the top to avoid transcoding. (closes issue #10500) + Reported by: stevedavies Patches: iax-prefer-current-codec.patch + uploaded by stevedavies (license 184) + iax-prefer-current-codec.1.4.patch uploaded by stevedavies + (license 184) Tested by: stevedavies, pj, sheldonh ........ + +2008-01-17 22:22 +0000 [r99002] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Fixing trunk IMAP build (closes issue + #11788) Reported by: DEA Patches: vm-imap-build-fix.txt uploaded + by DEA (license 3) + +2008-01-17 20:51 +0000 [r98998] Jason Parker <jparker@digium.com> + + * Makefile, build_tools/cflags.xml, channels/chan_zap.c, + main/dsp.c, configs/zapata.conf.sample: Add several busy + detection related defines to menuselect. Allow better busy detect + debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and + BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches: + busydetect_enhancement.patch uploaded by agx (license 298) + busydetect-r94975.diff uploaded by sergee (license 138) + Additional changes/cleanup by me. + +2008-01-17 16:33 +0000 [r98993-98994] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: state_interface could be NULL, so use the + never-NULL cur->state_interface for this check + + * apps/app_queue.c: Get the device state of the state interface + instead of the interface when creating a new queue member. Thanks + to Atis Lezdins for bringing this up on the Asterisk-Dev mailing + list. + +2008-01-17 16:21 +0000 [r98992] Jason Parker <jparker@digium.com> + + * /, configs/zapata.conf.sample: Merged revisions 98991 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes + issue #11784) ........ r98991 | qwell | 2008-01-17 10:19:46 -0600 + (Thu, 17 Jan 2008) | 4 lines Add a clarification about the + immediate= option of zapata.conf Issue 11784, patch by klaus3000. + ........ + +2008-01-17 16:17 +0000 [r98989-98990] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, configs/zapata.conf.sample: major + reliability and performance improvement in VWMI monitoring for + FXO ports (code by markster, me and dbailey) + + * res/res_config_curl.c: resolve (valid) compiler warning about + variable that could be used before being initialized + +2008-01-17 03:09 +0000 [r98988] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c, doc/tex/phoneprov.tex, + configs/phoneprov.conf.sample: Update res_phoneprov to default to + setting the SERVER variable to the IP the HTTP request for the + config came in on and the SERVER_PORT to the bindport setting in + sip.conf. I've left in the ability to override these options, + because I can't always guess how someone might decide to do + something weird with what is available to them--although needing + to is pretty unlikely. Documentation was updated to reflect + preference for not setting serveraddr, serveriface, or + serverport. Tested on Linux and OS X. + +2008-01-17 00:13 +0000 [r98987] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_adaptive_odbc.c: Change the way the new filter feature + works, by allowing it to be a column NOT logged into the + database. This will allow more granularity of a decision + evaluated in the dialplan, then takes effect when posting the + CDR. + +2008-01-17 00:05 +0000 [r98986] Russell Bryant <russell@digium.com> + + * CHANGES, main/asterisk.c: Add support for an easy way to + automatically execute some Asterisk CLI commands immediately at + startup. Any commands in the startup_commands file in the + Asterisk config diretory will get executed. (closes issue #11781) + Reported by: jamesgolovich Patches: asterisk-startupcmds.diff.txt + uploaded by jamesgolovich (license 176) -- With some changes by + me. + +2008-01-16 23:08 +0000 [r98985] Jason Parker <jparker@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + acinclude.m4: Change AST_EXT_TOOL_CHECK to attempt to build + against <package>_LIB, per recommendations from Russell. + +2008-01-16 22:36 +0000 [r98984] Tilghman Lesher <tlesher@digium.com> + + * CHANGES: Info about res_config_curl + +2008-01-16 22:36 +0000 [r98983] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 98982 via svnmerge ........ r98982 | russell + | 2008-01-16 16:36:24 -0600 (Wed, 16 Jan 2008) | 5 lines Add an + unused pointer to the ast_channel struct. This makes the + ast_channel structure retain the same size as it had in previous + 1.4 releases. Also, all of the offsets for members in the + structure are still the same (except for the two pointers that + got replaced for the new spy/whisper architecture.) ........ + +2008-01-16 22:20 +0000 [r98981] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_curl.c (added), main/utils.c: New module + res_config_curl (closes issue #11747) Reported by: Corydon76 + Patches: res_config_curl.c uploaded by Corydon76 (license 14) + 20080116__bug11747.diff.txt uploaded by Corydon76 (license 14) + Tested by: jmls + +2008-01-16 21:53 +0000 [r98978] Russell Bryant <russell@digium.com> + + * CREDITS, channels/chan_sip.c, configs/sip.conf.sample: Merge the + changes from issue #10665 from the team/group/sip_session_timers + branch. This set of changes introduces SIP session timers support + (RFC 4028). In short, this prevents stuck SIP sessions that were + not properly torn down due to network or endpoint failures during + an established SIP session. To quote some of the documentation + supplied with the patch: "The SIP Session-Timers is an extension + of the SIP protocol that allows end-points and proxies to refresh + a session periodically. The sessions are kept alive by sending a + RE-INVITE or UPDATE request at a negotiated interval. If a + session refresh fails then all the entities that support Session- + Timers clear their internal session state. In addition, UAs + generate a BYE request in order to clear the state in the proxies + and the remote UA (this is done for the benefit of SIP entities + in the path that do not support Session-Timers)." (closes issue + #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by + rjain (license 226) chan_sip.c.diff uploaded by rjain (license + 226) sip.conf.sample.diff uploaded by rjain (license 226) + proc_422_rsp_comment.diff uploaded by rjain (license 226) + chan_sip.c.cache.diff uploaded by rjain (license 226) + chan_sip.memalloc uploaded by rjain (license 226) + chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches + tracked in team/group/sip_session_timers, with some additional + fixes by russell and oej. Tested by: jtodd, rjain, loloski + +2008-01-16 20:36 +0000 [r98974-98975] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 98973 via svnmerge ........ r98973 | file | + 2008-01-16 16:34:30 -0400 (Wed, 16 Jan 2008) | 2 lines Bump up + cleancount due to previous commit that changed the channel + structure. ........ + + * /: Blocked revisions 98972 via svnmerge ........ r98972 | file | + 2008-01-16 16:33:47 -0400 (Wed, 16 Jan 2008) | 2 lines Replace + current spy architecture with backport of audiohooks. This should + take care of current known spy issues. ........ + +2008-01-16 19:41 +0000 [r98968-98971] Jason Parker <jparker@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac: + Partially revert r93898, because it broke the way netsnmp was + being detected. rizzo, do you want to discuss so we can rethink + this, or do you have another way? + + * CHANGES: Add note about new update.log to CHANGES, by request of + jmls and further prodding by jsmith. + + * Makefile, /: Add logging for 'make update' command (also fixes + updates in some places). Issue #11766, initial patch by jmls. + +2008-01-16 17:51 +0000 [r98967] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 98966 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6 + lines Add missing NULLs at end of two ast_load_realtimes. (closes + issue #11769) Reported by: tequ Patches: chaniax.patch uploaded + by dimas (license 88) ........ + +2008-01-16 17:21 +0000 [r98965] Mark Michelson <mmichelson@digium.com> + + * channels/chan_local.c, /: Merged revisions 98964 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 + Jan 2008) | 10 lines Fix a deadlock in chan_local in + local_hangup. There was contention because the local_pvt was held + and it was attempting to lock a channel, which is the incorrect + locking order. (closes issue #11730) Reported by: UDI-Doug + Patches: 11730.patch uploaded by putnopvut (license 60) Tested + by: UDI-Doug ........ + +2008-01-16 16:06 +0000 [r98962] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c: Make users list static + +2008-01-16 15:09 +0000 [r98954-98961] Joshua Colp <jcolp@digium.com> + + * main/dial.c, /: Merged revisions 98960 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6 + lines Introduce a lock into the dialing API that protects it when + destroying the structure. (closes issue #11687) Reported by: + callguy Patches: 11687.diff uploaded by file (license 11) + ........ + + * /, main/rtp.c: Merged revisions 98958 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4 + lines Add two more SDP names for ulaw and alaw. (closes issue + #11777) Reported by: tootai ........ + + * /, channels/chan_sip.c: Merged revisions 98955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6 + lines Don't drop the old record route information when dealing + with packets related to a reinvite. (closes issue #11545) + Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by + kebl0155 (license 356) ........ + + * channels/chan_sip.c: Remove DNS lookup from sip_devicestate. This + seems to come from way back when and I can't think of a reason + for it being here, plus it could cause needless DNS lookups. + (closes issue #10983) Reported by: jtodd + +2008-01-16 01:35 +0000 [r98953] Steve Murphy <murf@digium.com> + + * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Terry found + this problem with running the expr2 parser on OSX. Make the + #defines come out the same between the parser & lexer. + +2008-01-16 01:17 +0000 [r98952] Joshua Colp <jcolp@digium.com> + + * /, build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, codecs/codec_speex.c, + configure.ac, makeopts.in: Merged revisions 98951 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan + 2008) | 4 lines Add autoconf logic for speexdsp. Later versions + use a separate library for some things so we need to use it if + present in codec_speex. (closes issue #11693) Reported by: yzg + ........ + +2008-01-15 23:53 +0000 [r98948] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 98946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | + 11 lines Change a buffer in check_auth() to be a thread local + dynamically allocated buffer, instead of a massive buffer on the + stack. This fixes a crash reported by Qwell due to running out of + stack space when building with LOW_MEMORY defined. On a very + related note, the usage of BUFSIZ in various places in chan_sip + is arbitrary and careless. BUFSIZ is a system specific define. On + my machine, it is 8192, but by definition (according to google) + could be as small as 256. So, this buffer in check_auth was 16 + kB. We don't even support SIP messages larger than 4 kB! Further + usage of this define should be avoided, unless it is used in the + proper context. ........ + +2008-01-15 23:52 +0000 [r98947] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample: + Add the "filter" keyword + +2008-01-15 23:35 +0000 [r98944-98945] Russell Bryant <russell@digium.com> + + * main/translate.c, include/asterisk/translate.h: Clean up + something I did for ABI compatability in 1.4 + + * main/frame.c, /, main/translate.c, main/abstract_jb.c, + channels/chan_iax2.c, codecs/codec_zap.c, + include/asterisk/frame.h, main/rtp.c, + include/asterisk/translate.h: Merged revisions 98943 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 + Jan 2008) | 25 lines Commit a fix for some memory access errors + pointed out by the valgrind2.txt output on issue #11698. The + issue here is that it is possible for an instance of a translator + to get destroyed while the frame allocated as a part of the + translator is still being processed. Specifically, this is + possible anywhere between a call to ast_read() and + ast_frame_free(), which is _a lot_ of places in the code. The + reason this happens is that the channel might get masqueraded + during this time. During a masquerade, existing translation paths + get destroyed. So, this patch fixes the issue in an API and ABI + compatible way. (This one is for you, paravoid!) It changes an + int in ast_frame to be used as flag bits. The 1 bit is still used + to indicate that the frame contains timing information. Also, a + second flag has been added to indicate that the frame came from a + translator. When a frame with this flag gets released and has + this flag, a function is called in translate.c to let it know + that this frame is doing being processed. At this point, the flag + gets cleared. Also, if the translator was requested to be + destroyed while its internal frame still had this flag set, its + destruction has been deffered until it finds out that the frame + is no longer being processed. Admittedly, this feels like a hack. + But, it does fix the issue, and I was not able to think of a + better solution ... ........ + +2008-01-15 20:10 +0000 [r98895-98935] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 98934 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4 + lines Based on the boundary found move over the correct amount. + (closes issue #11750) Reported by: tasker ........ + + * /, channels/chan_sip.c: Merged revisions 98894 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4 + lines Accept "; boundary=" not just ";boundary=" in the multipart + mixed content type. (closes issue #11750) Reported by: tasker + ........ + +2008-01-14 22:19 +0000 [r98889] Jason Parker <jparker@digium.com> + + * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add + backupdeleted option to app_voicemail (closes issue #10740) + Reported by: ruffle Patches: app_voicemail.diff uploaded by + ruffle (license 201) 10740-voicemail.diff uploaded by qwell + (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak + (license 7) Tested by: blitzrage, mvanbaak, qwell + +2008-01-14 22:11 +0000 [r98850-98888] Mark Michelson <mmichelson@digium.com> + + * apps/app_directory.c: Big improvement for app_directory. This + patch breaks the do_directory function up so that it is more + easily parsed by the human brain. It also fixes some errors. I'll + quote dimas from the original bug description: "app_directory + contained some duplicate code even before addition of 'm' option. + Addition of that option doubled amount of that code. Worst of + all, there are minor differences between these code block and + bugs caused by these differences. 1. There is a memory leak. In + the 'menu' mode, result of the convert(pos) function is not freed + while it should be. 2. In the 'menu' mode check for + OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result, + application works in the mode opposite to what user expect + (checking last name when user wants the first nd vice versa). 3. + select_item function plays message for user using res = func1() + || func2() || func3()... construct. This construct loses the + actual value returned by ast_waitstream() for example so at the + end, res does not contain digit user dialed while listening to + the message. 4. (also in 1.4) application announces entries from + voicemail.conf/realtime separately from entries from users.conf. + I see no reason why doing so instead of building combined list. + 5. Alot of duplicated code as already mentioned." This was tested + by dimas and I (I tested under valgrind). A word of caution: any + bug fixes that happen in app_directory in 1.4 will almost + certainly not merge cleanly into trunk as a result of this, but + it is well worth it. Huge thanks to dimas for this wonderful + submission. (closes issue #11744) Reported by: dimas Patches: + dir3.patch uploaded by dimas (license 88) Tested by: putnopvut, + dimas + + * /: Blocked revisions 98849 via svnmerge ........ r98849 | + mmichelson | 2008-01-14 14:59:26 -0600 (Mon, 14 Jan 2008) | 4 + lines Adding in appropriate unlocks for the locks I added. Thanks + to joetester on IRC for pointing this out. ........ + +2008-01-14 20:01 +0000 [r98830] Joshua Colp <jcolp@digium.com> + + * main/manager.c: Make sure the user's manager secret exists, even + if it is blank. (closes issue #11749) Reported by: srt + +2008-01-14 18:42 +0000 [r98811] Terry Wilson <twilson@digium.com> + + * CHANGES: Add description of TOUPPER and TOLOWER dialplan + functions to CHANGES. + +2008-01-14 17:40 +0000 [r98776] Jason Parker <jparker@digium.com> + + * channels/chan_skinny.c: Add proper call forwarding (all and busy) + support for chan_skinny. Note: NoAnswer support is currently not + implemented, as it would take a significant amount of work to + figure out how to do correctly. Closes issue #11310, patches, + testing, and support by DEA, mvanbaak, and myself. + +2008-01-14 17:39 +0000 [r98775] Russell Bryant <russell@digium.com> + + * /, main/translate.c: Merged revisions 98774 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) | + 3 lines Revert a change that introduces an unacceptable + performance hit and is causing memory leaks ... (from rev 97973) + ........ + +2008-01-14 17:18 +0000 [r98773] Jason Parker <jparker@digium.com> + + * channels/chan_skinny.c: Fix for potential crash with vmexten + +2008-01-14 16:36 +0000 [r98735-98738] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Merged revisions 98737 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan + 2008) | 3 lines Fixing another compilation error. I'm a bit off + today :( ........ + + * /: Blocked revisions 98734 via svnmerge ........ r98734 | + mmichelson | 2008-01-14 10:30:33 -0600 (Mon, 14 Jan 2008) | 3 + lines Oops. Last commit had compilation error. ........ + + * /, apps/app_queue.c: Merged revisions 98733 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan + 2008) | 8 lines Adding explicit defaults for missing options to + init_queue. This is necessary because if a user either removes or + comments one of these options and reloads their queues, the + option will not reset to its default, instead maintaining the + value from prior to the reload. Thanks to John Bigelow for + pointing this error out to me. ........ + +2008-01-14 15:07 +0000 [r98695-98714] Joshua Colp <jcolp@digium.com> + + * main/pbx.c: Print out a warning when spaces are used in the + variable name in Set and MSet. It is extremely hard to debug this + issue so this should make it easier. (closes issue #11759) + Reported by: caio1982 Patches: setvar_space_warning1.diff + uploaded by caio1982 (license 22) + + * apps/app_meetme.c, doc/tex/qos.tex, doc/tex/realtime.tex: Update + documentation. (closes issue #11763) Reported by: IgorG Patches: + docupd.v1.diff uploaded by IgorG (license 20) + +2008-01-14 04:53 +0000 [r98558-98676] Russell Bryant <russell@digium.com> + + * apps/app_jack.c: Add another small option for the JACK app and + JACK_HOOK function. The 'n' option tells JACK not to start jackd + automatically if it is not already running. Otherwise, the + default is that jackd will get started for you if it isn't + running already. + + * CHANGES: - Break up the Misc. section a bit with a new section + for Misc. New Modules - Change spacing a bit in some places for + consistent indentation + + * CHANGES, apps/app_jack.c (added): Bring in the code from + team/russell/jack/. Add a new module, app_jack, which provides + interfaces to JACK, the Jack Audio Connection Kit + (http://www.jackaudio.org/). Two interfaces are provided; there + is a JACK() application, and a JACK_HOOK() function. Both + interfaces create an input and output JACK port. The application + makes these ports the endpoint of the call. The audio coming from + the channel goes out the output port and whatever comes back in + on the input port is what gets sent to the channel. The + JACK_HOOK() function turns on a JACK audiohook on the channel. + This lets you run the audio coming from a channel through JACK, + and whatever comes back in is what gets forwarded on as the + channel's audio. This is very useful for building custom vocoders + or doing recording or analysis of the channel's audio in another + application. In case anyone is curious, the platform that + inspired me to write this is PureData (http://puredata.info/). I + wrote these JACK interfaces so that I could use Pd to do + interesting things with the audio of phone calls ... + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add + configure script check for JACK. + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: + Remove KDE configure script check that isn't used + + * main/audiohook.c: Remove a duplicate lock of the audiohook lock + when destroying manipulate audiohooks. This causes an error when + we attempt to destroy the lock later when freeing the audiohook. + + * main/pbx.c, CHANGES: Add a new CLI command, "core set chanvar", + which allows you to set a channel variable (or function) on an + active channel from the CLI. + +2008-01-12 18:12 +0000 [r98536] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c: Conversion to load manager.conf into memory did + not convert the password functions correctly. (Closes issue + #11749) + +2008-01-12 05:13 +0000 [r98514] Pari Nannapaneni <paripurnachand@digium.com> + + * /, main/http.c: merging a comment added in 1.4 + +2008-01-12 00:20 +0000 [r98488] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, CHANGES: Add 'zap set dnd' CLI command, and + ensure that the AMI DNDState event always gets generated. (closes + issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded + by tzafrir (modified by me) (license 46) + +2008-01-12 00:17 +0000 [r98487] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_odbc.c: Merged revisions 98467 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008) + | 4 lines Add a connection timeout attribute, as that was what + was intended with the login timeout, but ODBC divides it up into + 2 different timeouts. (Closes issue #11745) ........ + +2008-01-11 23:57 +0000 [r98454] Russell Bryant <russell@digium.com> + + * configure, doc/tex/Makefile, configure.ac, makeopts.in: Add some + extra checking to help out with a potential error when trying to + run "make asterisk.pdf" when not all of the right packages are + installed. (closes issue #10763) Reported by: Corydon76 Patches: + 20070919__bug10763.diff.txt uploaded by Corydon76 (license 14) + Tested by: Corydon76 + +2008-01-11 23:10 +0000 [r98436] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add + 'auto' signalling mode for Zaptel channels. (closes issue #11690) + Reported by: tzafrir Patches: signaling_to_signalling.diff + uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded + by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir + (license 46) zap_no_default_sig.diff uploaded by tzafrir (license + 46) zap_signal_auto.diff uploaded by tzafrir (license 46) + +2008-01-11 23:09 +0000 [r98424-98435] Joshua Colp <jcolp@digium.com> + + * main/event.c: Goodbye again drumkilla. + + * main/event.c: drumkilla ftw. + + * main/audiohook.c: I am no longer Rockin' + + * main/audiohook.c: Testing something... + +2008-01-11 22:52 +0000 [r98400] Russell Bryant <russell@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 98390 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) | + 9 lines Fix up setting the EID on BSD based systems. (closes + issue #11646) Reported by: caio1982 Patches: + dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22) + dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested + by: caio1982, mvanbaak ........ + +2008-01-11 19:53 +0000 [r98318-98334] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 98325 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 + lines If the incoming RTP stream changes codec force the bridge + to break if the other side does not support it. (closes issue + #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch + uploaded by tsearle (license 373) ........ + + * /, res/res_agi.c: Merged revisions 98317 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6 + lines If the channel is hungup during RECORD FILE send a result + code of -1 to be uniform with everything else. (closes issue + #11743) Reported by: davevg Patches: res_agi.diff uploaded by + davevg (license 209) ........ + +2008-01-11 19:12 +0000 [r98316] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 98315 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan + 2008) | 5 lines Properly report the hangup cause as no answer + when someone does not answer (closes issue #10574, reported by + boch, patched by moy) ........ + +2008-01-11 19:05 +0000 [r98270-98308] Russell Bryant <russell@digium.com> + + * codecs/codec_resample.c: Kevin noted that the thing that I + _actually_ changed here was that I converted a value from a + double, to a float, back to a double. Sure enough, when I changed + my interim variable back to a double, it still blows up. + Switching all of these to a float fixes the problem. This seems + like a compiler bug where a double passed as an argument isn't + getting properly aligned, so I'll have to see if I can replicate + it with a small test program. (related to issue #11725) + + * codecs/codec_resample.c: Fix a bus error that happened when + asterisk was built with optimizations on with platforms that + explode on unaligned access. I'm not exactly sure why this fixes + it, but it fixed it on the machine I was testing on. If it makes + sense to you, feel free to enlighten me. :) (closes issue #11725, + patched by me) + +2008-01-11 18:35 +0000 [r98268-98269] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_adaptive_odbc.c: Port Nick Gorham's timestamp patch to + adaptive_odbc, too + + * cdr/cdr_odbc.c: Commit Nick Gorham's suggestion for timestamp fix + +2008-01-11 18:26 +0000 [r98267] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 98265 via svnmerge ........ r98265 | russell + | 2008-01-11 12:25:30 -0600 (Fri, 11 Jan 2008) | 11 lines + Backport the ability to set the ToS bits on Linux when not + running as root. Normally, we would not backport features into + 1.4, but, I was convinced by the justification supplied by the + supplier of this patch. He pointed out that this patch removes a + requirement for running as root, thus reducing the potential + impacts of security issues. (closes issue #11742) Reported by: + paravoid Patches: libcap.diff uploaded by paravoid (license 200) + ........ + +2008-01-11 17:27 +0000 [r98220] Joshua Colp <jcolp@digium.com> + + * /, apps/app_followme.c: Merged revisions 98219 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 + lines Ensure the return value of ast_bridge_call is passed back + up as the application return value. This is needed for transfers + to function so the PBX core knows to continue execution. (closes + issue #10327) Reported by: kkiely ........ + +2008-01-11 17:17 +0000 [r98218] Russell Bryant <russell@digium.com> + + * codecs/codec_g722.c: At one point during working on this module, + I had the lin/lin16 versions of the framein callbacks different. + However, they are now the same again, so remove the duplicate + code and use the same functions for the lin/lin16 versions. + +2008-01-11 16:08 +0000 [r98152-98193] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 98164 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008) + | 2 lines Back out changes from revision 97077, since it wasn't + perfect ........ + + * doc/manager_1_1.txt: Documentation updates + +2008-01-11 12:51 +0000 [r98124] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_sip.c: Ascom phones send Flash events as SIP INFO + using '!' as the 'digit' + +2008-01-11 03:40 +0000 [r98081-98083] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 98082 via svnmerge ........ r98082 | russell + | 2008-01-10 21:39:33 -0600 (Thu, 10 Jan 2008) | 2 lines Fix + samples vs. length calculations for g722 ........ + + * codecs/codec_g722.c, main/frame.c: - Fix the last set of places + where incorrect assumptions were made about the sample length + with g722. It is _2_ samples per byte, not 1. This was all over + the place, and I believed it, and it is what caused me to take so + long to figure out what was broken. - Update copyright + information on codec_g722. + +2008-01-11 00:54 +0000 [r98047] Mark Michelson <mmichelson@digium.com> + + * main/translate.c: Fix "core show translation" to not output + information for "unknown" codecs. This fix was made in favor of + the proposed patch since it doesn't involve changing a core codec + define. (closes issue #11722, reported and initially patched by + caio1982, final patch by me) + +2008-01-11 00:38 +0000 [r98024-98027] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add a new + global and per-peer option to chan_sip, qualifyfreq, which allows + you to set the qualify frequency. (closes issue #11597) Reported + by: wilder Patches: qualifyfreq5.patch uploaded by wilder + (license 362) -- with some mods by me + + * /: Blocked revisions 98025 via svnmerge ........ r98025 | russell + | 2008-01-10 18:14:59 -0600 (Thu, 10 Jan 2008) | 3 lines Simplify + this code with a suggestion from Luigi on the asterisk-dev list. + Instead of using is16kHz(), implement a format_rate() function. + ........ + + * main/translate.c: Simplify this code with a suggestion from Luigi + on the asterisk-dev list. Instead of using is16kHz(), implement a + format_rate() function. + +2008-01-10 23:40 +0000 [r97978] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c, main/translate.c: Merged revisions 97973 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) + | 6 lines 1) When we get a translated frame out, clone it, + because if the translator pvt is freed before we use the frame, + bad things happen. 2) Getting a failure from ast_sched_delete + means that the schedule ID is currently running. Don't just + ignore it. (Closes issue #11698) ........ + +2008-01-10 23:33 +0000 [r97974-97977] Russell Bryant <russell@digium.com> + + * /, main/translate.c: Merged revisions 97976 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | + 3 lines Fix various timing calculations that made assumptions + that the audio being processed was at a sample rate of 8 kHz. + ........ + + * codecs/codec_g722.c: Fix various issues in codec_g722. - The most + common fix being made here is to fix all of the places where the + number of output samples and output bytes gets updated in the + translator state structure. - Fix a number of other places where + the number of samples provided as an initialization value to a + struct was incorrect. + + * codecs/codec_resample.c: Fix the buffer_samples value. For signed + linear, the number of samples needed to fill the buffer is half + the buffer size. + +2008-01-10 21:58 +0000 [r97933] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 97925 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan + 2008) | 6 lines Let us leave a voicemail for ourself if we have + logged into VoiceMailMain and chosen to leave a message. (closes + issue #11735, reported and patched by jamessan) ........ + +2008-01-10 21:46 +0000 [r97850-97890] Steve Murphy <murf@digium.com> + + * /, res/ael/ael_lex.c, res/Makefile, res/ael/ael.flex: Merged + revisions 97889 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1 + line Applied the same fixes for ael.flex as was done in 97849 for + ast_expr2.fl; overrode the normally generate yyfree func with our + own version that checks the pointer for non-null before passing + to free(). Also takes care of a little problem with 2.5.33 and + the use of the __STDC_VERSION__ macro. ........ + + * /, main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Merged + revisions 97849 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1 + line This is a fix for 2 things: a problem Terry was having in + OSX with null pointers, which was my fault, as I probably forgot + to run the sed script last time I made mods. So, I moved the fix + into the flex input itself. Then, I found when I used flex + 2.5.33, that it was using __STDC_VERSION__, and that's not real + good; so I added back in a DIFFERENT sed script to fix that + little mess. Tested everything, a couple different ways. Hope I + did no harm, at the least. ........ + +2008-01-10 20:13 +0000 [r97848] Jason Parker <jparker@digium.com> + + * /, include/asterisk/frame.h: Merged revisions 97847 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan + 2008) | 1 line Fix a comment that is no longer true. ........ + +2008-01-10 20:05 +0000 [r97846] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Use the appropriate line ending for the + X-Asterisk-VM-Message-Type header. (closes issue #11734, reported + and patched by jaroth) + +2008-01-10 19:07 +0000 [r97825-97826] Terry Wilson <twilson@digium.com> + + * main/ast_expr2f.c: heh, remove patch to generated file. + + * main/ast_expr2f.c, main/cli.c: Check pointers before freeing (was + getting WARNINGS under MALLOC_DEBUG) + +2008-01-10 17:38 +0000 [r97805] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_odbc.c: Fix problem with timestr going out of scope + (Closes issue #11726, closes issue #11731) + +2008-01-10 17:30 +0000 [r97745-97804] Russell Bryant <russell@digium.com> + + * formats/format_sln16.c: minor formatting changes + + * main/translate.c: spaces to tabs + + * configure, configure.ac: Use AST_EXT_TOOL_CHECK() for the GTK + check again. I changed this to an inline implementation to fix a + small bug, but after a discussion with rizzo, I went to change it + back. Also, it turns out that the implementation of the macro + already supported what was needed to fix the problem. + + * pbx/pbx_kdeconsole.h (removed), /, configs/modules.conf.sample, + pbx/kdeconsole_main.cc (removed): Merged revisions 97753 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) | + 2 lines Remove other remnants of pbx_kdeconsole ........ + + * /, build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, + pbx/pbx_kdeconsole.cc (removed): Merged revisions 97734 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) | + 4 lines Remove pbx_kdeconsole from the tree. It hasn't worked in + ages, and nobody has complained. (closes issue #11706, reported + by caio1982) ........ + +2008-01-10 15:12 +0000 [r97698] Joshua Colp <jcolp@digium.com> + + * funcs/func_groupcount.c, /: Merged revisions 97697 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r97697 | file | 2008-01-10 11:07:12 -0400 (Thu, 10 Jan + 2008) | 6 lines Don't try to copy the category from the group if + no category exists. (closes issue #11724) Reported by: IgorG + Patches: group_count.v1.patch uploaded by IgorG (license 20) + ........ + +2008-01-10 00:54 +0000 [r97657] Russell Bryant <russell@digium.com> + + * include/asterisk.h: These prototypes are not supposed to be in + asterisk.h. They are already in version.h. + +2008-01-10 00:50 +0000 [r97656] Steve Murphy <murf@digium.com> + + * include/asterisk.h, channels/console_video.c, utils/astman.c, + channels/console_board.c, channels/vgrabbers.c: The fixes in this + commit are mainly to allow compiling of trunk with + --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly, + the version.c needs to be in the OBJS line; asterisk.h was chosen + to have the prototypes for ast_get_version, ast_get_version_num; + and the ASTERISK_FILE_VERSION macro needs to be used after + including asterisk.h in a few files. I hope I did the right + thing. If not, let me know. + +2008-01-10 00:39 +0000 [r97655] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c: oops, missed the case of a 0 permission (which + should mean everybody is allowed, not nobody) + +2008-01-10 00:22 +0000 [r97653] Terry Wilson <twilson@digium.com> + + * res/res_phoneprov.c: Attempt at making lookup_iface work under + FreeBSD. Not yet tested, but it compiles under OS X. And still + works under linux. + +2008-01-10 00:17 +0000 [r97652] Russell Bryant <russell@digium.com> + + * codecs/Makefile: Fix this so it doesn't force codec_g722 to get + relinked every time + +2008-01-10 00:12 +0000 [r97651] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, main/manager.c, channels/chan_sip.c, + res/res_features.c, pbx/pbx_realtime.c, + configs/manager.conf.sample, CHANGES, channels/chan_iax2.c, + include/asterisk/manager.h, apps/app_stack.c, main/db.c, + apps/app_voicemail.c: Several manager changes: 1) Add the + Dialplan class, for NewExten and VarSet events, which should cut + down on the volume of traffic in the Call class. 2) Permit some + commands to be run from multiple classes, such as allowing DBGet + to be run from either the System or the Reporting class. 3) + Heavily document each class in the sample config, as there were + several that made no sense to be in the write= line, and two that + made no sense to be in the read= line (since they controlled no + permissions there). (Closes issue #10386) + +2008-01-10 00:11 +0000 [r97641-97650] Russell Bryant <russell@digium.com> + + * codecs/Makefile: Ensure that libg722.a gets rebuilt if one of the + files changes + + * /, pbx/pbx_gtkconsole.c: Merged revisions 97645 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) | + 2 lines Strip terminal sequences from the verbose messages + ........ + + * configure: re-gen configure + + * configure.ac: re-add check for gtk1, which is used for + pbx_gtkconsole (related to issue #11706) + + * /, pbx/pbx_gtkconsole.c: Merged revisions 97640 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97640 | russell | 2008-01-09 16:26:33 -0600 (Wed, 09 Jan 2008) | + 3 lines Make pbx_gtkconsole build ... but doesn't actually load + on my system still (related to issue #11706) ........ + +2008-01-09 21:37 +0000 [r97634] Terry Wilson <twilson@digium.com> + + * phoneprov/000000000000.cfg, phoneprov/000000000000-directory.xml, + phoneprov/polycom.xml, res/res_phoneprov.c (added), + funcs/func_strings.c, phoneprov/000000000000-phone.cfg, + configs/modules.conf.sample, main/acl.c, + include/asterisk/localtime.h, CHANGES, + configs/phoneprov.conf.sample (added), Makefile, phoneprov + (added), doc/tex/phoneprov.tex (added), main/stdtime/localtime.c, + doc/tex/asterisk.tex: Added a new module, res_phoneprov, which + allows auto-provisioning of phones based on configuration + templates that use Asterisk dialplan function and variable + substitution. It should be possible to create phone profiles and + templates that work for the majority of phones provisioned over + http. It is currently only intended to provision a single user + account per phone. An example profile and set of templates for + Polycom phones is provided. NOTE: Polycom firmware is not + included, but should be placed in AST_DATA_DIR/phoneprov/configs + to match up with the included templates. + +2008-01-09 20:30 +0000 [r97620-97623] Jason Parker <jparker@digium.com> + + * /, main/cli.c: Merged revisions 97622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11718) ........ r97622 | qwell | 2008-01-09 14:28:43 -0600 + (Wed, 09 Jan 2008) | 5 lines Correctly display a message if a + command could not be found. Also fix a comment which may have led + to this happening. Issue 11718, reported by kshumard. ........ + + * /, main/cli.c: Merged revisions 97618 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1 + line Fix some locking and return value funkiness. We really + shouldn't be unlocking this lock inside of a function, unless we + locked it there too. ........ + +2008-01-09 18:53 +0000 [r97577] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 97575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97575 | mmichelson | 2008-01-09 12:48:15 -0600 (Wed, 09 Jan + 2008) | 3 lines Part 2 of app_queue doxygen improvements. Some + smaller functions this time ........ + +2008-01-09 18:12 +0000 [r97532-97533] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c: remove a wrong 'const' + + * images/kpad2.jpg: add annotations for the two message windows we + use. + +2008-01-09 18:04 +0000 [r97531] Russell Bryant <russell@digium.com> + + * /, res/res_features.c: Merged revisions 97529 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97529 | russell | 2008-01-09 12:02:08 -0600 (Wed, 09 Jan 2008) | + 2 lines Fix saying the parking space number to the caller doing + the parking ... ........ + +2008-01-09 18:03 +0000 [r97530] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c, channels/console_board.c, + channels/console_video.h: Two changes: - support scrolling of + message window; - simplify the code for creating a message + window, and try it using a second one in the top of the keypad + (where we echo the dialed number). The 'skin' that supports these + two windows will be committed separately. + +2008-01-09 17:30 +0000 [r97495] Kevin P. Fleming <kpfleming@digium.com> + + * /, codecs/codec_zap.c: Merged revisions 97491 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008) + | 2 lines report the same message whether Zaptel does not have + transcoder support loaded or no transcoders were found ........ + +2008-01-09 16:59 +0000 [r97490] Philippe Sultan <philippe.sultan@gmail.com> + + * /, channels/chan_gtalk.c: Merged revisions 97489 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 + Jan 2008) | 7 lines Set the caller id within the gtalk_alloc + function. As underlined in issue #10437 by Josh, we need to + prevent a possible memory leak. We only set the name part of the + caller id, the number part is not relevant when dealing with + JIDs. Closes issue #11549. ........ + +2008-01-09 16:44 +0000 [r97488] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c, channels/console_video.c, + channels/console_board.c, channels/console_video.h: Implement + keyboard handling, and use it to enter a number to dial in the + 'message' area under the keypad. Now you can make calls using the + keypad as a regular phone (or the keyboard for chars not present + on the keypad) + +2008-01-09 16:13 +0000 [r97451] Joshua Colp <jcolp@digium.com> + + * /, apps/app_meetme.c: Merged revisions 97450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97450 | file | 2008-01-09 12:11:17 -0400 (Wed, 09 Jan 2008) | 6 + lines Don't do conferencing totally in Zaptel if Monitor is + running on the channel. (closes issue #11709) Reported by: + BigJimmy Patches: patch-meetmerec uploaded by BigJimmy (license + 371) ........ + +2008-01-09 15:45 +0000 [r97421-97449] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, /: Merged revisions 97448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97448 | kpfleming | 2008-01-09 09:43:19 -0600 (Wed, 09 Jan 2008) + | 2 lines pass the right variable to get an error string... oops + ........ + + * channels/chan_zap.c, /: Merged revisions 97410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97410 | kpfleming | 2008-01-09 09:26:23 -0600 (Wed, 09 Jan 2008) + | 2 lines add error number output to ioctl failure messages to + help with debugging ........ + +2008-01-09 12:23 +0000 [r97389-97390] Luigi Rizzo <rizzo@icir.org> + + * channels/console_video.c, channels/console_video.h: implement the + "console startgui" and "console stopgui" commands so you can + start and stop the gui even outside of a call. This is convenient + for testing, and also for using the keypad to pick up a call, and + to dial a number (the latter not yet implemented, but should be + close). + + * channels/chan_oss.c: make get_video_desc() return the active + console if passed a null argument (channel). + +2008-01-09 00:58 +0000 [r97364-97365] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c: New option in trunk, needs strdupa to be safe, + too + + * /, main/editline/readline.c, main/cli.c: Merged revisions 97350 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008) + | 5 lines Allow filename completion on zero-length modules, + remove a memory leak, remove a file descriptor leak, and make + filename completion thread-safe. Patched and tested by tilghman. + (Closes issue #11681) ........ + +2008-01-09 00:18 +0000 [r97307-97309] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 97308 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97308 | mmichelson | 2008-01-08 18:17:40 -0600 (Tue, 08 Jan + 2008) | 3 lines use the \retval doxygen command properly ........ + + * /, apps/app_queue.c: Merged revisions 97304 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97304 | mmichelson | 2008-01-08 17:49:11 -0600 (Tue, 08 Jan + 2008) | 5 lines Part 1 of N of adding doxygen comments to + app_queue. I picked some of the most common functions used (which + also happen to be some the biggest/ugliest functions too) to + document first. I'm pretty new to doxygen so criticism is + welcome. ........ + +2008-01-08 23:51 +0000 [r97305] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Add a new flag 'd' (with optional context) + permitting any extension within that context to be entered as a + new extension during the playback of a voicemail greeting. Patch + inspired by bluecrow76, by tilghman. (Closes issue #7063) + +2008-01-08 23:35 +0000 [r97280-97303] Luigi Rizzo <rizzo@icir.org> + + * channels/console_board.c: add copyright (most of this code was + written by Marta Carbone), remove some unused code, add/clarify + some comments. + + * images/kpad2.jpg: Add the annotation for the textarea used for + messages, and also change the background from white to something + different to show that we can make use of fonts with transparent + background. + + * images/font.png (added): add a font suitable for use with the + console GUI. The background of this particular image is + transparent so we can preserve the original background when we + draw strings. + + * channels/console_gui.c, channels/console_video.c, + channels/console_board.c (added), channels/Makefile: add support + for textareas, used for various dialog windows on the gui. The + main code to implement the textarea is in console_board.c, and + uses a simple png image with the font, blitting characters on the + designated areas of the main screen. Additionally we provide some + annotations in the image used as a skin to indicate which areas + are used for text messages. (images will be committed + separately). At the moment the dialog area is only used to + display a running counter, just as a proof of concept. + +2008-01-08 21:56 +0000 [r97248] Terry Wilson <twilson@digium.com> + + * apps/app_queue.c: Initialize new variable to NULL + +2008-01-08 21:28 +0000 [r97203-97208] Mark Michelson <mmichelson@digium.com> + + * /: Blocked revisions 97206 via svnmerge ........ r97206 | + mmichelson | 2008-01-08 15:24:48 -0600 (Tue, 08 Jan 2008) | 3 + lines Some coding guidelines-related cleanup ........ + + * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the + option of specifying a second interface in a member definition + for a queue. app_queue will monitor this second device's state + for the member, even though it actually calls the first + interface. This ability has been added for statically defined + queue members, realtime queue members, and dynamic queue members + added through the CLI, dialplan, or manager. (closes issue + #11603, reported by acidv) + +2008-01-08 21:01 +0000 [r97199-97200] Olle Johansson <oej@edvina.net> + + * channels/chan_console.c: Change reference to external library so + it appears on the extref listing + http://www.asterisk.org/doxygen/trunk/extref.html + + * res/res_jabber.c: Iksemel is alive in a new home. Release 1.3 is + out with bug fixes. + +2008-01-08 20:56 +0000 [r97198] Tilghman Lesher <tlesher@digium.com> + + * main/autoservice.c, /, main/utils.c: Merged revisions 97194 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008) + | 3 lines Increase constants to where we're less likely to hit + them while debugging. (Closes issue #11694) ........ + +2008-01-08 20:52 +0000 [r97196-97197] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: One line documentation ftw! + + * /, channels/chan_mgcp.c: Merged revisions 97195 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6 + lines Fix various DTMF issues in chan_mgcp. (closes issue #11443) + Reported by: eferro Patches: + dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license + 337) ........ + +2008-01-08 20:45 +0000 [r97193] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 97192 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan + 2008) | 9 lines Making some changes designed to not allow for a + corrupted mailstream for a vm_state. 1. Add locking to the + vm_state retrieval functions so that no linked list corruption + occurs. 2. Make sure to always grab the persistent vm_state when + mailstream access is necessary. 3. Correct an incorrect return + value in the init_mailstream function. (closes issue #11304, + reported by dwhite) ........ + +2008-01-08 20:06 +0000 [r97153-97154] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Move common code for setting T38 + capabilities and fix a bug with fax detection in the SIP RTP read + callback. It's still sort of silly... but more on that later. + (closes issue #11239) Reported by: dimas Patches: + sipt38prop.patch uploaded by dimas (license 88) + + * funcs/func_groupcount.c, /: Merged revisions 97152 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r97152 | file | 2008-01-08 15:53:52 -0400 (Tue, 08 Jan + 2008) | 4 lines If no group has been provided to the GROUP_COUNT + dialplan function then use the first one specific to the channel. + (closes issue #11077) Reported by: m4him ........ + +2008-01-08 19:06 +0000 [r97125] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c, main/asterisk.c: Merged revisions 97077 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008) + | 3 lines Apply multiple crash fixes, found in issue #11386, but + not completely closing that issue. ........ + +2008-01-08 18:42 +0000 [r97041-97103] Joshua Colp <jcolp@digium.com> + + * /, apps/app_queue.c: Merged revisions 97093 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r97093 | file | 2008-01-08 14:36:40 -0400 (Tue, 08 Jan 2008) | 4 + lines Make app_queue calls work with directed pickup. (closes + issue #11700) Reported by: jbauer ........ + + * utils/extconf.c: Make ast_atomic_fetchadd_int_slow magically + appear in extconf. (closes issue #11703) Reported by: dmartin + +2008-01-07 23:03 +0000 [r96988] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c: add support for cropping the keypad image + while displaying it. This way it can contain additional elements + (e.g. fonts, buttons, widgets) without having to use a zillion + files to store them. + +2008-01-07 22:31 +0000 [r96987] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Explicitly make literal constants long + where they are expected to be. + +2008-01-07 21:12 +0000 [r96936] Jason Parker <jparker@digium.com> + + * main/config.c: Display a message if no config mappings are found + with "core show config mappings". Closes issue #11704, patch by + kshumard. + +2008-01-07 21:10 +0000 [r96934-96935] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Document some weird casting magic that's + necessary to interface with the c-client + + * doc/tex/imapstorage.tex, apps/app_voicemail.c: Adding + user-configurable TCP timeout settings to IMAP voicemail. This + could go a long way towards preventing unexplainable hangs + experienced by people. In the case of MWI hangs, this also will + mean that the SIP port isn't blocked anymore. (closes issue + #11665, reported by yehavi) + +2008-01-07 20:48 +0000 [r96885-96933] Russell Bryant <russell@digium.com> + + * /, configs/extensions.conf.sample: Merged revisions 96932 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r96932 | russell | 2008-01-07 14:47:52 -0600 + (Mon, 07 Jan 2008) | 10 lines Merged revisions 96931 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 + Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com + ........ ................ + + * configs/http.conf.sample: Add a note about viewing the default + set of documentation using the built-in http server + + * Makefile: If the HTML documentation exists, install it in the + static-http/docs directory so that it can be viewed through the + Asterisk http server if it is turned on. + + * build_tools/prep_tarball: Build the HTML version of the doc files + for tarballs, as well + + * res/res_smdi.c, /: Merged revisions 96884 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r96884 | russell | 2008-01-07 10:39:23 -0600 (Mon, 07 Jan 2008) | + 3 lines Don't crash if something happens when setting up an SMDI + interface and it gets destroyed before the SMDI port handling + thread gets created. ........ + +2008-01-07 16:17 +0000 [r96862] Kevin P. Fleming <kpfleming@digium.com> + + * formats/format_sln16.c (added): add a file-format driver for + 16KHz signed linear... which may or may not work + +2008-01-07 15:52 +0000 [r96858] Joshua Colp <jcolp@digium.com> + + * main/manager.c, main/loader.c: Move ModuleLoad and ModuleCheck + manager commands from loader.c to manager.c. Previously they + would get registered twice because of the way manager.c operates. + (closes issue #11699) Reported by: caio1982 Patches: + manager_module_commands1.diff uploaded by caio1982 (license 22) + +2008-01-07 15:06 +0000 [r96776-96836] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c: update comments to reflect reality (or at + least planned behaviour). minor code cleanups + + * channels/console_gui.c: resolve a load-time problem avoiding a + call to console_do_answer. On passing, fix dialling from the + keypad. + +2008-01-05 23:05 +0000 [r96645-96743] Russell Bryant <russell@digium.com> + + * res/snmp/agent.c: Convert this file over the new method of + getting the Asterisk version. (I don't have this building on this + machine, so caio1982 on IRC is going to test it for me. :) ) + + * Makefile, funcs/func_version.c, main/manager.c, + channels/chan_sip.c, main/Makefile, build_tools/make_version_c + (added), include/asterisk/version.h (added), res/res_agi.c, main, + main/http.c, build_tools/make_version_h (removed), + include/asterisk, main/asterisk.c: Now that the version.h file + was getting properly regenerated every time the svn revision + changed, every module that used the version was getting rebuilt + after every svn update. This severly annoyed me pretty quickly, + so I have improved the situation. Now, instead of generating + version.h, main/version.c is generated. version.c includes the + version information, as well as a couple of API calls for modules + to retrieve the version. So now, only version.c will get rebuilt, + and the main asterisk binary relinked, which is must faster than + rebuilding http.c, manager.c, asterisk.c, relinking the asterisk + binary, chan_sip.c, func_version.c, res_agi ... The only minor + change in behavior here is that the version information reported + by chan_sip, for example, is the version of the Asterisk core, + and not necessarily the Asterisk version that the chan_sip module + came from. + + * main/pbx.c: Print out the name of a function being registered in + color, just like the name of applications when they get + registered. + + * UPGRADE.txt: Add a note about changing modules.conf since another + console channel driver is now present that can not be used at the + same time as chan_alsa or chan_oss. + + * channels/chan_console.c: Add the URL to the home page for + portaudio. Also add the location of the svn repository to check + out portaudio v19. + + * /, main/devicestate.c: Merged revisions 96644 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) | + 2 lines Don't pass an empty string as the device name. ........ + +2008-01-05 01:05 +0000 [r96621] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_usbradio.c: improve chan_usbradio to use + indications just like chan_alsa/chan_oss do now + +2008-01-04 23:12 +0000 [r96576] Tilghman Lesher <tlesher@digium.com> + + * /, main/devicestate.c: Merged revisions 96575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008) + | 7 lines Fix the problem of notification of a device state + change to a device with a '-' in the name. Could probably do with + a better fix in trunk, but this bug has been open way too long + without a better solution. Reported by: stevedavies Patch by: + tilghman (Closes issue #9668) ........ + +2008-01-04 22:57 +0000 [r96574] Jason Parker <jparker@digium.com> + + * /, res/res_features.c: Merged revisions 96573 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes + issue #11237) ........ r96573 | qwell | 2008-01-04 16:55:56 -0600 + (Fri, 04 Jan 2008) | 4 lines Properly continue in the dialplan if + using PARKINGEXTEN and the slot is full. Issue 11237, patch by + me. ........ + +2008-01-04 19:35 +0000 [r96547] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 96525 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008) + | 4 lines If you change the bindaddr in sip.conf to a non-bound + address and reload, sip goes kablooie. Reported and patched by: + one47 (Closes issue #11535) ........ + +2008-01-04 17:21 +0000 [r96500] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, + configure.ac, acinclude.m4: [commit message] (closes issue + #10393) Reported by: tzafrir Patches: chan_alarm_asterisk.diff + uploaded by tzafrir (license 46) (modified by me and added + configure script support) + +2008-01-04 17:19 +0000 [r96499] Philippe Sultan <philippe.sultan@gmail.com> + + * res/res_jabber.c: Use SASL DIGEST-MD5 authentication over + unsecured network connections only. This authentication mechanism + is implemented under the iksemel API, which makes use of GnuTLS, + whereas we use OpenSSL. Note : there's ongoing dicsussion at the + SASL IETF WG in order to deprecate SASL DIGEST-MD5, see + http://ietfreport.isoc.org/ids-wg-sasl.html. + +2008-01-04 16:21 +0000 [r96450] Russell Bryant <russell@digium.com> + + * channels/chan_zap.c, /: Merged revisions 96449 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | + 7 lines Make use of the temporary channel pointer while the pvt + is unlocked. (closes issue #11675) Reported by: flefoll Patches: + chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll + (license 244) ........ + +2008-01-03 23:14 +0000 [r96397-96398] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile: we have to *always* use a completely silent 'make' + invocation for generating the module embedding rules + + * Makefile: there was no reason to add this define for non-Solaris + platforms + +2008-01-03 22:46 +0000 [r96395] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 96394 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) | + 3 lines Don't crash if the iax2 pvt structure has been destroyed + before we get to this point (closes issue #11672, reported by + snuffy, patched by me) ........ + +2008-01-03 21:58 +0000 [r96301-96368] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/channel.h: Document recent API addition + + * res/res_config_pgsql.c, /: Merged revisions 96318 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r96318 | tilghman | 2008-01-03 15:37:02 -0600 (Thu, 03 + Jan 2008) | 4 lines Missed initialization caused crash. Reported + and fixed by: tiziano (Closes issue #11671) ........ + + * main/channel.c: Allow the uniqueid to be used for searching for a + channel in the list. Reported and initially patched by: + michael-fig (Closes issue #11340) + +2008-01-03 20:04 +0000 [r96245-96272] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, tests/Makefile (added), tests/test_skel.c (added), + tests (added): add some simple infrastructure for modules to be + used for testing parts of Asterisk + + * channels/answer.h (removed), channels/ring10.h (removed), + channels/busy.h (removed), channels/ringtone.h (removed), + channels/Makefile, channels/chan_oss.c, channels/gentone.c + (removed), channels: eliminiate sound_thread() and other stuff + from chan_oss since Asterisk indications can handle it remove + gentone and all the headers containing tones that are no longer + needed + + * channels/chan_alsa.c: coding guidelines cleanup remove background + thread and all sound generation mechanisms, as the built-in + indications can handle everything that is needed + +2008-01-03 14:47 +0000 [r96221] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /: Merged revisions 96198 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03 + Jan 2008) | 1 line when overlapdial was used and no number was + dialed, the call was dropped, now we just jump into the s + extension, which makes a lot more sense. ........ + +2008-01-03 06:16 +0000 [r96147-96174] Tilghman Lesher <tlesher@digium.com> + + * res/res_agi.c: Add coordination between AMI and AGI applications, + with an asyncagi method Feature proposed and patched by: moy + (Closes issue #11282) + + * apps/app_mp3.c, apps/app_ices.c, main/asterisk.c: Compatibility + fix for OpenBSD Report and fix by: mvanbaak (Closes issue #11669) + +2008-01-02 23:48 +0000 [r96103] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 96102 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan + 2008) | 4 lines We need to reset the membername to NULL on each + iteration of this loop, otherwise the result is that multiple + members can have the same name, since the variable was not reset + on each iteration of the loop. ........ + +2008-01-02 23:22 +0000 [r96076-96079] Russell Bryant <russell@digium.com> + + * channels/chan_console.c: Add support for generating a ringing + sound on an incoming call. This is a bit of a hack. It just asks + the core to generate the same tone that it would when you hear + ringback when making an outbound call. But hey, it works, and you + get the localized ring tone for the appropriate language set on + the channel. + + * channels/chan_console.c: Note that this module doesn't actually + play a ringing sound for an incoming call ... oops + + * channels/chan_console.c: Show the correct CLI command to answer + the call + +2008-01-02 22:41 +0000 [r96073] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c: actually parse and store echocan parameters + from zapata.conf... this *should* work <G> + +2008-01-02 22:40 +0000 [r96071] Joshua Colp <jcolp@digium.com> + + * configure, include/asterisk/autoconfig.h.in, configure.ac: Don't + use AST_C_DEFINE_CHECK for the two pthread things that may not + actually be definitions, they could be enums for example. + +2008-01-02 22:29 +0000 [r96028] Mark Michelson <mmichelson@digium.com> + + * channels/chan_zap.c: Add curly braces around a compound if + statement so that trunk will build properly + +2008-01-02 22:15 +0000 [r96021-96025] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 96024 via svnmerge ........ r96024 | russell + | 2008-01-02 16:14:28 -0600 (Wed, 02 Jan 2008) | 2 lines Convert + locks of the contexts list in pbx_config to the appropriate + rdlock or wrlock ........ + + * /: Blocked revisions 96022 via svnmerge ........ r96022 | russell + | 2008-01-02 16:04:47 -0600 (Wed, 02 Jan 2008) | 2 lines + pbx_dundi only needs a rdlock on the contexts list. ........ + + * /: Blocked revisions 96020 via svnmerge ........ r96020 | russell + | 2008-01-02 16:00:21 -0600 (Wed, 02 Jan 2008) | 2 lines + app_macro only needs a rdlock on the contexts list. ........ + +2008-01-02 21:51 +0000 [r96019] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, configs/zapata.conf.sample: another + checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS + ioctl if it is present, but doesn't parse any supplied parameters + yet (this implementation is not very memory efficient as the + parameters and their values will be duplicated for each channel + that has the same settings, but we can worry about that later + once it is working) + +2008-01-02 21:49 +0000 [r96018] Russell Bryant <russell@digium.com> + + * main/libresample/include/libresample.h: Add doxygen documentation + to libresample.h while it's still fresh on my mind + +2008-01-02 21:08 +0000 [r95994] Mark Michelson <mmichelson@digium.com> + + * funcs/func_odbc.c, channels/chan_agent.c, funcs/func_strings.c, + apps/app_rpt.c: Change instances of AST_NONSTANDARD_APP_ARGS(foo, + bar, ',') to AST_STANDARD_APP_ARGS(foo, bar) (closes issue + #11668, reported and patched by mvanbaak) + +2008-01-02 20:26 +0000 [r95947] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 95946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4 + lines Allocate a SIP refer structure when performing a transfer + using BYE with Also so that the transfer information is properly + stored. (AST-2008-001) (closes issue #11637) Reported by: + greyvoip ........ + +2008-01-02 20:23 +0000 [r95944-95945] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Since ',' is the standard argument separator in + trunk, change app_queue to use AST_STANDARD_APP_ARGS instead of + AST_NONSTANDARD_APP_ARGS for determining member data. + + * include/asterisk/app.h: Fix a typo in a comment. + AST_STANDARD_APP_ARGS uses ',' as the separator, not '|'. + +2008-01-02 19:47 +0000 [r95893-95939] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c: clean up hwgain CLI command and improve docs + for swgain CLI command + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + acinclude.m4: improve AC_C_DEFINE_CHECK to not try to evaluate + the macro being checked for, but just check for its existence + finish implementation of check for Zaptel HWGAIN support add + check for Zaptel ECHOCANCEL_PARAMS support + + * codecs/Makefile, include/asterisk/libresample.h (added), + codecs/codec_resample.c: and now just to keep the libresample + party going... if the functions from libresample are going to be + in the main Asterisk binary, it makes sense for the header that + defines them to be available without any special CFLAGS and to + out-of-tree modules building against /usr/include/asterisk + + * channels/chan_zap.c: umm... this did not compile on x86-64, and + could not possibly have worked on any platform as it was passing + string pointers to a function expecting ints + +2008-01-02 18:05 +0000 [r95891] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 95890 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan + 2008) | 9 lines A change to improve the accuracy of queue logging + in the case where a member does not answer during the specified + timeout period. Prior to this change, there was a small chance + that the member name recorded in this case would be blank. Also + prior to this change, if using the ringall strategy, if no one + answered the call during the specified timeout, the member name + listed in the queue log would randomly be one of the members that + was rung. (closes issue #11498, reported and tested by hloubser, + patched by me) ........ + +2008-01-02 17:38 +0000 [r95888] Jason Parker <jparker@digium.com> + + * apps/app_osplookup.c: Update osplookup documentation to use + commas instead of pipes. Closes issue #11666, patch by Laureano. + +2008-01-02 16:20 +0000 [r95864] Russell Bryant <russell@digium.com> + + * main/Makefile, main/translate.c: For some odd reason, the last + set of libresample build changes from Kevin did not work for + everyone, but it did for some. This set of changes makes trunk + start again for those having problems. Instead of building + libresample as a static library, it just links the object files + in directly with the asterisk binary. + +2008-01-02 14:53 +0000 [r95816-95841] Kevin P. Fleming <kpfleming@digium.com> + + * channels/Makefile: fix some long-time breakage that kept + chan_misdn from being embedded + + * channels/Makefile: use the proper technique for including + submodules so that embedding will work + + * CHANGES: note that chan_console requires portaudio v19 + + * configure, configure.ac: actually check for a function present in + libiconv (don't know how this test could have worked before) and + don't do the check on Linux/GNU systems because libiconv is not + present there and attempting to link with '-liconv' always fails + (it's not necessary as the iconv functionality is always + available) + + * main/libresample/src/filterkit.h, + main/libresample/src/resample.c, + main/libresample/win/libresample.dsp, main/libresample/configure, + main/libresample/Makefile.in, res/Makefile, + main/libresample/configure.in, main/libresample/src, + main/libresample/tests/testresample.c, + main/libresample/win/libresample.vcproj, + main/libresample/tests/compareresample.c, main/libresample/tests, + codecs/codec_resample.c, res/res_resample.c (removed), + main/libresample/README.txt, main/libresample/src/resamplesubs.c, + main/libresample/tests/resample-sndfile.c, + main/libresample/src/configtemplate.h, + main/libresample/install-sh, main/Makefile, main/translate.c, + main/libresample/include, main/libresample/src/resample_defs.h, + codecs/Makefile, main/libresample/config.guess, + main/libresample/config.sub, main/libresample/win, + main/libresample/LICENSE.txt, main/libresample (added), + main/libresample/Makefile.asterisk, build_tools/strip_nonapi, + res/libresample (removed), main/libresample/src/filterkit.c, + main/libresample/include/libresample.h: go back to including + libresample in the main Asterisk binary, but this time including + a small hack to ensure that it does get linked in (and also + modify the strip_nonapi script to leave the resample_<foo> + symbols alone) + +2008-01-02 11:34 +0000 [r95794] Philippe Sultan <philippe.sultan@gmail.com> + + * res/res_jabber.c: Set stream flags to zero upon initialization. + When the XMPP over TLS/SSL connection resets for some reason, it + is wrongly believed as being secured, which makes the + re-connection process endlessly fail. This was reported by + mvanbaak in issue #11644. + +2008-01-02 09:16 +0000 [r95771-95772] Luigi Rizzo <rizzo@icir.org> + + * main/loader.c: some cleanup of this code while I am trying to + debug a problem with gdb dying while debugging asterisk. The + problem seems to be related with a race in the handling of + module_list, which in turn is triggeded by calling dlopen() on a + system which uses initializers to create locks. + + * include/asterisk/module.h: There are three instances of the + module definition macros, which make maintaining this file very + error prone. This commit merges the embedded and !embedded + versions, and fixes the C++ version. Eventually we should move to + a single version of the macro. Too bad C++ doesn't like the + C-style struct initializers .foo = some_value + +2008-01-02 04:33 +0000 [r95697-95746] Russell Bryant <russell@digium.com> + + * res/libresample/src/resample_defs.h, + res/libresample/src/resample.c: Don't make libresample print out + debugging output + + * main/translate.c: Make the translation table show slin16 + + * apps/app_meetme.c: fix a spacing issue introduced in revision + 95443. + + * main/Makefile, res/libresample/README.txt, res/Makefile, + res/libresample/install-sh, res/libresample/configure, + res/libresample/Makefile.in, res/libresample/include, + codecs/Makefile, res/libresample/configure.in, + res/libresample/src, res/libresample/config.guess, + main/libresample (removed), res/libresample/config.sub, + res/libresample/win, codecs/codec_resample.c, + res/libresample/LICENSE.txt, res/libresample (added), + res/libresample/Makefile.asterisk, res/libresample/tests, + res/res_resample.c (added): Instead of linking libresample into + the main Asterisk binary, build it as res_resample, and mark + codec_resample as dependent upon res_resample. This prevents the + linker from optimizing away libresample, and also makes it so the + libresample code isn't linked in to multiple places. (I have + another module in a branch that needs it, too.) + +2008-01-01 23:55 +0000 [r95671-95673] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c: call directly the cli command to + implement hangup. + + * channels/vcodecs.c: prevent a panic when destroying a channel + with no incoming video. + + * channels/console_video.c: remove a leftover sleep(1) used for + debugging + +2008-01-01 23:09 +0000 [r95648] Joshua Colp <jcolp@digium.com> + + * codecs/Makefile: Fix building of codec_resample on platforms + other then Cygwin. On everything else it actually gets built + after codec_resample, so you can't exactly link it in since it + doesn't exist. + +2008-01-01 22:21 +0000 [r95624-95625] Luigi Rizzo <rizzo@icir.org> + + * codecs/Makefile, codecs/codec_resample.c: make codec_resample + build on __CYGWIN__, and make it load on FreeBSD (and probably + other systems as well). Both need libresample.a to be specified + in the linking phase, and cygwin needs <float.h> as other BSD. + The checks for OS-specific headers should really be moved to some + common header though. + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, + funcs/func_iconv.c, makeopts.in: implement "configure" checks for + libiconv, and add the iconv dependency for func_iconv. This fixes + some build issues on CYGWIN and FreeBSD and probably other + platforms where libiconv is not there by default + +2007-12-31 23:44 +0000 [r95578] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c, /: Merged revisions 95577 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec + 2007) | 9 lines Avoiding a potentially bad locking situation. + ast_merge_contexts_and_delete writelocks the conlock, then calls + ast_hint_extension, which attempts to readlock the same lock. + Recursion with read-write locks is dangerous, so the inner lock + needs to be removed. I did this by copying the "guts" of + ast_hint_extension into ast_merge_contexts_and_delete (sans the + extra lock). (this change is inspired by the locking problems + seen in issue #11080, but I have no idea if this is the + problematic area experienced by the reporters of that issue) + ........ + +2007-12-31 22:41 +0000 [r95501-95550] Russell Bryant <russell@digium.com> + + * codecs/codec_resample.c: Use float.h to fix the build on FreeBSD. + Also, add some other platforms as they are likely the same. + + * channels/chan_console.c: Update chan_console to natively use a 16 + kHz sample rate. If it is talking to an 8 kHz endpoint, then + codec_resample will automatically be used to properly resample + the audio before sending it to/from chan_console. + + * main/libresample/src/filterkit.h, main/libresample/README.txt, + main/libresample/tests/resample-sndfile.c, + main/libresample/src/resamplesubs.c, main/Makefile, + main/libresample/install-sh, + main/libresample/src/configtemplate.h, + main/libresample/src/resample.c, + main/libresample/win/libresample.dsp, main/libresample/configure, + main/libresample/Makefile.in, main/libresample/include, CHANGES, + main/libresample/src/resample_defs.h, + main/libresample/configure.in, main/libresample/src, + main/libresample/config.guess, codecs/Makefile, + main/libresample/tests/testresample.c, codecs/slin_resample_ex.h + (added), main/libresample/config.sub, main/libresample/win, + main/libresample/win/libresample.vcproj, + main/libresample/LICENSE.txt, main/libresample (added), + main/libresample/Makefile.asterisk, main/libresample/tests, + main/libresample/tests/compareresample.c, codecs/codec_resample.c + (added), main/libresample/src/filterkit.c, + main/libresample/include/libresample.h: Merge changes from + team/russell/codec_resample This commit imports libresample for + use in Asterisk. It also adds a new codec module, codec_resample. + This module uses libresample to re-sample signed linear audio + between 8 kHz and 16 kHz. It also provides an alternative for + converting between 16 kHz G.722 and 8 kHz signed linear when + using G.722, which will likely be useful as some people have + complained about volume issues when the current codec_g722 + converts to 8 kHz signed linear. But, to test this, you will have + to disable the g722-to-slin and g722-to-slin16 translators in + codec_g722.c. + +2007-12-31 20:33 +0000 [r95490] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_env.c: Merged revisions 95470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r95470 | tilghman | 2007-12-31 14:27:26 -0600 (Mon, 31 Dec 2007) + | 3 lines Allow the default "0" to be returned if the STAT fails + (Closes issue #11659) ........ + +2007-12-31 18:46 +0000 [r95443] Mark Michelson <mmichelson@digium.com> + + * apps/app_meetme.c: Fix a compiler warning (closes issue #11658, + reported and patched by eliel) + +2007-12-31 16:13 +0000 [r95383-95412] Russell Bryant <russell@digium.com> + + * configs/console.conf.sample (added), configs/modules.conf.sample, + channels/chan_console.c (added), CHANGES: Merge the main set of + changes from team/russell/chan_console. Add a new console channel + driver, chan_console, which is a console channel driver that uses + portaudio as a cross platform audio interface. It was written to + provide a console channel driver that works with Mac CoreAudio, + but it supports a number of other audio interfaces, as well, + including OSS and ALSA. It could one day be the single console + channel driver, but does not yet have as many features as + chan_oss. + + * include/asterisk/channel.h: fix a spelling error in a comment + + * include/asterisk/config.h: Add CV_STRINGFIELD() macro. This lets + you set a config variable to a string field. (from + team/russell/chan_console) + + * configure, include/asterisk/autoconfig.h.in: Regenerate configure + script to include check for portaudio. + + * build_tools/menuselect-deps.in, configure.ac, makeopts.in: Add + configure script checking for portaudio. + +2007-12-29 02:02 +0000 [r95262-95313] Luigi Rizzo <rizzo@icir.org> + + * channels/vcodecs.c, channels/console_video.c, channels/Makefile, + channels/console_video.h, channels/vgrabbers.c (added): Move + grabbers definitions to a separate file, vgrabbers.c, so it is + easier to add more entries. This required moving struct grab_desc + to the common header, and adding an entry in the Makefile. On + passing, cleanup some comments and file headers (some are still + missing). + + * channels/console_gui.c, channels/console_video.c: virtualize the + interface for video grabbers, which should make it easier to add + support for more grabbers (V4L2, firewire, and so on). + + * channels/console_video.c: Add a few entries up to 1408x1152 in + the table of known video resolutions. This makes it very + convenient to enlarge images using the right-click on the video + window. + + * channels/vcodecs.c, channels/console_video.c: change the + interface of video encapsulation routines, they only need the + buffer and mtu as input. + + * channels/console_gui.c, channels/vcodecs.c, + channels/console_video.c, channels/console_video.h: various + rearrangements and renaming of console_video stuff + +2007-12-28 18:39 +0000 [r95233] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: The diff for this change looks really bad, but + all I did here was decrease the indentation of most of the + queue_exec function by reversing the logic of an if statement. + This change makes the function comply better with the coding + guidelines. Since this change is purely a cosmetic change to the + code, I am only committing the change to trunk. + +2007-12-28 18:26 +0000 [r95192] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 95191 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) | + 6 lines Remove duplicate increment of the header count in the + add_header() function. (closes issue #11648) Reported by: makoto + Patch provided by sergee, committed patch by me, inspired by + comments from putnopvut ........ + +2007-12-28 16:12 +0000 [r95167] Mark Michelson <mmichelson@digium.com> + + * apps/app_amd.c, CHANGES: Some changes to app_amd. The channel + name is printed in verbose messages maximumWordLength option + added. Duration of words that do not meet the minimum word + duration will be logged The duration of pre-greeting silence will + be logged Only consider us in the greeting if we actually + detected a valid word duration. (closes issue #11650, reported + and patched by davevg) + +2007-12-28 08:57 +0000 [r95139] Luigi Rizzo <rizzo@icir.org> + + * channels/console_video.c: fix a small bug in printing out + geometries - wrong input. + +2007-12-28 00:17 +0000 [r95096] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 95095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec + 2007) | 8 lines I found a bug while browsing the queue code and + managed to reproduce it in a small setup. If a queue uses the + ringall strategy, it was possible through unfortunate coincidence + for a single member at a given penalty level to make app_queue + think that all members at that penalty level were unavailable and + cause the members at the next penalty level to be rung. With this + patch, we will only move to the next penalty level if ALL the + members at a given penalty level are unreachable. ........ + +2007-12-27 23:32 +0000 [r95073] Luigi Rizzo <rizzo@icir.org> + + * apps/app_dictate.c, apps/app_mp3.c, apps/app_voicemail.c: remove + more unnecessary casts for NULL. main/say.c is a big offender in + this respect. + +2007-12-27 23:28 +0000 [r95070] Jason Parker <jparker@digium.com> + + * doc/asterisk.8, main/asterisk.c: Fix -s socket option, and + document it as well. Closes issue #11645, patch by Laureano. + +2007-12-27 23:13 +0000 [r95068-95069] Luigi Rizzo <rizzo@icir.org> + + * apps/app_ices.c, apps/app_queue.c, apps/app_voicemail.c: NULL + does not need to be cast to (char *) + + * channels/chan_oss.c: remove useless casts + +2007-12-27 21:41 +0000 [r95025] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 95024 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) | + 9 lines Don't report a syntax error when an empty string is + passed to ast_get_group. Just return 0. (closes issue #11540) + Reported by: tzafrir Patches: group_empty.diff uploaded by + tzafrir (license 46) -- slightly changed by me ........ + +2007-12-27 20:11 +0000 [r94978] Mark Michelson <mmichelson@digium.com> + + * /, main/io.c: Merged revisions 94977 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94977 | mmichelson | 2007-12-27 14:09:06 -0600 (Thu, 27 Dec + 2007) | 3 lines Fixing a typo in a comment. ........ + +2007-12-27 17:34 +0000 [r94908-94934] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_h323.c: Merged revisions 94924 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6 + lines Include types.h in chan_h323 as without it it can not be + compiled on some operating systems like FreeBSD to name one. + (closes issue #11585) Reported by: sobomax Patches: + chan_h323.c.diff uploaded by sobomax (license 359) ........ + + * /, channels/chan_sip.c: Merged revisions 94905 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94905 | file | 2007-12-27 13:27:11 -0400 (Thu, 27 Dec 2007) | 4 + lines Use ast_strlen_zero to see if our_contact is set or not on + the dialog. It is possible for it to be a pointer to NULL. + (closes issue #11557) Reported by: FuriousGeorge ........ + +2007-12-27 17:26 +0000 [r94904] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c, channels/console_video.c: more + localization of gui stuff + +2007-12-27 17:18 +0000 [r94903] Mark Michelson <mmichelson@digium.com> + + * doc/manager_1_1.txt: Adding documentation for new manager actions + and events in app_queue + +2007-12-27 16:51 +0000 [r94902] Luigi Rizzo <rizzo@icir.org> + + * CHANGES: clarify the type of video support in chan_oss + +2007-12-27 16:11 +0000 [r94830-94877] Russell Bryant <russell@digium.com> + + * codecs/codec_g722.c: I went looking for where we downloaded the + g722 implementation and came across these two links. So, I'm + adding them so they are available for reference later. + + * /: Blocked revisions 94831 via svnmerge ........ r94831 | russell + | 2007-12-27 09:16:56 -0600 (Thu, 27 Dec 2007) | 5 lines Now that + the contexts lock is a read/write lock, it should not be locked + here in ast_hint_state_changed(). This makes it get locked + recursively which now causes a deadlock. (closes issue #11080, + thanks to callguy for the access to a deadlocked machine) + ........ + + * /, main/translate.c, include/asterisk/translate.h: Merged + revisions 94828-94829 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) | + 9 lines Change ast_translator_best_choice() to only pay attention + to audio formats. This fixes a problem where Asterisk claims that + a translation path can not be found for channels involving video. + (closes issue #11638) Reported by: cwhuang Tested by: cwhuang + Patch suggested by cwhuang, with some additional changes by me. + ........ r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27 + Dec 2007) | 2 lines Use the constant that I really meant to use + here ... ........ + +2007-12-27 09:13 +0000 [r94826-94827] Olle Johansson <oej@edvina.net> + + * funcs/func_dialplan.c: This function checks more than just + contexts... + + * apps/app_pickupchan.c: - Add Copyright - Doxygen fixes Note: - + This application needs better documentation and a RESULT code in + the dialplan. + +2007-12-27 01:03 +0000 [r94825] Kevin P. Fleming <kpfleming@digium.com> + + * main/manager.c, /: Merged revisions 94824 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94824 | kpfleming | 2007-12-26 18:01:47 -0700 (Wed, 26 Dec 2007) + | 2 lines make this comment explain the situation in an even more + explicit fashion ........ + +2007-12-27 00:48 +0000 [r94819-94823] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c: more steps to decouple the gui from the + rest of the code. + + * channels/console_gui.c, channels/console_video.c, + channels/console_video.h: Enable building the code even if SDL is + not present (similarly, SDL is also detected at runtime). Now we + should be able to stream video even without a rendering device + (useful for remote monitoring). + + * channels/console_gui.c, channels/console_video.c: more + localizations around sdl_setup + + * channels/console_gui.c: use fread instead of mmap to read in the + comment area from the keypad. fread is simpler and more portable, + and there is no performance gain in using mmap. + + * images/kpad2.jpg: update the region description with an empty + line at the beginning. + +2007-12-26 22:38 +0000 [r94818] Tilghman Lesher <tlesher@digium.com> + + * build_tools/cflags.xml, channels/chan_zap.c: Allow more spans + than 32. Also, rearrange compiler flags so the most often used + flags appear closer to the top. Reported by: tzafrir Patch by: + tzafrir,tilghman (Closes issue #11528) + +2007-12-26 22:29 +0000 [r94817] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c, channels/console_video.c: another bunch + of gui localizations + +2007-12-26 22:14 +0000 [r94814] Jason Parker <jparker@digium.com> + + * apps/app_exec.c: Make 'else' argument to ExecIf optional. Clean + up the description and usage text a bit. Closes issue #11564, + patch by pnlarsson (with some extra cleanup by me). + +2007-12-26 22:10 +0000 [r94810-94813] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c, channels/console_video.c: more + localization of sdl stuff + + * channels/console_gui.c, channels/console_video.c, + channels/console_video.h: move more gui stuff into console_gui.c + +2007-12-26 20:49 +0000 [r94809] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, /: Merged revisions 94808 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94808 | tilghman | 2007-12-26 14:43:38 -0600 (Wed, 26 Dec 2007) + | 6 lines Workaround for what is probably a glibc bug (but we'll + see this crop up again and again, if we don't add the + workaround). Reported by: rolek Patch by: tilghman (Closes issue + #11601, closes issue #11426) ........ + +2007-12-26 20:02 +0000 [r94806] Jason Parker <jparker@digium.com> + + * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c, + apps/app_authenticate.c, apps/app_zapscan.c, apps/app_zapras.c, + apps/app_alarmreceiver.c, apps/app_amd.c, pbx/pbx_realtime.c, + pbx/pbx_dundi.c, apps/app_zapateller.c, pbx/pbx_config.c, + pbx/pbx_gtkconsole.c, apps/app_adsiprog.c, apps/app_cdr.c: Use + defined return values in load_module in more places. (closes + issue #11096) Patches: pbx_config.c.patch uploaded by moy + (license 222) pbx_dundi.c.patch uploaded by moy (license 222) + pbx_gtkconsole.c.patch uploaded by moy (license 222) + pbx_loopback.c.patch uploaded by moy (license 222) + pbx_realtime.c.patch uploaded by moy (license 222) + pbx_spool.c.patch uploaded by moy (license 222) + app_adsiprog.c.patch uploaded by moy (license 222) + app_alarmreceiver.c.patch uploaded by moy (license 222) + app_amd.c.patch uploaded by moy (license 222) + app_authenticate.c.patch uploaded by moy (license 222) + app_cdr.c.patch uploaded by moy (license 222) + app_zapateller.c.patch uploaded by moy (license 222) + app_zapbarge.c.patch uploaded by moy (license 222) + app_zapras.c.patch uploaded by moy (license 222) + app_zapscan.c.patch uploaded by moy (license 222) + +2007-12-26 20:01 +0000 [r94805] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c, channels/vcodecs.c, + channels/console_video.c, channels/console_video.h: more + preparation for untangling of the various console_video stuff + +2007-12-26 19:09 +0000 [r94796-94802] Russell Bryant <russell@digium.com> + + * main/autoservice.c, /: Merged revisions 94801 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) | + 4 lines Just in case the AST_FLAG_END_DTMF_ONLY flag was already + set before starting autoservice, remember it and ensure that the + channel has the same setting when autoservice gets stopped. + (pointed out by d1mas, patched up by me) ........ + + * funcs/func_dialplan.c (added), CHANGES: Add a new dialplan + function, DIALPLAN_EXISTS(), which allows you to check for the + existence of a dialplan target. (closes issue #11579) Reported + by: irroot Patches: func_dialplan2.c uploaded by irroot (license + 52) -- Additional changes by me. + + * main/autoservice.c, /: Merged revisions 94797 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) | + 4 lines When a channel is in autoservice, mark a flag on the + channel that says that we only care about the END of a digit. + That way, no magic digit emulation stuff will happen when all + we're doing is queueing up END frames. ........ + + * main/channel.c: Leave a note for a minor bug that was pointed out + by d1mas + +2007-12-26 18:05 +0000 [r94795] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_zap.c: Convert raw bits for callprogress bitfield + to use constants, for greater code clarity Reported by: dimas + Patch by: dimas (Closes issue #11280) + +2007-12-26 17:26 +0000 [r94787-94794] Russell Bryant <russell@digium.com> + + * /, res/res_features.c: Merged revisions 94793 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94793 | russell | 2007-12-26 11:24:17 -0600 (Wed, 26 Dec 2007) | + 3 lines Don't try to send a parked call back to itself. (closes + issue #11622, reported by djrodman, patched by me) ........ + + * Makefile, /: Merged revisions 94789 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94789 | russell | 2007-12-26 11:00:03 -0600 (Wed, 26 Dec 2007) | + 5 lines List include/asterisk/version.h as a .PHONY target + because we want the commands listed for this target to be + executed regardless of whether the file exists or not. This fixes + having the version not up to date when running from svn. (closes + issue #11619, reported by plack, fixed by me) ........ + + * main/autoservice.c, /: Merged revisions 94790 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) | + 5 lines Don't store DTMF BEGIN frames while a channel is in + autoservice. It's just going to make ast_read() do a lot of extra + work when the channel comes back out of autoservice. (closes + issue #11628, patched by me) ........ + + * channels/chan_iax2.c: Fix a bug in peer handling that caused + multiple instances of a peer to end up in the peers container + after a reload. Somehow, this bug doesn't exist in 1.4 ... + (closes issue #11626) (reported by pnlarsson, additional info + from mvanbaak, fixed by me) + + * utils: update svn:ignore for astcanary + +2007-12-26 15:58 +0000 [r94782] Mark Michelson <mmichelson@digium.com> + + * configs/extconfig.conf.sample, main/logger.c, CHANGES: Adding + support for storing the queue log entries in a realtime backend. + (closes issue #11625, reported and patched by sergee) Thank you + very much to sergee for adding this new feature! + +2007-12-26 10:14 +0000 [r94774] Luigi Rizzo <rizzo@icir.org> + + * channels/console_gui.c (added), channels/vcodecs.c (added), + channels/console_video.c: Split console_video.c so that video + codecs and gui functions are in separate files (still #include'd + because of tangling in the data structures, but this is going to + be cleaned up). The video grabbing functions still need to be + moved to a separate file. + +2007-12-25 04:10 +0000 [r94771-94773] Tilghman Lesher <tlesher@digium.com> + + * apps/app_pickupchan.c (added): Add pickup by channel (Closes + issue #11161) + + * channels/chan_zap.c, configs/zapata.conf.sample: Change the + abbreviated TON from 'A' to 'V', since 'A' is a legitimate DTMF + character. Also, fix the documentation to match the code. + + * res/res_agi.c: Add channel thread ID to the information passed to + AGI. Reported by: dror99 Patch by: tilghman (Closes issue #11162) + +2007-12-25 02:28 +0000 [r94770] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 94769 via svnmerge ........ r94769 | file | + 2007-12-24 22:27:08 -0400 (Mon, 24 Dec 2007) | 2 lines file + says... build on the builders. ........ + +2007-12-24 19:43 +0000 [r94764-94768] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 94767 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007) + | 5 lines Race: we need to wait to queue a NewChannel event until + after the channel is inserted into the channel list. The reason + is because some manager users immediately queue requests from the + channel when they see that event and are confused when Asterisk + reports no such channel. (Closes issue #11632) ........ + + * /: Blocked revisions 94765 via svnmerge ........ r94765 | + tilghman | 2007-12-24 10:17:01 -0600 (Mon, 24 Dec 2007) | 5 lines + More deadlock avoidance code (this time between sip_monitor and + sip_hangup) Reported by: apsaras Patch by: tilghman (Closes issue + #11413) ........ + + * /: Merged revisions 94763 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94763 | tilghman | 2007-12-24 09:39:56 -0600 (Mon, 24 Dec 2007) + | 5 lines Another bit of bad logic in realtime_peer Reported by: + dimas Patch by: dimas (Closes issue #11631) ........ + +2007-12-23 14:51 +0000 [r94713-94741] Luigi Rizzo <rizzo@icir.org> + + * channels/console_video.c, channels/console_video.h: support + sdl_videodriver to send output to x11/aalib/console + + * channels/console_video.c: move reading info from the keypad to a + separate function. Remove an unused keypad field and some + debugging messages. Adjust formatting on config file parsing + + * channels/console_video.c: make sure the minimum surface depth is + 16bpp so we can create YUVoverlays. With this change we can do + setenv SDL_VIDEODRIVER aalib and output to an ascii window (which + is still in an X11 window). If you also do unsetenv DISPLAY then + the output goes into the main asterisk window, unfortunately it + interferes with the normal output so you don't see much. In any + case, i don't think we are very far away from having a working + xterm videophone! + + * channels/Makefile: avoid rebuilding dependent files if the + generated busy.h and ringtone.h do not change. Ths masks (but + does not solve) a but that i am seeing in doing a 'gmake install' + without donig a 'gmake all' first. + +2007-12-23 01:38 +0000 [r94662] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 94660 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94660 | tilghman | 2007-12-22 19:21:03 -0600 (Sat, 22 Dec 2007) + | 2 lines Argh... I suppose third time's the charm. ........ + +2007-12-22 22:44 +0000 [r94615-94638] Luigi Rizzo <rizzo@icir.org> + + * configs/oss.conf.sample, channels/console_video.c: Change the + name of config file entries for keypad regions from + 'keypad_entry' to 'region'. Fix the example file accordingly. + Also make some fixes in the code do reset entries on reload of + the keypad. The recently committed kpad2.jpg has the correct + names. + + * images/kpad2.jpg (added): add a sample keypad (with annotations) + for console video + + * channels/console_video.c, channels/Makefile, channels/chan_oss.c, + channels/console_video.h (added): Build console_video support by + linking in, as opposed to including, console_video.c This will + ease the task of splitting console_video.c into its components + (V4L and X11 grabbers, various video codecs and packetizers, + SDL), as well as ease future extensions (e.g. additional video + sources, codecs and rendering engines). For the time being + nothing changes for users: video support is off by default, and + requires -DHAVE_VIDEO_CONSOLE on the command line to be included + (if SDL and FFMPEG are available). + +2007-12-21 21:19 +0000 [r94593] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Something I've been itching to do for a + while now. A minor optimization in app_voicemail. Since the + dtable in base_encode always gets populated with the same values + every time and never changes, make it static and const and only + initialize it once. Also, there's no reason to define + BASEMAXINLINE twice, so remove the redundant #define. + +2007-12-21 20:50 +0000 [r94549-94551] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: We should only clear this value if we have + to + + * channels/chan_zap.c: Commit non TCP transport part of #11506. + Includes numerous additional parameters, as well as RLT support + for DMS type switches + +2007-12-21 20:38 +0000 [r94542-94548] Mark Michelson <mmichelson@digium.com> + + * res/res_config_sqlite.c: Store dates using local time instead of + UTC (closes issue #11610, reported and patched by + rbraun_performatique) + + * apps/app_queue.c: Fix a memory leak when reloading queue rules. + + * CHANGES: The one documentation source I forgot to update after + the merge of the queue-penalty branch was the CHANGES file. No + longer! + + * apps/app_voicemail.c: Lots of coding guidelines cleanup. + + * /: Blocked revisions 94543 via svnmerge ........ r94543 | + mmichelson | 2007-12-21 14:21:59 -0600 (Fri, 21 Dec 2007) | 3 + lines Bunch of coding guidelines cleanup ........ + + * /, apps/app_voicemail.c: Merged revisions 94540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94540 | mmichelson | 2007-12-21 14:11:34 -0600 (Fri, 21 Dec + 2007) | 8 lines Better quota support for using IMAP storage + voicemail (closes issue #11415, reported by jaroth) (closes issue + #11152, reported by selsky) Patch provided by jaroth ........ + +2007-12-21 20:12 +0000 [r94541] Jason Parker <jparker@digium.com> + + * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c, + codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c, + codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_zap.c: + codecs.conf really shouldn't be mandatory.. it never had been + before, so let's go back to being optional. A big "thank you" to + pnlarsson on IRC for allowing me access to his system to debug + this. Closes issue #11584. + +2007-12-21 20:01 +0000 [r94477-94539] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 94538 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94538 | mmichelson | 2007-12-21 13:59:45 -0600 (Fri, 21 Dec + 2007) | 5 lines The mail_copy c-client function does not expect a + full imap mailbox string, just the name of the mailbox. (closes + issue #11419, reported and patched by jaroth, with additional + patchwork from me) ........ + + * main/dial.c: AST_LIST_REMOVE_CURRENT only takes one argument in + trunk + + * main/dial.c, /: Merged revisions 94468 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec + 2007) | 6 lines Since we are freeing list elements within a list + traversal, we need to use the safe traversal and remove the item + from the list before freeing it. (closes issue 11612, reported by + dtyoo) ........ + +2007-12-21 16:42 +0000 [r94467] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 94466 via svnmerge ........ r94466 | russell + | 2007-12-21 10:37:47 -0600 (Fri, 21 Dec 2007) | 6 lines Convert + the contexts lock to a read/write lock to resolve a deadlock. + This has a nice side benefit of improving performance. :) (closes + issue #11609) (closes issue #11080) ........ + +2007-12-21 16:12 +0000 [r94463-94465] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 94464 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94464 | mmichelson | 2007-12-21 10:11:44 -0600 (Fri, 21 Dec + 2007) | 3 lines Removing a debug message I accidentally just + committed ........ + + * /, main/say.c, apps/app_queue.c: Merged revisions 94420 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94420 | mmichelson | 2007-12-21 09:45:14 -0600 (Fri, 21 Dec + 2007) | 5 lines Fixing Portuguese syntax for saying dates and + times. Also some coding guidelines cleanup. (closes issue #11599, + reported and patched by caio1982, coding guidelines cleanup by + me) ........ + +2007-12-21 15:14 +0000 [r94419] Tilghman Lesher <tlesher@digium.com> + + * /, main/asterisk.c: Merged revisions 94418 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94418 | tilghman | 2007-12-21 09:07:42 -0600 (Fri, 21 Dec 2007) + | 2 lines Fix for restart-as-user problem reported via the -dev + list ........ + +2007-12-21 01:14 +0000 [r94345-94396] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Moved the update of the queue_ent's rule list + to just before we try to call queue members. This allows for the + change in penalty levels to be executed at the most logical time + frame. + + * configs/queues.conf.sample, doc/tex/channelvariables.tex, + apps/app_queue.c, configs/queuerules.conf.sample (added): Merging + the queue-penalty branch. In short, this allows one to + dynamically adjust the QUEUE_MAX_PENALTY and the newly introduced + QUEUE_MIN_PENALTY during a call depending on the amount of time + passed. The purpose is to allow the call to open up to more (or + maybe just different) members without the caller's losing his + place in the queue. See configs/queuerules.conf.sample for an + example of how to set up queue rules and + configs/queues.conf.sample for how to associate a rule with a + queue. Along with the functional changes, new CLI and manager + commands exist to show the rules defined and there is an + additional CLI command to reload the queue rules. Future + enhancements that may be made: support for realtime queue rules + and support for dynamically adding a rule through the manager or + CLI. Also a manager command to reload the queue rules (I'll + probably write this myself very soon). + + * apps/app_voicemail.c: The changes to header inclusion in trunk + broke compilation of app_voicemail when using IMAP storage. The + reason is that c-client has its own definitions for LOG_WARNING + and LOG_DEBUG, so we need to be sure to include asterisk's + definitions last so that we use the proper values in + app_voicemail. (closes issue #11437, reported by blitzrage, patch + suggested by blitzrage) + +2007-12-20 22:39 +0000 [r94320] Russell Bryant <russell@digium.com> + + * configs/zapata.conf.sample: Add a bit more to the description of + the "mwimonitor" option. + +2007-12-20 22:28 +0000 [r94319] Steve Murphy <murf@digium.com> + + * build_tools/make_buildopts_h: closes issue #11287; thanks to + snuffy for this fix, which will surely make all solaris owners + shout praises to his name. + +2007-12-20 20:25 +0000 [r94252-94257] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 94256 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r94256 | russell | 2007-12-20 14:22:22 -0600 + (Thu, 20 Dec 2007) | 13 lines Merged revisions 94255 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 + Dec 2007) | 5 lines Fix another potential seg fault ... (closes + issue #11606) Reported by: dimas ........ ................ + + * channels/chan_zap.c, /: Merged revisions 94251 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94251 | russell | 2007-12-20 14:08:42 -0600 (Thu, 20 Dec 2007) | + 10 lines Fix a deadlock in d-channel handling in chan_zap. This + deadlock was introduced by the fix to ensure that channels are + properly locked when handling channel variables. There were + sections of this code where the channel pvt was locked before the + channel lock, when in fact it _must_ be the other way around. + (closes issue #11582) Reported by: bugi ........ + +2007-12-20 12:56 +0000 [r94168-94191] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_usbradio.c, include/asterisk/config.h, + channels/console_video.c, channels/chan_oss.c: add some macros to + simplify parsing the config file, see description in config.h . + They are a variant of the set of macros i used in chan_oss.c, + structured in a way to be more robust to the presence of spurious + ';' - basically, they define wrappers for 'do {' and '} while + (0)', plus some helper functions to deal with simple cases such + as ast_copy_string, ast_malloc, strtoul, ast_true ... The prefix + (CV_ as 'Config Variable') tries to be easy to remember and has + been chosen to not conflict with other existing macros in the + tree. For the time being, I have only updated the three source + files in the tree that used the old M_* macros. Hopefully, more + files will be converted. NOTE: I understand that inventing my own + dialect of C is generally wrong; however, the lack of adequate + support in the language encourages lazy programming practices + (such as ignoring errors, bounds, etc.) and this increases the + chance of vulnerability in the code, especially because we are + parsing user input here. Hopefully, these macros and the use of + ast_parse_arg (in config.h) should encourage the programmer to + write more robust code. + + * include/asterisk/paths.h, res/snmp/agent.c, utils/ael_main.c, + utils/extconf.c, main/asterisk.c, utils/conf2ael.c: modify + http://svn.digium.com/view/asterisk?view=rev&rev=93603 so that + paths and filename are writable by asterisk.c without causing + segfaults. This involves defining the variables as const char *, + and having them point to as static, writable buffer defined in + asterisk.c On passing, fix some errors in using these variables + in some files in utils/ , and in res/snmp/agent.c which was + redefining a variable without using paths.h (not applicable to + 1.4) + +2007-12-19 23:17 +0000 [r94123-94124] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: 1. Unify the check for a penalty < 0 into the + set_member_penalty code. 2. Fix an error when checking the CLI + command for setting a member's penalty. 3. Fix a logging error if + the incorrect parameter was the queue name or interface. (closes + issue #11544, reported and patched by Laureano) + + * /, res/res_monitor.c: Merged revisions 94122 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec + 2007) | 6 lines Sox versions 13.0.0 and newer do not have + "soxmix" and instead use sox -m. res_monitor needs to use this if + the user does not have soxmix. (closes issue #11589, reported by + amessina, patch inspired by amessina but with a flourish from me) + ........ + +2007-12-19 22:51 +0000 [r94085] Russell Bryant <russell@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 94077 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r94077 | russell | 2007-12-19 16:48:48 -0600 (Wed, 19 Dec 2007) | + 4 lines Check for the existence of the soxmix application on the + target platform and have the result available in autoconfig.h. + (part of issue #11589) ........ + +2007-12-19 20:20 +0000 [r94052-94053] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Add 'voicemail reload' command. Reported + by: eliel Patch by: eliel (Closes issue #11365) + + * apps/app_waituntil.c (added): Add contributed WaitUntil app. + Original code by pprindeville, updated for trunk by tilghman. + (Closes issue #11487) + +2007-12-19 19:29 +0000 [r94029] Russell Bryant <russell@digium.com> + + * include/asterisk/time.h: Add a couple of new time API calls - + ast_tvdiff_sec and ast_tvdiff_usec (closes issue #11270) Reported + by: dimas Patches: tvdiff_us-4.patch uploaded by dimas (license + 88) + +2007-12-19 17:58 +0000 [r94002] Luigi Rizzo <rizzo@icir.org> + + * channels/console_video.c: Add instructions on how to generate + your own font. + +2007-12-19 17:31 +0000 [r93956] Joshua Colp <jcolp@digium.com> + + * /: Merged revisions 93955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r93955 | file | 2007-12-19 13:29:20 -0400 (Wed, 19 Dec 2007) | 2 + lines Make the 1.4 builders happy, ensure var is NULL. ........ + +2007-12-19 17:13 +0000 [r93952] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 93949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r93949 | tilghman | 2007-12-19 11:04:13 -0600 (Wed, 19 Dec 2007) + | 3 lines Avoid segfault in chan_iax when peer isn't defined + (Closes issue #11602) ........ + +2007-12-19 17:09 +0000 [r93925-93950] Luigi Rizzo <rizzo@icir.org> + + * main/utils.c, include/asterisk/strings.h: Add a new API function, + written at least twice in app_voicemail.c and likely in other + places too. This is quite useful when placing mail/html stuff in + config files. /*! \brief Convert some C escape sequences + (\b\f\n\r\t) into the equivalent characters. \brief s The string + to be converted (will be modified). \return The converted string. + */ char *ast_unescape_c(char *s); + + * include/asterisk/config.h, main/config.c: add support for + PARSE_DOUBLE, and remove identifiers for types not supported + (INT16 and UINT16) + +2007-12-19 09:20 +0000 [r93899] Olle Johansson <oej@edvina.net> + + * CHANGES: Reorganize CHANGES a bit. The "misc" section grew too + large... + +2007-12-19 08:57 +0000 [r93898] Luigi Rizzo <rizzo@icir.org> + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + acinclude.m4, makeopts.in: Properly document AST_EXT_TOOL_CHECK() + and use it to check for NETSMP and GTK (GTK is not used thoug). + AST_EXT_TOOL_CHECK() could be used for checking curl status as + well, perhaps with a small addition because we currently seem to + require a curl version greater than X.Y.Z Add a NETSMP_INCLUDE + entry in makeopts.in We don't have yet any macros for using + pkg-config to check for a specific package (right now there is + only gtk2+ in the category). + +2007-12-19 08:57 +0000 [r93897] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding the + ability to specify the To: header in an outbound INVITE by adding + an exclamation mark to the dial string. This patch also exists + for 1.4 in the fixtoheader-1.4 branch and has been in production + for quite some time. + +2007-12-19 08:12 +0000 [r93875] Luigi Rizzo <rizzo@icir.org> + + * res/snmp/agent.c: make netsmp build under AST_DEVMODE. + Description, included in the source, is below. I should note that + the PACKAGE_* macros that asterisk defines in autoconfig.h are + not used anywhere in the tree so they should just be removed. /* + * There is some collision collision between netsmp and asterisk + names, * causing build under AST_DEVMODE to fail. * * The + following PACKAGE_* macros are one place. * Also netsnmp has an + improper check for HAVE_DMALLOC_H, using * #if HAVE_DMALLOC_H + instead of #ifdef HAVE_DMALLOC_H * As a countermeasure we define + it to 0, however this will fail * when the proper check is + implemented. */ No + +2007-12-19 07:01 +0000 [r93854] Olle Johansson <oej@edvina.net> + + * CHANGES, main/asterisk.c, doc/asterisk.sgml: Add option for + starting remote Asterisk by naming the actual runtime socket + instead of pointing to configuration file with -C Reported by: + sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax + (license 359) doc changes by committer (closes issue #11598) + +2007-12-19 00:09 +0000 [r93827] Dwayne M. Hubbard <dhubbard@digium.com> + + * apps/app_osplookup.c: add missing header file + +2007-12-18 23:38 +0000 [r93804-93805] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c: Making the canary error message a little more + obvious. + + * utils/Makefile, utils/astcanary.c (added), main/asterisk.c: Add a + canary process, for high priority mode (asterisk -p) to ensure + that if Asterisk goes into a busy loop, the machine will be + recoverable. We'd still need to do a restart to put Asterisk back + into high priority mode, but at least a reboot won't be required. + (Closes issue #11559) + +2007-12-18 22:44 +0000 [r93765] Jason Parker <jparker@digium.com> + + * /: Blocked revisions 93764 via svnmerge ........ r93764 | qwell | + 2007-12-18 16:42:41 -0600 (Tue, 18 Dec 2007) | 4 lines FreeBSD + also does not have byte swap functions. Issue 11586, patch by + sobomax. ........ + +2007-12-18 21:13 +0000 [r93741] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Move some warnings away to debug since some + devices send a packet with a silly string as a NAT keepalive + packet. + +2007-12-18 18:39 +0000 [r93672] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions + 93668 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r93668 | tilghman | 2007-12-18 12:29:39 -0600 + (Tue, 18 Dec 2007) | 10 lines Merged revisions 93667 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 + Dec 2007) | 2 lines Fixing AST-2007-027 (Closes issue #11119) + ........ ................ + +2007-12-18 18:20 +0000 [r93666] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/paths.h: remove a leftover line with only a '#' + (wonder why the compiler does not complain!) and variables that + are only used in asterisk.c + +2007-12-18 17:05 +0000 [r93626] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 93625 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec + 2007) | 6 lines Rework deadlock avoidance used in ast_write, + since it meant that agent channels which were being monitored had + one audio file recorded and one empty audio file saved. (closes + issue #11529, reported by atis patched by me) ........ + +2007-12-18 10:24 +0000 [r93558-93603] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/paths.h, channels/chan_sip.c, res/res_crypto.c, + utils/ael_main.c, utils/extconf.c, main/asterisk.c, + res/res_monitor.c, utils/conf2ael.c: make configuration variable + const so they are not accidentally modified. This requires + casting the strings in asterisk.c when writing to them, so we do + it through a macro to do it consistently. + + * channels/chan_unistim.c, res/res_crypto.c, main/astmm.c, + apps/app_ices.c, utils/extconf.c, channels/chan_iax2.c, + main/asterisk.c, main/config.c, main/db.c, apps/app_adsiprog.c, + cdr/cdr_csv.c: remove unnecessary (char *) casts for + ast_config_AST_* variables. There are some left in the .flex + files, left to the maintainer... + + * build_tools/make_defaults_h, main/asterisk.c: Rename the macros + in defaults.h - they are not meant to be globally visible. + Document the fact that DEFAULT_TMP_DIR cannot be overridden from + the default configuration (this needs to be fixed, as you could + have a totally different spooldir configured at runtime, and yet + DEFAULT_TMP_DIR keeps the compile-time default). Remove two + unused entries for sounds and images. + + * Makefile.moddir_rules: make the code match documentation - now + you can specify multiple words in MODULE_PREFIX. + + * CREDITS: Name the people responsible for some recent + contributions to the tree. + + * Makefile: Two small changes: + document the difference between + "A=foo make ..." and "make A=foo ..." and suggest using + COPTS/LDOPTS if you need to use the second form to pass compiler + and loader flags; + define only in one place the environment used + to build stuff in menuselect/ + +2007-12-18 07:56 +0000 [r93557] Olle Johansson <oej@edvina.net> + + * doc/CODING-GUIDELINES: A minor update, caused by a recent bug + report ;-) + +2007-12-18 07:22 +0000 [r93536] Luigi Rizzo <rizzo@icir.org> + + * doc/CODING-GUIDELINES: small documentation update (nothing + important). + +2007-12-18 02:57 +0000 [r93514] Joshua Colp <jcolp@digium.com> + + * channels/chan_unistim.c: You... will... build! I say so and + therefore you will. + +2007-12-18 02:42 +0000 [r93493] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_unistim.c, include/asterisk/threadstorage.h: minor + cleanups + +2007-12-17 23:10 +0000 [r93464] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_unistim.c: fix building under cygwin. At this point + WINARCH should go away. + +2007-12-17 22:57 +0000 [r93424] Jason Parker <jparker@digium.com> + + * /: Blocked revisions 93420 via svnmerge ........ r93420 | qwell | + 2007-12-17 16:56:58 -0600 (Mon, 17 Dec 2007) | 1 line Missed a + spot.. ........ + +2007-12-17 22:54 +0000 [r93405] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_unistim.c: remove some unnecessary includes + +2007-12-17 22:50 +0000 [r93390] Jason Parker <jparker@digium.com> + + * /, main/translate.c: Merged revisions 93381 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4 + lines What was I thinking when I wrote this masterpiece? -1 + 1 = + 0.. who woulda thunk it?. ........ + +2007-12-17 22:38 +0000 [r93380] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_oss.c: surprising as it may be, chan_oss compiles + correctly under cygwin as well, provided you look for soundcard.h + in the right place... + +2007-12-17 22:29 +0000 [r93378] Joshua Colp <jcolp@digium.com> + + * /, main/utils.c: Merged revisions 93377 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7 + lines Do not try to access information about a lock when printing + out a trylock attempt. It is possible for the lock that it + references to no longer be valid. This would have caused + segfaults or deadlocks. (issue #BE-263) (closes issue #11080) + Reported by: callguy (closes issue #11100) Reported by: callguy + ........ + +2007-12-17 21:14 +0000 [r93337] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/time.h: Merged revisions 93336 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r93336 | tilghman | 2007-12-17 15:12:42 -0600 (Mon, 17 + Dec 2007) | 6 lines Today is tomorrow's yesterday, and + yesterday's tomorrow is today, and tomorrow's tomorrow is the day + after tomorrow, so who cares if you recycle anyway? If this + confuses you, that's nothing compared to what this fixes. ;-) + ........ + +2007-12-17 21:12 +0000 [r93335] Olle Johansson <oej@edvina.net> + + * channels/chan_zap.c, /, channels/chan_sip.c, apps/app_queue.c, + channels/chan_iax2.c, channels/chan_mgcp.c: Merged revisions + 93182 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8 + lines Issue 11574: Add dependencies on res_monitor and + res_features. I wonder if Asterisk can run at all without + res_features. My guess is that there's propably a lot of more + modules and the core that depends on it. Reported by: caio1982 + (closes issue #11574) ........ + +2007-12-17 20:42 +0000 [r93293-93297] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Removing some leftover debug messages from a + while back. + + * /, apps/app_voicemail.c: Merged revisions 93291 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec + 2007) | 6 lines We need to create the directory for a voicemail + user even if they are using IMAP storage since greetings are + stored in the filesystem. (closes issue #11388, reported by + spditner, patch by me inspired by a patch by spditner) ........ + +2007-12-17 18:07 +0000 [r93252] Joshua Colp <jcolp@digium.com> + + * channels/chan_zap.c, /: Merged revisions 93250 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r93250 | file | 2007-12-17 14:05:55 -0400 (Mon, 17 Dec 2007) | 6 + lines If a call is received with a called number IE containing + nothing go to the 's' extension. (closes issue #9099) Reported + by: kb1_kanobe2 Patches: 20070906__9099.diff.txt uploaded by + Corydon76 (license 14) ........ + +2007-12-17 17:16 +0000 [r93191-93224] Kevin P. Fleming <kpfleming@digium.com> + + * utils: all created files need to be listed in the ignore property + + * channels/chan_unistim.c, build_tools/menuselect-deps.in, + configure, configure.ac, channels/Makefile, channels/chan_oss.c: + make the configure script detect that it is running on a Windows + platform, and report that information so that menuselect can use + it (all information that is used to decide whether to build + modules or not must be fed to menuselect so the user knows what + will be built and why... don't make module build decisions in the + makefiles, please) + + * Makefile: make using PRINT_DIR a little easier + +2007-12-17 15:18 +0000 [r93187-93190] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Fix usage of rtptimeout. It can be used + without rtpkeepalive, and the value can not be accessed directly + in the SIP pvt structure. All RTP related timeouts have to be + retrieved using the ast_rtp_* function calls. (closes issue + #11562) Reported by: ibc + + * channels/chan_unistim.c: If no timezone is available use the + default message. (closes issue #11576) Reported by: junky + + * channels/chan_unistim.c: Make chan_unistim actually be able to + unload. When creating a thread that you want to pthread_join you + have to explicitly create it as joinable, and also if using + pthread_cancel you have to have a pthread_testcancel to see if it + has been called. + +2007-12-17 07:27 +0000 [r93184-93185] Kevin P. Fleming <kpfleming@digium.com> + + * /: Blocked revisions 93183 via svnmerge ........ r93183 | + kpfleming | 2007-12-16 23:21:08 -0800 (Sun, 16 Dec 2007) | 2 + lines fix some copy-and-paste leftovers ........ + + * codecs, /, build_tools/make_version, + include/asterisk/autoconfig.h.in, configure.ac, apps, + Makefile.moddir_rules, res/Makefile, pbx/Makefile, + build_tools/prep_moduledeps (removed), channels/Makefile, cdr, + formats, Makefile, codecs/Makefile, funcs, apps/Makefile, + configure, build_tools/embed_modules.xml, cdr/Makefile, + build_tools/prep_tarball, makeopts.in, formats/Makefile, res, + pbx, channels, funcs/Makefile: Merged revisions 93180 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) + | 23 lines In + http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html, + rizzo brought up some issues related to the way that the metadata + required for menuselect and the rest of the build system is + extracted from the source files. Since I had a few hours to kill + on an airplane today, I decided to improve this situation... so + now the system caches the extracted metadata and uses it to build + the menuselect 'tree' as much as it can. The result of this is + that when a single source file is changed, only the metadata for + that file needs to be extracted again, and the rest is used from + the cache files. I also reduced the number of forked processes + required to do the metadata extraction; it was actually possible + to do most of what we needed in the Makefiles themselves without + using any shell scripts at all! On my laptop, these changes + resulted in an 80% decrease in the time required for the + 'menuselect.makeopts' automatic check to occur after editing a + single source file. While doing this work I also cleaned up a few + minor things in the Makefiles, adding a check for 'awk' to the + configure script and changed all remaining places we use 'grep' + or 'awk' to use the ones found by the configure script, and + changed the 'prep_tarball' script to build the menuselect + metadata so that tarballs of Asterisk will include it and won't + require the user to wait while it is extracted after unpacking. + ........ + +2007-12-16 19:06 +0000 [r93173] Luigi Rizzo <rizzo@icir.org> + + * Makefile: menuselect.makeopts is not a .PHONY target + +2007-12-16 13:38 +0000 [r93163-93167] Olle Johansson <oej@edvina.net> + + * pbx/pbx_dundi.c: Convert from LOG_DEBUG etc to ast_debug. Thanks, + dimas! (closes issue #11572) Reported by: dimas Patches: + dundilog-trunk.patch uploaded by dimas (license 88) + + * main/manager.c, CHANGES: Adding a new CLI command for "manager + reload", which is important now that you need to reload after + changes. Thanks YS. Reported by: ys Patches: + trunk93163_manager_reload.c.diff uploaded by ys (license 281) + (related to issue #11414) + + * main/manager.c, CHANGES: Change manager so that registered + accounts are stored in memory. This opens for a manager realtime + implementation. If you change accounts in manager.conf, you now + need to reload to activate the changes (deletions, additions). + This was not the case with 1.4. Reported by: ys Patches: + trunk93163_manager_reload.c.diff uploaded by ys (license 281) + (closes issue #11414) + + * CHANGES: Adding console_video to CHANGES. It's important that we + keep this file up to date, even with experimental stuff. + + * channels/chan_unistim.c, main/udptl.c, configs/dundi.conf.sample, + channels/chan_sip.c, include/asterisk/rtp.h, + include/asterisk/netsock.h, channels/iax2-provision.c, + UPGRADE.txt, doc/tex/qos.tex, configs/skinny.conf.sample, + CHANGES, channels/chan_iax2.c, main/rtp.c, main/netsock.c, + configs/h323.conf.sample, configs/iax.conf.sample, + channels/chan_skinny.c, configs/mgcp.conf.sample, + configs/unistim.conf.sample, channels/chan_h323.c, + configs/iaxprov.conf.sample, pbx/pbx_dundi.c, + configs/sip.conf.sample, channels/chan_mgcp.c: HUGE improvements + to QoS/CoS handling by IgorG - Refer to the proper documentation + - Implement separate signalling/media QoS/CoS in many channels + using RTP - Improve warnings and verbose messages - Deprecate + some old settings Minor modifications by me, a big effort from + IgorG. Thanks! Reported by: IgorG Patches: + qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) + Tested by: IgorG (closes issue #11145) + +2007-12-16 10:34 +0000 [r93162] Luigi Rizzo <rizzo@icir.org> + + * Makefile: use a simpler idiom for 'cmp -s ...' + +2007-12-16 09:37 +0000 [r93152-93161] Olle Johansson <oej@edvina.net> + + * main/asterisk.c: Don't drop the first character randomly in long + listings in the CLI. Reported by: slavon Patches: + asterisk-consolerefresh2.diff.txt uploaded by jamesgolovich + (license 176) Tested by: eliel (closes issue #9325) + + * configs/sip.conf.sample, CHANGES: Update documentation + + * channels/chan_sip.c, configs/sip.conf.sample: Make more timers + settable in SIP so that we can force timeout earlier on + non-responsive SIP servers. Thanks, jcmoore, for the patch! + Reported by: jcmoore Patches: peer_t1_timerb_trunk_v3.patch.txt + uploaded by jcmoore (license 9) (closes issue #9771) + + * include/asterisk/file.h: Typo fixed earlier, that wasn't a typo + after all. Didn't a clever guy once say "Compile before you + commit" ? :-) + +2007-12-15 08:10 +0000 [r93151] Russell Bryant <russell@digium.com> + + * include/asterisk/file.h: fix a typo from revision 93138 + +2007-12-15 00:44 +0000 [r93138-93145] Luigi Rizzo <rizzo@icir.org> + + * configs/oss.conf.sample: configuration options related to video + support. + + * channels/console_video.c (added): Bring in video console support + for chan_oss (and later chan_alsa too). This is disabled in the + default build, you need to explicitly enable it compiling with + make COPTS=-DHAVE_VIDEO_CONSOLE In return, you will be able to do + a video call with chan_oss, using the webcam (or X11 grabbing) as + local source, and rendering the incoming stream on your screen. + Currently supported formats are h261, h263, h263+, h264, mpeg4 + (all through the avcodec lib, part of ffmpeg). Incoming video is + on the left, outgoing video is on the right, while the center + displays a keypad (if configured so). Right clicking on the video + windows increases the size, center clicking reduces the size. + Dragging the mouse (with the left key) on the right window while + the X11 grabber is active moves the grab area. This is the result + of work by Sergio Fadda, Marta Carbone and myself, all properly + disclaimed to digium. Note, there is a lot of work left to do in + this module, including adding support for Video4LinuxV2 (I have + patches from Matteo Brancaleoni which should be integrated), and + making the GUI a lot more friendly than it is now (e.g. + supporting merging or switching among multiple sources, a text + window, and more). + + * channels/chan_oss.c: remove some redundant headers + + * include/asterisk/file.h: include mmap header if detected by + configure + +2007-12-14 22:02 +0000 [r93094-93115] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Resolve a compiler warning + + * apps/app_voicemail.c: Change places where the name "INBOX" was + hardcoded to use the imapfolder setting from voicemail.conf + instead. This commit will help to get issue #11415 moving towards + commitment. + +2007-12-14 21:09 +0000 [r93090] Tilghman Lesher <tlesher@digium.com> + + * Makefile, channels/chan_unistim.c, codecs/ilbc/iLBC_define.h: + Solaris compat fixes Reported by: snuffy Patch by: + snuffy,tilghman (Closes issue #11315) + +2007-12-14 19:31 +0000 [r93067] Russell Bryant <russell@digium.com> + + * pbx/pbx_dundi.c: make something static + +2007-12-14 19:27 +0000 [r93066] Tilghman Lesher <tlesher@digium.com> + + * apps/app_privacy.c, UPGRADE.txt, CHANGES, + configs/privacy.conf.sample (removed): Remove use of privacy.conf + by the Privacy app. Reported by: eliel Patch by: eliel (Closes + issue #11344) + +2007-12-14 19:19 +0000 [r93042-93065] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c, main/manager.c, funcs/func_timeout.c: I needed to + increment the numbers used on the VERBOSITY_ATLEAST calls by 1. + Thanks to kpfleming for pointing this out. + + * include/asterisk/logger.h, main/pbx.c, main/manager.c, + funcs/func_timeout.c: Changed VERBOSITY_LEVEL to + VERBOSITY_ATLEAST to be more accurate. + + * include/asterisk/logger.h, main/pbx.c, main/manager.c, + funcs/func_timeout.c, main/logger.c: After reading Russell's + e-mail to the dev list stating that checking option_verbose is + not equivalent to the check done by ast_verb, I wrote a macro, + VERBOSITY_LEVEL, which does this check. I did a quick look in the + source and used this macro in some places where option_verbose + was used. I also converted some verbose messages in logger.c to + use ast_verb instead of ast_verbose. + +2007-12-14 18:24 +0000 [r93041] Tilghman Lesher <tlesher@digium.com> + + * apps/app_meetme.c: gcc 4.1.3 wants a union used here. + +2007-12-14 17:49 +0000 [r93001-93004] Russell Bryant <russell@digium.com> + + * main/config.c: Print an error message if a #included file does + not exist + + * /: Blocked revisions 93000 via svnmerge ........ r93000 | russell + | 2007-12-14 11:36:08 -0600 (Fri, 14 Dec 2007) | 7 lines There + are a lot of existing systems that #include non-existent files. + So, to make the transition to treating this as an error a bit + less painless, just issue a huge error message for now. Then, + later, we can reinstate the code that treats it as a failure. + (Thanks to philippel for the feedback) ........ + +2007-12-14 17:29 +0000 [r92999] Tilghman Lesher <tlesher@digium.com> + + * res/res_agi.c: Publish the AGI events to manager. Reported by: + moy Patch by: moy,tilghman (Closes issue #11337) + +2007-12-14 15:59 +0000 [r92976] Mark Michelson <mmichelson@digium.com> + + * funcs/func_timeout.c: Reintroduce an optimization that was lost + when converting trunk to use ast_verb. + +2007-12-14 15:49 +0000 [r92939] Tilghman Lesher <tlesher@digium.com> + + * main/editline/sys.h: If malloc.h is included in a Solaris build, + the compilation breaks. Reported by: snuffy Patch by: snuffy + (Closes issue #11313) + +2007-12-14 15:18 +0000 [r92938] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 92937 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92937 | file | 2007-12-14 11:16:15 -0400 (Fri, 14 Dec 2007) | 4 + lines Up the length of the format on the SIP channel since it can + now be rather long. (closes issue #11552) Reported by: + francesco_r ........ + +2007-12-14 15:14 +0000 [r92936] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 92933 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92933 | tilghman | 2007-12-14 09:01:10 -0600 (Fri, 14 Dec 2007) + | 5 lines Change help documentation to match actual behavior + (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman + (Closes issue #11548) ........ + +2007-12-14 15:08 +0000 [r92935] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /: Merged revisions 92934 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r92934 | crichter | 2007-12-14 16:05:28 +0100 (Fr, 14 + Dez 2007) | 1 line fixed the sequencing of WAITING_4DIGS state + setting and overlap_task thread starting. ........ + +2007-12-14 14:48 +0000 [r92913] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, main/pbx.c, main/srv.c, channels/chan_skinny.c, + res/res_features.c, apps/app_minivm.c, apps/app_amd.c, + res/snmp/agent.c, apps/app_chanspy.c, apps/app_mixmonitor.c, + main/asterisk.c, main/netsock.c, apps/app_voicemail.c: Convert + ast_verbose to ast_verb. Reported by: snuffy Patch by: snuffy + (Closes issue #11547) + +2007-12-14 01:25 +0000 [r92876] Mark Michelson <mmichelson@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 92875 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r92875 | mmichelson | 2007-12-13 19:24:06 -0600 (Thu, 13 + Dec 2007) | 7 lines When compiling with DETECT_DEADLOCKS, don't + spam the CLI with messages about possible deadlocks. Instead just + print the intended single message every five seconds. (closes + issue 11537, reported and patched by dimas) ........ + +2007-12-13 23:10 +0000 [r92816-92855] Tilghman Lesher <tlesher@digium.com> + + * apps/app_meetme.c: When working with dates, use numeric form + whenever possible, as it's faster. Also, a bunch of coding + guidelines fixes. + + * channels/chan_zap.c, /: Merged revisions 92815 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92815 | tilghman | 2007-12-13 15:28:39 -0600 (Thu, 13 Dec 2007) + | 5 lines Properly initialize polarity statuses, so that they are + detected properly. Reported by: julianjm Patch by: julianjm + (Closes issue #10238) ........ + +2007-12-13 20:23 +0000 [r92811] Joshua Colp <jcolp@digium.com> + + * include/asterisk/app.h, include/asterisk/module.h, res/res_agi.c, + apps/app_rpt.c: Move usage of the old LOCAL_USER_* macros to the + new ast_module_user_* functions in a few documentation places. + (closes issue #11533) Reported by: IgorG Patches: + oldmacroclean.v1.diff uploaded by IgorG (license 20) + +2007-12-13 20:14 +0000 [r92810] Jason Parker <jparker@digium.com> + + * main/pbx.c, /: Merged revisions 92809 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92809 | qwell | 2007-12-13 14:13:48 -0600 (Thu, 13 Dec 2007) | 1 + line Make application help text a little more clear about the use + of extensions in a filename. ........ + +2007-12-13 20:12 +0000 [r92806-92808] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 92807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92807 | mmichelson | 2007-12-13 14:03:20 -0600 (Thu, 13 Dec + 2007) | 3 lines Prevent another potential fd leak ........ + + * /, apps/app_voicemail.c: Merged revisions 92803 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92803 | mmichelson | 2007-12-13 13:49:55 -0600 (Thu, 13 Dec + 2007) | 3 lines Prevent a possible fd leak. ........ + +2007-12-13 17:46 +0000 [r92779] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_adaptive_odbc.c: Don't use backslash as an escape + character, unless it really is an escape character. + +2007-12-13 16:23 +0000 [r92758] Jason Parker <jparker@digium.com> + + * channels/chan_sip.c: Remove remnants of a poorly merged commit. + (92697) + +2007-12-13 15:40 +0000 [r92737] Doug Bailey <dbailey@digium.com> + + * apps/app_voicemail.c: Tag voicemails with UTC time as opposed to + local time zone + +2007-12-13 00:18 +0000 [r92697] Jason Parker <jparker@digium.com> + + * /, channels/chan_sip.c, channels/chan_h323.c, main/config.c: + Merged revisions 92696 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #10690) ........ r92696 | qwell | 2007-12-12 18:11:09 -0600 + (Wed, 12 Dec 2007) | 7 lines If a typo is found in a config file, + we previous continued on with what was already loaded. We do not + want to do this (see bug below for details). This makes it so + that if a [ is found without a ], the entire config will fail, + and nothing in it will be loaded. Issue 10690. ........ + +2007-12-12 23:44 +0000 [r92676] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c: Revert an "optimization" that I added in + revision 89887, as the user who reported issue #11449 has + demonstrated that it actually was a performance hit on his + machine. I think that it is possible that it could still be a + benefit on systems under higher load, especially SMP systems, but + I don't have enough time or interest to find out at the moment. + (closes issue #11449) + +2007-12-12 21:22 +0000 [r92618] Jason Parker <jparker@digium.com> + + * /, apps/app_meetme.c, channels/ringtone.h: Merged revisions 92617 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11048) ........ r92617 | qwell | 2007-12-12 15:15:45 -0600 + (Wed, 12 Dec 2007) | 4 lines Don't increment user count until + after name has been recorded (if enabled). Issue 11048, tested by + pep. ........ + +2007-12-12 20:05 +0000 [r92594] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, main/logger.c, main/utils.c, + apps/app_mixmonitor.c: Conversions of free to ast_free, where + applicable, and several other formatting fixes. Reported by: + eliel Patch by: eliel,tilghman (Closes issue #11209) + +2007-12-12 19:50 +0000 [r92562] Russell Bryant <russell@digium.com> + + * res/res_features.c: Merged revisions 92556 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92556 | russell | 2007-12-12 13:40:02 -0600 (Wed, 12 Dec 2007) | + 1 line resolve compiler warning ........ + +2007-12-12 17:51 +0000 [r92511-92526] Mark Michelson <mmichelson@digium.com> + + * res/res_features.c: Same change to trunk as revision 92510. I'm + not sure why I merged this way, but I did. + + * /: Blocked revisions 92510 via svnmerge ........ r92510 | + mmichelson | 2007-12-12 11:46:14 -0600 (Wed, 12 Dec 2007) | 7 + lines Correctly detect where a dynamic feature was activated. + Before this patch, the channel which initiated the bridge was + always assumed to have been the one which activated the dynamic + feature. This patch corrects this. (closes issue #11529, reported + and patched by nic_bellamy) ........ + +2007-12-12 17:15 +0000 [r92476-92507] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c: Correctly handle possible memory allocation + failure Reported by: eliel Patch by: eliel (Closes issue #11512) + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 92463 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92463 | tilghman | 2007-12-12 10:52:56 -0600 (Wed, 12 Dec 2007) + | 4 lines Test directly for the API that fixed AST-2007-026, to + ensure that older versions of PostgreSQL are no longer + acceptable. (Closes issue #11526) ........ + +2007-12-12 16:11 +0000 [r92444] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 92443 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92443 | mmichelson | 2007-12-12 10:08:55 -0600 (Wed, 12 Dec + 2007) | 3 lines Removing an unused variable. ........ + +2007-12-11 22:20 +0000 [r92423] Olle Johansson <oej@edvina.net> + + * include/asterisk/term.h, channels/misdn/isdn_msg_parser.c, + channels/ringtone.h, include/asterisk/ulaw.h, + include/jitterbuf.h, include/asterisk/manager.h, + include/asterisk/transcap.h, channels/misdn/isdn_lib.c, + channels/gentone.c, include/asterisk/zapata.h, + channels/misdn/isdn_lib.h, include/asterisk/doxyref.h, + channels/DialTone.h, channels/misdn/ie.c, + channels/misdn/chan_misdn_config.h, channels/iax2.h, + channels/misdn/portinfo.c, include/asterisk/udptl.h, + main/cygload.c, include/asterisk/translate.h: Doxygen updates, + formatting. misdn stuff needs a lot of doxygenification (Hello, + Qwell :-) ) + +2007-12-11 22:10 +0000 [r92422] Mark Michelson <mmichelson@digium.com> + + * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, + configure.ac: Trunk build would fail due to the nonexistence of + zaptel hwgain structures missing. Patched configure to check for + this stuff and put a #ifdef around the offending code in + chan_zap. Thanks to file for overseeing this. + +2007-12-11 21:58 +0000 [r92421] Jason Parker <jparker@digium.com> + + * channels/chan_sip.c: We need to set the address we want to match + against before we actually do the match.. Closes issue #11518. + +2007-12-11 21:46 +0000 [r92402] Mark Michelson <mmichelson@digium.com> + + * res/res_musiconhold.c: Removing a pointless memset. The memory + was just calloc'd, so the memory is already zeroed out + +2007-12-11 21:17 +0000 [r92401] Jason Parker <jparker@digium.com> + + * apps/app_controlplayback.c: Add variable to show which key was + pressed to stop playback. Issue #11377, initial patch by johan. + +2007-12-11 20:06 +0000 [r92364-92365] Joshua Colp <jcolp@digium.com> + + * res/res_monitor.c: Only look to see if options are set if some + have been provided. (closes issue #11505) Reported by: Mike + Anikienko + + * main/global_datastores.c, /: Merged revisions 92363 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r92363 | file | 2007-12-11 15:51:40 -0400 (Tue, 11 Dec + 2007) | 6 lines Fix potential memory leak with the dialed + interfaces list if another memory allocation fails. (closes issue + #11507) Reported by: eliel Patches: global_datastores.c.patch + uploaded by eliel (license 64) ........ + +2007-12-11 17:44 +0000 [r92324] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 92323 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec + 2007) | 10 lines Fixing autofill to be more accurate. + Specifically, if calls ahead of the current caller were ringing + members (but not yet bridged) there could be available members + and waiting callers who would not get matched up. The member + availability checker was correctly determining the number of + available members in this scenario, but the queue itself did not + parallelly reflect this status on the pending calls. This commit + corrects the issue. (closes issue #11459, reported by + equissoftware, patched by me) ........ + +2007-12-11 16:29 +0000 [r92305] Russell Bryant <russell@digium.com> + + * include/asterisk/unaligned.h, main/event.c: * In unaligned.h, + remove some unnecessary casts and mark the arg of the + get_unaligned functions as const * In event.c, use + get_unaligned_uint32() in a couple of places to fix issues on + architectures that don't allow unaligned access + +2007-12-11 14:17 +0000 [r92267-92285] Olle Johansson <oej@edvina.net> + + * include/asterisk/devicestate.h, include/asterisk/agi.h, + include/asterisk/astobj2.h, include/asterisk/extconf.h, + include/asterisk/io.h, include/asterisk/cdr.h, + include/asterisk/aes.h, include/asterisk/_private.h, + include/asterisk/localtime.h, include/asterisk/hashtab.h, + include/asterisk/callerid.h, include/asterisk/logger.h, + include/asterisk/doxyref.h, include/asterisk/app.h, + include/asterisk/adsi.h, include/asterisk/event.h, + include/asterisk/causes.h, include/asterisk/alaw.h, + include/asterisk/ast_expr.h, include/asterisk/dsp.h, + include/asterisk/mod_format.h, include/asterisk/ael_structs.h, + include/asterisk/astdb.h: A lot of doxygen updates + + * include/asterisk/frame.h: Doxygen updates + +2007-12-10 20:18 +0000 [r92243] Doug Bailey <dbailey@digium.com> + + * channels/chan_zap.c: Add CLI commands to dynamically set hw and + sw gains + +2007-12-10 16:48 +0000 [r92205-92206] Joshua Colp <jcolp@digium.com> + + * utils/check_expr.c: Add ast_atomic_fetchadd_int_slow to + check_expr for platforms that need it. (closes issue #11484) + Reported by: snuffy + + * /, main/rtp.c: Merged revisions 92204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6 + lines Add G729A as another possible payload name for G729. Some + devices use this instead of G729, which is perfectly normal since + the payload number itself is defined and can't be used by + anything else so the name doesn't matter that much. (closes issue + #11483) Reported by: revolution Patches: rtp.diff uploaded by + revolution (license 346) ........ + +2007-12-10 16:30 +0000 [r92203] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 92202 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec + 2007) | 7 lines If there are no members in a queue, then the loop + where the datastore for detecting duplicate dialed numbers will + be skipped, meaning the datastore isn't created. This means that + when we try to free it, there's a crash. This stops that crash + from occurring. (closes issue #11499, reported by slavon, patched + by eliel) ........ + +2007-12-10 16:15 +0000 [r92199-92201] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 92200 via svnmerge ........ r92200 | file | + 2007-12-10 12:13:43 -0400 (Mon, 10 Dec 2007) | 4 lines It is + possible for nativeformats to contain more then one codec, so + print out multiple ones. (closes issue #11366) Reported by: ovi + ........ + + * res/res_agi.c: Only send a SIGHUP if the pid is greater than -1, + otherwise all PIDs greater than -1 will get the SIGHUP... and + that is bad. (closes issue #11453) Reported by: alanmcmillan + +2007-12-10 14:18 +0000 [r92140-92160] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Removing some LOG_DEBUG items + + * /, channels/chan_sip.c: Merged revisions 92158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16 + lines Avoid reinvite race situations with two Asterisks trying to + reinvite each other in 1.4 and trunk. This patch implements + support for the 491 error code that Asterisk 1.4 generates on + situations where we get an incoming INVITE and already has one in + progress. Thanks to mavetju for reporting and to Raj Jain for an + excellent explanation of the problem. Patch by myself. Tested + with 8 Asterisk servers connected to each other in a training + network. Closes issue #10481 ........ + + * doc/manager_1_1.txt, apps/app_voicemail.c: Add a few extra + headers in the voicemail users listing in manager 1.1. Update + documentation too. (closes issue #11495) Reported by: caio1982 + Patches: extra_vm_manager_info1.diff uploaded by caio1982 + (license 22) + +2007-12-10 09:00 +0000 [r91929-92122] Luigi Rizzo <rizzo@icir.org> + + * build_tools/make_version, build_tools/make_version_h: + simplify/cleanup the scripts + + * utils/Makefile: remove relative paths and use ASTTOPDIR instead. + Give a default value to ASTTOPDIR if unset so we can at least do + a 'make clean' without too much trouble. The proper fix, however, + is to partition the top level Makefile in a 'setup' and a 'main' + part, in a way that the 'setup' part can be included from + subdirs' Makefiles and allow targets to be built without going + through the top level Makefile. + + * utils/clicompat.c: simplify this file + + * doc/CODING-GUIDELINES: add a bit of info on the build + infrastructure + + * Makefile: Fix the detection of modules installed from this build. + You can now add the path of local module subdirs from the command + line with make LOCAL_MOD_SUBDIRS= .... + + * codecs/Makefile, apps/Makefile, Makefile.moddir_rules, + cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile, + formats/Makefile, funcs/Makefile: Put into Makefile.moddir_rules + the common instructions used to generate loadable and embedded + module lists. Individual Makefiles now are a lot simpler, + possibly as simple as this: -include + $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps + MODULE_PREFIX=cdr_ all: _all include + $(ASTTOPDIR)/Makefile.moddir_rules and also more flexible because + in a single directory we can combine various types of modules + (app_, cdr_, func_, ... ) by simply listing them in the + MODULE_PREFIX variable. The individual Makefiles can also create + list of modules to be excluded by listing them in the variablel + MODULE_EXCLUDE (see an example in channels/Makefile). With this + change it becomes trivial to integrate a directory with locally + created/modified sources into the main build. + + * Makefile, Makefile.moddir_rules: make the install target a bit + less noisy + + * Makefile: document usage of several exported variables + + * utils/Makefile: add hashtab.c to the list of files deleted + + * Makefile.moddir_rules: another place where ../ should have been + ASTTOPDIR + + * codecs/Makefile, utils/Makefile, apps/Makefile, cdr/Makefile, + pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile, + funcs/Makefile: normalize subdirs' Makefile by using ASTTOPDIR + and not .. to reference the top level directory. + + * Makefile: Implement the outcome of a discussion on the -dev list + re. the use of DESTDIR and INSTALL_PATH - many thanks to Tzafrir + Cohen and Simon Perreault for extremely useful feedback: 1. + comment out the [directories] section the created asterisk.conf ; + you can set the correct defaults at build time using + INSTALL_PATH, so the repetition here is redundant and often + wrong. (The next step now is move asterisk.conf outside the + Makefile to asterisk.conf.sample, because there is little if + anything here that needs to be constructed at build/install + time). 2. use DESTDIR?=$(INSTALL_PATH) so you only need to + specify a path once if the two coincide. This should have no ill + side effects, because if you don't specify DESTDIR, you really + need INSTALL_PATH="" to set the correct defaults, and if you + specify DESTDIR the value is not overridden. The second part + required moving the 'export DESTDIR' right after the assignment + to prevent DESTDIR getting set by the export (this is documented + in the Makefile).o hopefully avoid the mistake)$ With this change + you can now do something like this from your source tree: make + INSTALL_PATH=/some/place install samples and then main/asterisk + -vdc which will pick up the correct config files and libraries + from /some/place - i.e. great for developers! + + * main/config.c: remove unused code, and simplify the logic for + #include/#exec (still a lot of cleanup needed here). + + * main/config.c: Implement comment_buffer and lline_buffer in terms + of the ast_str_*() API. I don't know if comment_buffers etc are + actually used at all... + + * main/config.c: unify some common code + + * main/config.c: normalize formatting + + * main/config.c: document a nice technique to exit from a block in + case of errors. + + * main/config.c: a little bit of documentation on how lines are + parsed. + + * utils/ael_main.c: normalize header order, and add a comment on + the need to clean up this file. + + * include/asterisk/network.h: some platforms (e.g. FreeBSD4) need + netinet/in.h to be included before arpa/inet.h + +2007-12-07 23:32 +0000 [r91832-91891] Jason Parker <jparker@digium.com> + + * /, main/dsp.c: Merged revisions 91890 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11273) ........ r91890 | qwell | 2007-12-07 17:29:01 -0600 + (Fri, 07 Dec 2007) | 4 lines We need to make sure we free the + input frame if we return a different frame in ast_dsp_process. + Issue 11273, pointed out by dimas, with a patch by eliel. + ........ + + * pbx/pbx_lua.c, configs/extensions.lua.sample: Update + documentation for pbx_lua. Closes issue #11492, patch by + mnicholson. + +2007-12-07 21:25 +0000 [r91784-91831] Russell Bryant <russell@digium.com> + + * /, main/utils.c: Merged revisions 91830 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91830 | russell | 2007-12-07 15:24:33 -0600 (Fri, 07 Dec 2007) | + 5 lines Make the lock protecting each thread's list of locks it + currently holds recursive. I think that this will fix the + situation where some people have said that "core show locks" + locks up the CLI. (related to issue #11080) ........ + + * /, include/asterisk/lock.h: Merged revisions 91828 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r91828 | russell | 2007-12-07 15:17:24 -0600 (Fri, 07 + Dec 2007) | 6 lines Fix another bug in the DEBUG_THREADS code. + The ast_mutex_init() function had the mutex attribute object + marked as static. This means that multiple threads initializing + locks at the same time could step on each other and end up with + improperly initialized locks. (found when tracking down locking + issues related to issue #11080) ........ + + * /, include/asterisk/lock.h: Merged revisions 91826 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r91826 | russell | 2007-12-07 15:11:08 -0600 (Fri, 07 + Dec 2007) | 6 lines I love fixing lock related errors in the lock + debugging code. That's about as ironic as it gets in Asterisk + programming land. Anyway, I spotted this bug while trying to + track down why systems are locking up and acting weird in issue + #11080. The mutex attribute object was marked as static in this + function when it should not have been. ........ + + * apps/app_dial.c, /: Merged revisions 91783 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) | + 6 lines * Add channel locking around datastore operations that + expect the channel to be locked. * Document why we don't record + Local channels in the dialed interfaces list. * Remove the dialed + variable as it isn't needed. * Restructure some code for clarity + and coding guidelines stuff ........ + +2007-12-07 16:37 +0000 [r91782] Jason Parker <jparker@digium.com> + + * channels/chan_sip.c: Fix a small typo in a comment. Closes issue + #11490 + +2007-12-07 16:28 +0000 [r91781] Russell Bryant <russell@digium.com> + + * /, apps/app_queue.c: Merged revisions 91780 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) | + 7 lines * Add channel locking around datastore operations that + expect the channel to be locked. * Document why we don't record + Local channels in the dialed interfaces list. * Handle memory + allocation failure. * Remove the dialed variable, as it wasn't + actually needed. * Tweak some formatting to conform to coding + guidelines. ........ + +2007-12-07 16:11 +0000 [r91779] Jason Parker <jparker@digium.com> + + * doc/asterisk-mib.txt, main/pbx.c, res/snmp/agent.c, + include/asterisk/pbx.h, main/cli.c: Add count of total number of + calls processed by asterisk during it's lifetime. Add number of + total calls and current calls to SNMP. Closes issue #10057, patch + by jcmoore. + +2007-12-07 16:11 +0000 [r91778] Russell Bryant <russell@digium.com> + + * main/autoservice.c, /: Merged revisions 91777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91777 | russell | 2007-12-07 10:08:35 -0600 (Fri, 07 Dec 2007) | + 6 lines * Add a bit more of a verbose comment as to why a hangup + frame needs to be queued up if autoservice gets a NULL return + from ast_read(). * Make the process of queueing the hangup frame + more efficient by putting the frame where it is going to end up + and avoiding some locking and extra memory allocations and + freeing. ........ + +2007-12-07 15:40 +0000 [r91738] Mark Michelson <mmichelson@digium.com> + + * main/autoservice.c, /: Merged revisions 91737 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91737 | mmichelson | 2007-12-07 09:39:58 -0600 (Fri, 07 Dec + 2007) | 7 lines Hangups that happen during autoservice were not + processed appropriately. This is because a hangup actually causes + a NULL frame to be received, not a hangup frame. Queueing a + hangup if we receive a NULL frame during autoservice corrects + this problem (closes issue #11467, reported by jmls, patched by + me) ........ + +2007-12-07 02:52 +0000 [r91676-91700] Russell Bryant <russell@digium.com> + + * apps/app_dial.c, /: Merged revisions 91693 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) | + 2 lines Don't unlock the dialed_interfaces list until we're done + messing with the iterator. ........ + + * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 91677 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) | + 4 lines Allow dialing local channels from Queue() and Dial() + again. There was a slight flaw in the code to prevent call + forwards from looping that caused this problem. (related to issue + #11486) ........ + + * /, apps/app_queue.c: Merged revisions 91675 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) | + 7 lines Fix in an issue in the call forwarding handling code that + was causing crashes on every call into a queue. I'm not entirely + sure about the logic in this part of the code, so I want to look + at it some more tomorrow. However, this makes it safe and keeps + it from crashing. (closes issue #11486, reported by adamg, + patched by me) ........ + +2007-12-07 00:58 +0000 [r91617-91638] Tilghman Lesher <tlesher@digium.com> + + * /, main/rtp.c: Merged revisions 91637 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91637 | tilghman | 2007-12-06 18:52:17 -0600 (Thu, 06 Dec 2007) + | 5 lines At the end of a call, when we're reporting, RTCP may + already be partially torn down, so check for NULL dereference + Reported by: blitzrage Patch by: tilghman (Closes issue #11450) + ........ + + * channels/chan_zap.c: Add a manager event for PRI events: this + will help manager users detect when a D-channel goes down + + * main/cdr.c: If duration or billsec are not yet calculated, + calculate them on demand. + +2007-12-06 21:57 +0000 [r91598] Jason Parker <jparker@digium.com> + + * cdr/cdr_sqlite3_custom.c: Fix a problem with quoting in sqlite3 + cdr module.. Closes issue #11070, patch by seanbright. + +2007-12-06 21:03 +0000 [r91579] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Handle allocation failure of the heard and + deleted arrays of the vm_state. (closes issue #11408, reported + and patched by jaroth) + +2007-12-06 20:52 +0000 [r91561] Tilghman Lesher <tlesher@digium.com> + + * /, cdr/cdr_pgsql.c: Merged revisions 90166,90736,90753 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90166 | tilghman | 2007-11-29 13:48:10 -0600 (Thu, 29 Nov 2007) + | 3 lines Properly escape cdr->src and cdr->dst and ensure we use + thread-safe escaping (Fixes AST-2007-026) ........ r90736 | + tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines + If both dbhost and dbsock were not set, a NULL deref could result + Reported by: xrg Patch by: tilghman (Closes issue #11387) + ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03 + Dec 2007) | 5 lines Solaris requires the inclusion of + sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: + snuffy,tilghman (Closes issue #11430) ........ + +2007-12-06 19:12 +0000 [r91502] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 91501 via svnmerge ........ r91501 | russell + | 2007-12-06 13:11:35 -0600 (Thu, 06 Dec 2007) | 5 lines Add a + new module flag to indicate that a build sum is present. Modules + built against older Asterisk 1.4 headers will now load properly + with just a warning indicating that they are old and may cause + problems. (patch by paravoid) ........ + +2007-12-06 16:54 +0000 [r91472] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Make sure we clear these flags when libpri + is not installed + +2007-12-06 16:51 +0000 [r91440-91458] Joshua Colp <jcolp@digium.com> + + * main/udptl.c, /: Merged revisions 91450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91450 | file | 2007-12-06 12:49:42 -0400 (Thu, 06 Dec 2007) | 6 + lines Fix various in the udptl implementation. It could return + empty modem frames, have an incorrect sequence number on packets, + and display the wrong sequence number in the debug messages. + (closes issue #11228) Reported by: Cache Patches: udptl-4.patch + uploaded by dimas (license 88) ........ + + * /, channels/chan_sip.c: Merged revisions 91439 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4 + lines Add support for accepting and sending T.38 in the initial + INVITE. (closes issue #9402) Reported by: thdei ........ + +2007-12-06 15:56 +0000 [r91347-91438] Olle Johansson <oej@edvina.net> + + * doc/manager_1_1.txt (added), UPGRADE.txt: Adding documentation + for the massive manager changes to manager version 1.1 - + hopefully a more consistent manager interface. + + * main/manager.c: - The Ping Action - Now use Response: success - + New header "Ping: pong" :-) - The Events action - Now use + Response: Success - The new status is reported as "Events: On" or + "Events: Off" - Report if manager is enabled in the reload event + Small cleanups... From moremanager + + * main/channel.c: Changes to manager events in channel.c - Newstate + event - Now has "CalleridNum" for numeric caller id, like + Newchannel - The event does not send "<unknown>" for unknown + caller IDs just an empty field - Newstate and Newchannel events - + these have changed headers "State" -> ChannelStateDesc Text based + channel state -> ChannelState Numeric channel state - The events + does not send "<unknown>" for unknown caller IDs just an empty + field - Newstate event - Now has "CalleridNum" for numeric caller + id, like Newchannel - The event does not send "<unknown>" for + unknown caller IDs just an empty field - Link and Unlink events - + The "Link" and "Unlink" bridge events in channel.c are now + renamed to "Bridge" - The link state is in the bridgestate: + header as "Link" or "Unlink" - For channel.c bridges, + "Bridgetype: core" is added. This opens up for bridge events in + rtp.c and channel drivers - The "Rename" manager event has a + renamed header, to use the same terminology for the current + channel as other events - Oldname -> Channel (Moremanager) + + * main/cdr.c: New manager event when a channel changes account + code. Maybe belongs in the new cdr category? ---moremanager--- + Event: NewAccountCode Modules: cdr.c Purpose: To report a change + in account code for a live channel Example: Event: NewAccountCode + Privilege: call,all Channel: SIP/olle-01844600 Uniqueid: + 1177530895.2 AccountCode: Stinas account 1234848484 + OldAccountCode: Olles Account 12345 + + * apps/app_dial.c: - Dial event - Event Dial has new headers, to + comply with other events - Source -> Channel Channel name + (caller) - SrcUniqueID -> UniqueID Uniqueid (new) -> Dialstring + Dialstring in app data (moremanager) + + * apps/app_meetme.c: Adding small missing but important comma... + + * apps/app_meetme.c: A big oops... + + * apps/app_meetme.c: The MeetmeJoin now has caller ID name and + Caller ID number fields (like MeetMeLeave) (Moremanager) + + * channels/chan_zap.c: Update ZapShowChannels so that you can + specify one channel. Action ZapShowChannels Header changes - + Channel: -> ZapChannel For active channels, the Channel: and + Uniqueid: headers are added You can now add a "ZapChannel: " + argument to zapshowchannels actions to only get information about + one channel. From the moremanager branch + + * main/logger.c: Doxygen updates + + * include/asterisk/logger.h, /, main/logger.c, main/loader.c: + Merged revisions 91366 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91366 | oej | 2007-12-06 13:54:11 +0100 (Tor, 06 Dec 2007) | 4 + lines Make sure logger is reloaded at general reload in the cli. + (Discovered during Asterisk training in Portugal) ........ + + * main/manager.c: Change description of new manager command + + * main/manager.c, CHANGES: Add manager command for showing all + current channels. Thanks, eliel, for writing the original patch. + Modified by me to follow other manager events and the new + "moremanager" style. (closes issue #11478) Reported by: eliel + Patches: manager.c.patch uploaded by eliel (license 64) + +2007-12-06 04:37 +0000 [r91328] Joshua Colp <jcolp@digium.com> + + * main/channel.c: Instead of iterating through the entire epoll + events array just look at the ones that will actually contain + data. (props to eliel on IRC for this) + +2007-12-05 22:57 +0000 [r91291-91293] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 91292 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91292 | mmichelson | 2007-12-05 16:57:13 -0600 (Wed, 05 Dec + 2007) | 3 lines Reverting extra stuff I didn't mean to commit + ........ + + * apps/app_dial.c, /, apps/app_voicemail.c: Merged revisions 91273 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec + 2007) | 11 lines The 'G' option for Dial() did not properly + handle the case where only a label was provided. This was due to + the fact that the answering channel did not have an extension + set, so ast_parseable_goto would fail. This fix eliminates the + call to ast_parseable_goto on the answering channel since it is a + wasteful call. The answering channel and the calling channel are + both directed to the same extension and context, just different + priorities, so we can just copy the values from the calling + channel to the answering channel and increment the answering + channel's priority. (closes issue #11382, reported by jon, patch + by me with correction by jon) ........ + +2007-12-05 21:46 +0000 [r91238] Tilghman Lesher <tlesher@digium.com> + + * /, sounds/Makefile: Merged revisions 91237 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91237 | tilghman | 2007-12-05 15:38:13 -0600 (Wed, 05 Dec 2007) + | 2 lines Upgrade to the latest version of extra sounds ........ + +2007-12-05 17:49 +0000 [r91193-91197] Russell Bryant <russell@digium.com> + + * /, main/threadstorage.c: Merged revisions 91192 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91192 | russell | 2007-12-05 11:31:42 -0600 (Wed, 05 Dec 2007) | + 10 lines Make the lock in the threadstorage debugging code + untracked to avoid a deadlock on thread destruction. (closes + issue #11207) Reported by: ys Patches: threadstorage.c.diff + uploaded by ys (license 281) Also fixes an open bug report: + (closes issue #11446) ........ + + * apps/app_directory.c: Resolve compiler warnings. + +2007-12-05 16:46 +0000 [r91172-91173] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, UPGRADE.txt, configs/manager.conf.sample, + CHANGES, include/asterisk/manager.h, cdr/cdr_manager.c: Change + cdr_manager to use a "CDR" level, rather than the (overcrowded) + "call" level. (Closes issue #11015) + + * CHANGES, apps/app_directory.c: Added multiple name listing. + (Closes issue #10413) + +2007-12-05 16:14 +0000 [r91171] Joshua Colp <jcolp@digium.com> + + * configs/http.conf.sample: Remove second prefix line. Only need it + documented once in the same file. (closes issue #11472) Reported + by: eserra Patches: http.conf.sample.diff uploaded by eserra + (license 45) + +2007-12-05 13:09 +0000 [r91151-91152] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample: Rename + "username" to "defaultuser" to match with "defaultip". "Username" + still works, but is deprecated. + + * channels/chan_sip.c: Remove the cseqs from "sip show channel" and + make more place for the call ID. + +2007-12-05 03:48 +0000 [r91133] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c: revert part of my changes from earlier today + since this code is no longer dependent on libpri.h + +2007-12-05 03:34 +0000 [r91029-91131] Russell Bryant <russell@digium.com> + + * res/res_odbc.c: Use ast_free() instead of free(). (closes issue + #11309) Reported by: Laureano Patches: res_odbc.c.patch uploaded + by Laureano (license 265) + + * /, include/asterisk/lock.h: Merged revisions 91070 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r91070 | russell | 2007-12-04 18:35:31 -0600 (Tue, 04 + Dec 2007) | 11 lines Fix some crashes in chan_iax2 that were + reported as happening on Mac systems. It turns out that the + problem was the Mac version of the ast_atomic_fetchadd_int() + function. The Mac atomic add function returns the _new_ value, + while this function is supposed to return the old value. So, the + crashes happened on unreferencing objects. If the reference count + was decreased to 1, ao2_ref() thought that it had been decreased + to zero, and called the destructor. However, there was still an + outstanding reference around. (closes issue #11176) (closes issue + #11289) ........ + + * /, main/utils.c: Merged revisions 91074 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r91074 | russell | 2007-12-04 18:48:47 -0600 (Tue, 04 Dec 2007) | + 4 lines When DEBUG_THREADS is enabled, we only have the details + about who is holding a lock that we are waiting on for a mutex, + not rwlocks. This should fix the problem where people have + reported "core show locks" crashing sometimes. ........ + + * channels/chan_zap.c: Fix mwimonitornotify on reload ... again. + This option was only read at startup so a reload would erase it + and not reset it. (pointed out by tzafrir) + + * /: Blocked revisions 91032 via svnmerge ........ r91032 | russell + | 2007-12-04 17:46:40 -0600 (Tue, 04 Dec 2007) | 5 lines Modify + file.h to maintain API compatibility with earlier versions. If a + recent compiler is being used, then a warning will show up for + any modules still using the old name "private" instead of + "_private". (patch suggested by paravoid) ........ + + * utils/astman.c: Fix the build of astman. Any file that includes + any asterisk sub-headers needs to first include asterisk.h. + (closes issue #11394) + +2007-12-04 22:44 +0000 [r91012] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Don't error when we don't have libpri + installed with libss7 support. Also, print the debug message + anyway if we can't find the right PRI + +2007-12-04 22:07 +0000 [r91010-91011] Russell Bryant <russell@digium.com> + + * main/pbx.c, /: Merged revisions 90967 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) | + 7 lines Make some changes to some additions I made recently for + doing channel autoservice when looking up extensions. This code + was added to handle the case where a dialplan switch was in use + that could block for a long time. However, the way that I added + it, it did this for all extension lookups. However, lookups in + the in-memory tree of extensions should _not_ take long enough to + matter. So, move the autoservice stuff to be only around + executing a switch. ........ + + * channels/chan_zap.c: Fix resetting mwimonitornotify on reload. I + guess I only added this line in my head. (thanks to tzafrir for + pointing it out) + +2007-12-04 21:46 +0000 [r90993] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_usbradio.c: Coding guidelines fixups (Closes issue + #11412) + +2007-12-04 21:23 +0000 [r90991] Jason Parker <jparker@digium.com> + + * channels/chan_sip.c, CHANGES: Add manager action + 'sipshowregistry'. Closes issue #11464, patch by eliel. + +2007-12-04 19:08 +0000 [r90949] Russell Bryant <russell@digium.com> + + * include/asterisk/callerid.h, channels/chan_zap.c, + main/callerid.c, CHANGES, configs/zapata.conf.sample: Add support + for monitoring MWI on FXO lines. This introduces two new options + for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor + option enables MWI monitoring. When the MWI state on a line + changes, then the script specified by mwimonitornotify will be + executed for custom handling of the state change, similar to the + externnotify option of voicemail.conf. Also, when the MWI state + on an FXO line changes, an internal Asterisk event is generated + to indicate the new state of the associated mailbox. That may, + any module that cares about MWI information will get notified and + can handle it just as if app_voicemail had sent this + notification. (BE-253, original patch from markster, with some + minor modifications by me to add comments, documentation, and + internal event support) + +2007-12-04 18:29 +0000 [r90930] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Kevin suggested doing the reverse of my + last commit, since imap_retrieve_file does not modify the + contents of the "mailbox" string. In other words, I'm changing + the imap_retrieve_file function to take a const char* as the + third argument so that I don't need to cast const char*'s as + char*'s to suppress compiler warnings. + +2007-12-04 18:15 +0000 [r90929] Jason Parker <jparker@digium.com> + + * Makefile: Add Makefile alias target 'pdf' which does the same + thing as asterisk.pdf. Issue 11452, reported by blitzrage. + +2007-12-04 18:14 +0000 [r90928] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Suppress a compiler warning due to + discarding a "const" qualifier + +2007-12-04 18:09 +0000 [r90927] Jason Parker <jparker@digium.com> + + * main/global_datastores.c: Fix build, that some people aren't + seeing for some reason. + +2007-12-04 17:51 +0000 [r90899] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Wrong locking style got merged from 1.4 to + trunk. My mistake. + +2007-12-04 17:40 +0000 [r90880] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c: fix build of this module when libpri and/or + libss7 are or are not present + +2007-12-04 17:38 +0000 [r90879] Jason Parker <jparker@digium.com> + + * main/channel.c, /: Merged revisions 90876 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11454) ........ r90876 | qwell | 2007-12-04 11:28:08 -0600 + (Tue, 04 Dec 2007) | 4 lines If we fail to create a channel after + allocating a timing fd, we need to make sure to close it. Issue + 11454, patch by eliel. ........ + +2007-12-04 17:36 +0000 [r90878] Russell Bryant <russell@digium.com> + + * main/Makefile: Fix a silly little typo :) + +2007-12-04 17:35 +0000 [r90877] Jason Parker <jparker@digium.com> + + * apps/app_dial.c: Fix build in trunk. This was fixed in 1.4, but + blocked in trunk since this hadn't been merged yet. + +2007-12-04 17:08 +0000 [r90873] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, main/global_datastores.c (added), + channels/chan_local.c, /, main/Makefile, + include/asterisk/channel.h, include/asterisk/global_datastores.h + (added), apps/app_queue.c: Merged revisions 90735 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 + Dec 2007) | 22 lines A big one... This is the merge of the + forward-loop branch. The main change here is that call-forwards + can no longer loop. This is accomplished by creating a datastore + on the calling channel which has a linked list of all devices + dialed. If a forward happens, then the local channel which is + created inherits the datastore. If, through this progression of + forwards and datastore inheritance, a device is attempted to be + dialed a second time, it will simply be skipped and a warning + message will be printed to the CLI. After the dialing has been + completed, the datastore is detached from the channel and + destroyed. This change also introduces some side effects to the + code which I shall enumerate here: 1. Datastore inheritance has + been backported from trunk into 1.4 2. A large chunk of code has + been removed from app_dial. This chunk is the section of code + which handles the call forward case after the channel has been + requested but before it has been called. This was removed because + call-forwarding still works fine without it, it makes the code + less error-prone should it need changing, and it made this set of + changes much less painful to just have the forwarding handled in + one place in each module. 3. Two new files, global_datastores.h + and .c have been added. These are necessary since the datastore + which is attached to the channel may be created and attached in + either app_dial or app_queue, so they need a common place to find + the datastore info. This approach was taken in case similar + datastores are needed in the future, there will be a common place + to add them. ........ + +2007-12-04 15:16 +0000 [r90852-90854] Olle Johansson <oej@edvina.net> + + * apps/app_queue.c: (closes issue #11431) Reported by: Laureano + Patches: app_queue.c.patch uploaded by Laureano (license 265) + + * main/pbx.c, CHANGES: (closes issue #11422) Reported by: eliel + Patches: core.show.hint.patch uploaded by eliel (license 64) + + * CHANGES: (closes issue #11462) Reported by: eliel Patches: + CHANGES.patch uploaded by eliel (license 64) + +2007-12-04 15:01 +0000 [r90851] Tilghman Lesher <tlesher@digium.com> + + * res/res_agi.c: Pass the Asterisk version to AGI scripts as part + of the initial dump of info Reported by: acunningham Patch by: + acunningham (Closes issue #11398) + +2007-12-04 11:50 +0000 [r90834] Luigi Rizzo <rizzo@icir.org> + + * res/Makefile: fix build on cygwin + +2007-12-04 05:31 +0000 [r90799] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 90798 via svnmerge ........ r90798 | file | + 2007-12-04 01:29:33 -0400 (Tue, 04 Dec 2007) | 2 lines Fix build + issue on the build cluster. ........ + +2007-12-03 23:52 +0000 [r90760] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/compat.h: Merged revisions 90753 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03 + Dec 2007) | 5 lines Solaris requires the inclusion of + sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: + snuffy,tilghman (Closes issue #11430) ........ + +2007-12-03 23:49 +0000 [r90746] Steve Murphy <murf@digium.com> + + * main/hashtab.c: A small fix from snuffy + +2007-12-03 23:48 +0000 [r90738] Jason Parker <jparker@digium.com> + + * res/res_monitor.c: Add manager events for when a monitor is + started or stopped. Closes issue #10191, patch by dgradecak. + +2007-12-03 23:29 +0000 [r90737] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_pgsql.c, /: Merged revisions 90736 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 + Dec 2007) | 5 lines If both dbhost and dbsock were not set, a + NULL deref could result Reported by: xrg Patch by: tilghman + (Closes issue #11387) ........ + +2007-12-03 22:07 +0000 [r90697] Jason Parker <jparker@digium.com> + + * /, apps/app_meetme.c: Merged revisions 90696 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes + issue #11383) ........ r90696 | qwell | 2007-12-03 16:06:36 -0600 + (Mon, 03 Dec 2007) | 4 lines Make sure we always close the + conference fd if we have an open one. Issue 11383, reported by + markmhy, patch by eliel. ........ + +2007-12-03 21:24 +0000 [r90670] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Replacing some calls to free() with + ast_free(). (closes issue #11448, reported and patched by jaroth) + +2007-12-03 21:03 +0000 [r90656] Joshua Colp <jcolp@digium.com> + + * include/asterisk/agi.h, res/res_agi.c, CHANGES: Add AGI commands + for speech recognition. These mirror the dialplan applications + mostly but present the information in a nicer fashion. The SPEECH + RECOGNIZE command for example will return the results instead of + having to query the dialplan functions. + +2007-12-03 21:00 +0000 [r90644] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_mgcp.c: Merged revisions 90639 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec + 2007) | 5 lines Changing some bad logic when calculating the + interdigit timeout. (closes issue #11402, reported and patched by + eferro) ........ + +2007-12-03 20:58 +0000 [r90631] Jason Parker <jparker@digium.com> + + * /, res/res_features.c: Merged revisions 90607 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes + issue #11436) ........ r90607 | qwell | 2007-12-03 14:51:17 -0600 + (Mon, 03 Dec 2007) | 4 lines Fix crash in ParkAndAnnounce + application. Issue #11436, reported by lytledd, patch by eliel. + ........ + +2007-12-03 20:30 +0000 [r90591] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c: Avoid an additional function call. Reported by: + eliel Patch by: eliel (Closes issue #11438) + +2007-12-03 20:07 +0000 [r90550-90589] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 90588 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2 + lines Do not create a smoother for G723.1 frames, they need to be + left alone to their native 20/24 byte size. ........ + + * main/channel.c, /, include/asterisk/channel.h, .cleancount: + Merged revisions 90548 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2 + lines Preserve the indication currently playing on a channel when + a masquerade operation happens. (issue #BE-88) ........ + +2007-12-03 18:21 +0000 [r90547] Jason Parker <jparker@digium.com> + + * /: Blocked revisions 90546 via svnmerge ........ r90546 | qwell | + 2007-12-03 12:20:49 -0600 (Mon, 03 Dec 2007) | 4 lines Only log + debug messages if debug is enabled. Closes issue #11416, patch by + casper. ........ + +2007-12-03 16:46 +0000 [r90528] Mark Michelson <mmichelson@digium.com> + + * configs/queues.conf.sample: Updating sample queues.conf file to + show how multiple periodic announcements may be specified since + this was not documented previously (closes issue #11432, reported + and patched by Laureano) + +2007-12-03 14:14 +0000 [r90508] Joshua Colp <jcolp@digium.com> + + * apps/app_dial.c: Remove the file descriptors from the main poll + channel when the channel is hung up during the dialing attempt, + and make sure a channel exists before trying to remove it at the + end. (closes issue #11441) Reported by: blitzrage + +2007-12-02 18:20 +0000 [r90471] Russell Bryant <russell@digium.com> + + * /, apps/app_queue.c: Merged revisions 90470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) | + 6 lines The other day when I went through making changes as a + result of the ao2_link() change, I added some code to set + pointers to NULL after they were unreferenced. This pointed out + that in this place, the object was unreferenced before the code + was done using it. So, move the unref down a little bit. (crash + reported by jmls on IRC) ........ + +2007-12-02 09:42 +0000 [r90433] Tilghman Lesher <tlesher@digium.com> + + * main/autoservice.c, /: Merged revisions 90432 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007) + | 7 lines Clarify the return value on autoservice. Specifically, + if you started autoservice and autoservice was already on, it + would erroneously return an error. Reported by: adiemus Patch by: + dimas (Closes issue #11433) ........ + +2007-12-01 01:37 +0000 [r90410] Jason Parker <jparker@digium.com> + + * res/res_adsi.c: Only reload if the config file has changed. + Closes issue #11281, patch by eliel. + +2007-11-30 21:19 +0000 [r90388] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, include/asterisk/app.h, + include/asterisk/audiohook.h, res/res_features.c, + include/asterisk/channel.h, main/audiohook.c, apps/app_queue.c, + configs/features.conf.sample: Adding support for the + "automixmonitor" dial and queue options. This works in much the + same way as the automonitor, except that instead of using the + monitor app, it uses the mixmonitor app. By providing an 'x' or + 'X' as a dial or queue option, a DTMF sequence may be entered (as + defined in features.conf) to start the one-touch mixmonitor. This + patch also introduces some new API calls to the audiohooks code + for searching for an audiohook by type and for searching for a + running audiohook by type. Big thanks to joetester for writing + the initial patch, testing it and patiently waiting for it to be + committed. (closes issue #10185, reported and patched by + xmarksthespot) + +2007-11-30 19:34 +0000 [r90311-90351] Russell Bryant <russell@digium.com> + + * main/manager.c, /, include/asterisk/astobj2.h, apps/app_queue.c, + channels/chan_iax2.c, main/astobj2.c, main/config.c: Merged + revisions 90348 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | + 8 lines Change the behavior of ao2_link(). Previously, in + inherited a reference. Now, it automatically increases the + reference count to reflect the reference that is now held by the + container. This was done to be more consistent with ao2_unlink(), + which automatically releases the reference held by the container. + It also makes it so it is no longer possible for a pointer to be + invalid after ao2_link() returns. ........ + + * /, include/asterisk/astobj2.h: Merged revisions 90310 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90310 | russell | 2007-11-30 12:46:46 -0600 (Fri, 30 Nov 2007) | + 2 lines Add some notes on the behavior of ao2_unlink() after a + discussion with Tilghman ........ + +2007-11-30 14:45 +0000 [r90270] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 90269 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6 + lines Fix locking issues under one legged replaces scenarios. + (closes issue #11420) Reported by: irroot Patches: + chan_sip_oneleg.patch uploaded by irroot (license 52) ........ + +2007-11-30 00:16 +0000 [r90164-90232] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_mgcp.c: Merged revisions 90231 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov + 2007) | 5 lines Clear the DTMF buffer if the call times out. + (closes issue #11418, reported and patched by eferro) ........ + + * /, apps/app_queue.c: Merged revisions 90163 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov + 2007) | 6 lines This patch handles the case where a queue member + with a negative penalty is added via the manager. If a negative + value is submitted for a member penalty, we set it to 0. (closes + issue #11411, reported and patched by Laureano) ........ + +2007-11-29 19:35 +0000 [r90156-90162] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_pgsql.c, /: Merged revisions 90160 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r90160 | tilghman | 2007-11-29 13:24:11 -0600 (Thu, 29 + Nov 2007) | 2 lines Properly escape input buffers (Fixes + AST-2007-025) ........ + + * /, formats/format_wav.c, formats/format_pcm.c, + formats/format_ogg_vorbis.c, main/file.c, + include/asterisk/mod_format.h, formats/format_h263.c, + formats/format_h264.c, formats/format_wav_gsm.c, + formats/format_g726.c: Merged revisions 90155 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007) + | 5 lines Use of "private" as a field name in a header file + messes with C++ projects Reported by: chewbacca Patch by: casper + (Closes issue #11401) ........ + + * include/asterisk/lock.h: Fix build of trunk + + * /, sounds/Makefile: Merged revisions 90154 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90154 | tilghman | 2007-11-29 11:18:09 -0600 (Thu, 29 Nov 2007) + | 2 lines Upgrade the core sounds release version ........ + +2007-11-29 13:38 +0000 [r90149-90150] Kevin P. Fleming <kpfleming@digium.com> + + * utils/Makefile, utils/hashtest.c: let's try this again... *all* + compilation and linking in Asterisk should be done using the + standard compilation rules, not manually created ones. changing + hashtest.c to use these rules caused the compiler to notice a + large number of coding guidelines violations, so those are fixed + too. + + * main/manager.c: restore behavior from the 1.4 branch... manager + users created via users.conf should default to *all* permissions, + not none + +2007-11-29 00:37 +0000 [r90139-90148] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 90147 via svnmerge ........ r90147 | russell + | 2007-11-28 18:36:59 -0600 (Wed, 28 Nov 2007) | 1 line fix some + formatting i accidentally changed ........ + + * main/channel.c, /, include/asterisk/channel.h, + funcs/func_callerid.c: Merged revisions 90145 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | + 5 lines This set of changes is to make some callerID handling + thread-safe. The ast_set_callerid() function needed to lock the + channel. Also, the handlers for the CALLERID() dialplan function + needed to lock the channel when reading or writing callerid + values directly on the channel structure. ........ + + * include/asterisk/file.h, /, main/file.c: Merged revisions 90142 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) | + 4 lines Merge a change from team/russell/chan_refcount ... This + makes ast_stopstream() thread-safe. ........ + + * include/asterisk/audiohook.h: Merge another small doxygen change + from team/russell/chan_refcount to indicate that a channel + doesn't need to be locked before calling a certain function. + + * include/asterisk/channel.h: Merge some channel.h doxygen updates + from team/russell/chan_refcount This was mostly to note whether a + channel needed to be locked or not before calling these + functions. However, I added some other things, too. + +2007-11-28 23:03 +0000 [r90102] Joshua Colp <jcolp@digium.com> + + * /, res/res_musiconhold.c, apps/app_queue.c: Merged revisions + 90101 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6 + lines Fix a few memory leaks. (closes issue #11405) Reported by: + eliel Patches: load_realtime.patch uploaded by eliel (license 64) + ........ + +2007-11-28 22:44 +0000 [r90100] Kevin P. Fleming <kpfleming@digium.com> + + * configs/users.conf.sample, main/manager.c, /: Merged revisions + 90098 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007) + | 2 lines it is impossible to set permissions for manager + accounts created by users.conf (reported internally, patched by + me) ........ + +2007-11-28 22:32 +0000 [r90099] Joshua Colp <jcolp@digium.com> + + * main/cli.c: file says... compile before you commit! + +2007-11-28 22:17 +0000 [r90060-90061] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c: Removing a pointless check of option_debug + + * main/pbx.c, /: Merged revisions 90059 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov + 2007) | 13 lines Removing some seemingly pointless code. This + sets a channel variable for every priority executed in the + dialplan if you have debug set to anything non-zero. This seems + pointless due to the fact that these channel variables are not + referenced anywhere else in the code and their names are esoteric + enough that they would not be practical to reference in the + dialplan. Plus the fact that this behavior isn't documented + anywhere means that the change is not likely to cause any + disruption. If anything, this may actually cause a slight + performance increase if running with debug on. The motivating + influence for this code change is the eventwhencalled option for + queues. If set to vars, all channel variables will be output to + the manager. These unnecessary channel variables make the output + a lot more difficult to deal with. ........ + +2007-11-28 20:33 +0000 [r90039] Steve Murphy <murf@digium.com> + + * main/ast_expr2f.c, main/ast_expr2.fl: Made expr2 parser 8-bit + transparent + +2007-11-28 20:27 +0000 [r90038] Jason Parker <jparker@digium.com> + + * main/pbx.c, res/res_crypto.c, include/asterisk/cli.h, main/cli.c: + Remove "old"-style CLI handler, since nothing uses it anymore. + Closes issue #11403, patch by eliel. This also completes the + janitor project. + +2007-11-28 17:37 +0000 [r90000] Mark Michelson <mmichelson@digium.com> + + * /: Blocked revisions 89999 via svnmerge ........ r89999 | + mmichelson | 2007-11-28 11:30:47 -0600 (Wed, 28 Nov 2007) | 6 + lines Recording greetings when using IMAP storage was causing + zero-length files to be stored. Since greetings are not retrieved + from IMAP anyway, it is pointless to attempt storing them there. + (closes issue #11359, reported by spditner, patched by me) + ........ + +2007-11-28 15:48 +0000 [r89981-89982] Joshua Colp <jcolp@digium.com> + + * main/cli.c: Hide CLI commands starting with _ from tab completion + as was done previously. (closes issue #11395) Reported by: eliel + Patches: cli.c.patch uploaded by eliel (license 64) + + * main/abstract_jb.c, res/res_agi.c: Fix a few log messages. + (closes issue #11396) Reported by: IgorG Patches: spell.v1.diff + uploaded by IgorG (license 20) + +2007-11-28 00:49 +0000 [r89947] Russell Bryant <russell@digium.com> + + * apps/app_voicemail.c: Merge some little changes from + team/russell/chan_refcount to help reduce the diff to trunk. This + just removes some checks on the return value of alloca(), as + behavior is undefined if it runs out of stack space, and we don't + check it anywhere else. + +2007-11-28 00:47 +0000 [r89946] Mark Michelson <mmichelson@digium.com> + + * configs/musiconhold.conf.sample, configs/extconfig.conf.sample, + res/res_musiconhold.c, CHANGES: Adding support for realtime music + on hold. The following are the main points: 1. When moh is + started, we search first in memory to find the class. If we do + not find it in memory, we search realtime instead. 2. When moh is + restarted (as in, it had been started on this particular channel, + stopped, and now we're starting it again), if using the "files" + mode, then realtime will always be rechecked. If you are using + other modes, however, we will simply reattach to the external + running process which was playing moh earlier in the call. This + is a necessary compromise so that we don't end up with too many + background processes. 3. musiconhold.conf has a general section + now. It has one option: cachertclasses. If set to yes, then moh + classes found in realtime will be added to the in-memory list. + This has the advantage of not requiring database lookups each + time moh is started, but it has the disadvantage of not truly + being realtime. I have tested this for functionality, and it + passes. I also tested this under valgrind and there are no memory + problems reported under typical use. Special thanks to Sergee for + implementing this feature and enduring my complaints on the + bugtracker! (closes issue #11196, reported and patched by sergee) + +2007-11-28 00:24 +0000 [r89840-89915] Russell Bryant <russell@digium.com> + + * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 89893 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) | + 4 lines - update documentation for some of the goto functions to + note that they handle locking the channel as needed - update + ast_explicit_goto() to lock the channel as needed ........ + + * include/asterisk/channel.h: Document that the channel is not + locked when the send_digit_begin and end callbacks get called. + + * main/autoservice.c, /: Merged revisions 89886 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) | + 2 lines Don't do frame processing if ast_read() returned NULL. + ........ + + * channels/chan_iax2.c: Merge changes from + team/russell/iax2_frame_queue This patch is an optimization for + chan_iax2. This module is now heavily multi-threaded. However, + there is still a good number of globally shared resources that + prevent things from happen asynchronously. One of those things + was the global IAX frame queue. This queue was used to hold + frames that have been deferred for transmitting by another + thread, and frames that may need to get retransmitted. I changed + the frame queue to be per-call, since almost all of the frame + queue handling only cares about frames specific to a call number. + + * /, apps/app_queue.c: Merged revisions 89844 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) | + 3 lines Instead of depending on the return value of ast_true(), + explicitly set the eventwhencalled variable to 1. ........ + + * main/pbx.c, /: Merged revisions 89839 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) | + 2 lines Don't start/stop autoservice in pbx_extension_helper() + unless a channel exists ........ + +2007-11-27 23:11 +0000 [r89838] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 89837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov + 2007) | 12 lines Two changes with regards to the + 'eventwhencalled' option of queues.conf 1) Due to some signed vs. + unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes' + did exactly the same thing. Thus the sign change of the ast_true + call. 2) The vars2manager function overwrote a \n for every + channel variable it parsed, resulting in bizarre output for the + channel variables. This patch remedies this. (related to issue + #11385, however I'm not sure if this will actually be enough to + close it) ........ + +2007-11-27 22:42 +0000 [r89835] Russell Bryant <russell@digium.com> + + * channels/chan_misdn.c: Bring in a small change from + team/russell/chan_refcount This replaces tab completion code with + the use of a public function that does the same thing + +2007-11-27 22:14 +0000 [r89792] Steve Murphy <murf@digium.com> + + * main/pbx.c, pbx/pbx_config.c: closes issue #11294; missed the + conditional unlock of the contexts when the hash table is used + instead; also, used the ast_free_ptr as advised. + +2007-11-27 22:05 +0000 [r89791] Russell Bryant <russell@digium.com> + + * main/autoservice.c, main/pbx.c, /: Merged revisions 89790 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) | + 41 lines Merge changes from team/russell/autoservice_1.4 This set + of changes fixes an issue that was reported to me on IRC + yesterday. The user, d1mas, was using chan_zap for incoming calls + and was having DTMF recognition issues in some situations. + Specifically, he noticed that the problem occurred when using + DISA or WaitExten. He also noticed that when using Read, the + problem did not occur. His system also used DUNDi for dialplan + lookups. So, he theorized that if the DUNDi lookups blocked for + some period of time, that audio from the zap channel could get + lost. If the audio got lost, then it wouldn't be run through the + DTMF detector, and digits could get lost. He was correct, and the + following set of changes fixes the problem. However, the changes + go a little bit further than what was necessary to fix this exact + problem. 1) I updated pbx_extension_helper() to autoservice the + associated channel to handle cases where extension lookups may + take a long time. This would normally be a dialplan switch that + does some lookup over the network, such as the DUNDi or IAX2 + switches. This ensures that even while a DUNDi lookup is + blocking, the channel will be continuously serviced. 2) I made a + change to the autoservice code. This is actually something that + has bothered me for a long time. When a channel is in + autoservice, _all_ frames get thrown away. However, some frames + really shouldn't be thrown away. The most notable examples are + signalling (CONTROL) frames, and DTMF. So, this patch queues up + important frames while a channel is in autoservice. When + autoservice is stopped on the channel, the queued up frames get + stuck back on the channel so that they can get processed instead + of thrown away. 3) I made another change to the autoservice code + to handle the case where autoservice is started on channels + recursively. Previously, you could call ast_autoservice_start() + multiple times on a channel, and it would stop the first time + ast_autoservice_stop() gets called. Now, it will ensure that + autoservice doesn't actually stop until the final call to + ast_autoservice_stop(). ........ + +2007-11-27 21:10 +0000 [r89769-89772] Olle Johansson <oej@edvina.net> + + * main/dnsmgr.c, res/res_jabber.c, main/enum.c, main/asterisk.c: A + few more "moremanager" fixes + + * include/asterisk.h, main/asterisk.c, main/loader.c: More + "moremanager" fixes. Manager commands to check module status. + + * include/asterisk/manager.h: More "moremanager" changes - doxygen + docs and changing manager version (finally) before making more + dramatic changes. + + * channels/chan_iax2.c: More additions from the "moremanager" + branch, this time for IAX2. + +2007-11-27 20:24 +0000 [r89733] Mark Michelson <mmichelson@digium.com> + + * /: Blocked revisions 89727 via svnmerge ........ r89727 | + mmichelson | 2007-11-27 14:22:59 -0600 (Tue, 27 Nov 2007) | 6 + lines Changing some calls from free() to ast_free() since they + were allocated with ast_calloc(). (closes issue #11390, reported + and patched by Laureano) ........ + +2007-11-27 20:21 +0000 [r89721] Kevin P. Fleming <kpfleming@digium.com> + + * /, main/app.c: Merged revisions 89709 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007) + | 2 lines on second thought... revert all the other changes i've + made in app options parsing leaving only one: if an empty + argument is supplied for an option, set that argument pointer to + point to an empty string rather than NULL, so that the + application can do normal checks on it without worrying about it + being NULL ........ + +2007-11-27 20:17 +0000 [r89710] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: remove a duplicate manager event + +2007-11-27 19:50 +0000 [r89706] Olle Johansson <oej@edvina.net> + + * channels/chan_gtalk.c: Manager events from the "moremanager" + branch + +2007-11-27 19:47 +0000 [r89704] Kevin P. Fleming <kpfleming@digium.com> + + * /, main/app.c: Merged revisions 89701 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007) + | 2 lines generate a warning when an application option that + requires an argument is ignored due to lack of an argument + ........ + +2007-11-27 19:45 +0000 [r89698-89702] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Starting to merge changes from the + "moremanager" branch. Documentation will follow. + + * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: The + following patch with updates for trunk. Works much better in + trunk. Also by accident fixed a bad typo by a previous committer, + which actually made video calls not work fully... Merged + revisions 89630 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 + lines If we get a codec offer using a well-known payload type, + but using it for another codec that we don't know, Asterisk did + not remove that codec from the list. With this patch, we remove + the codec from audio and video rtp objects and deny it ever + existed. Thanks to lasse for testing. (closes issue #11376) + Reported by: lasse Patches: bug11376.txt uploaded by oej (license + 306) Tested by: lasse ........ + +2007-11-27 19:12 +0000 [r89683] Jason Parker <jparker@digium.com> + + * include/asterisk/strings.h: Add an S_COR macro, which is similar + to the existing S_OR macro, except with an additional boolean + arg. A hack such as: foo ? S_OR(bar, "baz") : "baz" becomes: + S_COR(foo, bar, "baz") + +2007-11-27 18:50 +0000 [r89682] Steve Murphy <murf@digium.com> + + * res/ael/ael.y, pbx/ael/ael-test/ref.ael-test11, + pbx/ael/ael-test/ref.ael-test20, 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, pbx/ael/ael-test/ref.ael-test18, + pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ref.ael-ntest10, res/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-ntest22, + res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test3, + pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, + pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex, + pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8: + made AEL 8-bit transparent; mainly the lexer was tossing chars + with the hi-order bit set. Not nice. Also, allow @ in extension + names, and a backslash, also. + +2007-11-27 17:01 +0000 [r89637] Joshua Colp <jcolp@digium.com> + + * main/utils.c: Ensure the value returned from ast_random is + between 0 and RAND_MAX on 64-bit platforms. (closes issue #11348) + Reported by: sperreault + +2007-11-27 16:13 +0000 [r89635] Russell Bryant <russell@digium.com> + + * /, configs/voicemail.conf.sample: Merged revisions 89634 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) | + 3 lines Add a note to the sample voicemail config noting that + when using IMAP storage, only the first format specified will be + attached to the message. ........ + +2007-11-27 15:41 +0000 [r89632] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_env.c: Merged revisions 89631 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007) + | 3 lines Default result of STAT should be "0" not "". Reported + via the -users mailing list, fixed by me. ........ + +2007-11-27 07:36 +0000 [r89625] Olle Johansson <oej@edvina.net> + + * /, configs/sip.conf.sample: Merged revisions 89624 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov + 2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304) + Reported by: pj ........ + +2007-11-27 06:47 +0000 [r89623] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, main/cdr.c, /, configs/cdr.conf.sample, + include/asterisk/cdr.h: Merged revisions 89622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 + line closes issue #11379; OK, this is an attempt to make both + sides happy. To the cdr.conf file, I added the option + 'unanswered', which defaults to 'no'. In this mode, you will see + a cdr for a call, whether it was answered or not. The disposition + will be NO ANSWER or ANSWERED, as appropriate. The src is as + you'd expect, the destination channel will be one of the channels + from the Dial() call, usually the last in the list if more than + one chan was specified. With unanswered set to 'yes', you will + still see this cdr entry in both cases. But in the case where the + dial timed out, you will also see a cdr for each line attempted, + marked NO ANSWER, with no destination channel name. The new + option defaults to 'no', so you don't see the pesky extra cdr's + by default, and you will not see the irritating 'not posted' + messages. ........ + +2007-11-26 23:15 +0000 [r89617-89621] Mark Michelson <mmichelson@digium.com> + + * pbx/ael/ael-test/ael-test19/extensions.ael, + pbx/ael/ael-test/ael-vtest13/extensions.ael, doc/osp.txt, + pbx/ael/ael-test/ael-test3/extensions.ael, + pbx/ael/ael-test/ref.ael-vtest13, + pbx/ael/ael-test/ael-test7/extensions.ael: Change all instances + of "CALLERID(number)" to "CALLERID(num)" for consistency's sake + (closes issue #11381, reported and patched by jon) + + * /, apps/app_playback.c: Merged revisions 89618 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov + 2007) | 7 lines After issuing a "say load new", if a caller hangs + up during the middle of playback of a number, app_playback will + continue to try to play the remaining files. With this change, no + more files will be played back upon hangup. (closes issue #11345, + reported and patched by IgorG) ........ + + * /: Blocked revisions 89616 via svnmerge ........ r89616 | + mmichelson | 2007-11-26 17:02:30 -0600 (Mon, 26 Nov 2007) | 5 + lines After issuing a "say load new" tons of warning messages are + printed out to the CLI every time do_say in app_playback is + called. Removing these warnings ........ + +2007-11-26 22:52 +0000 [r89615] Russell Bryant <russell@digium.com> + + * configure, configure.ac: Update the configure script check for + libpri to check for the newest function that was just added. + Cresl1n, please keep this in mind when making these changes to + libpri or libss7. + +2007-11-26 21:23 +0000 [r89613] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c, configs/sip.conf.sample: Rename + "limitonpeer" to "counteronpeer" since the call-limit is + deprecated. Both still works in this version. + +2007-11-26 21:14 +0000 [r89612] Joshua Colp <jcolp@digium.com> + + * main/dial.c, /: Merged revisions 89610 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2 + lines Fix issues with async dialing with an application + executing. The application has to be terminated and control + returned to the thread before hanging things up. (issue #BE-252) + ........ + +2007-11-26 21:12 +0000 [r89606-89611] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Formatting, doxygenification + + * channels/chan_sip.c: Formatting changes, cleaning up some code + + * include/asterisk/doxyref.h, channels/chan_sip.c: Start using + Doxygen groupings to group variables and defines. + + * apps/app_meetme.c, UPGRADE.txt, CHANGES, main/cli.c: - Mark + "concise" as deprecated - Restructure other changes to + UPGRADE.txt and CHANGES We're still looking for scripts that + replace asterisk -rx "show shannels concise" by using the manager + interface, but still produces the same output. Anyone? + +2007-11-26 18:11 +0000 [r89600-89602] Joshua Colp <jcolp@digium.com> + + * res/res_features.c, apps/app_queue.c: Perform some module use + counting audits. This is now done outside the scope of the + application/dialplan function so they do not need to worry about + it. + + * /, res/res_features.c: Merged revisions 89599 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6 + lines Add module counting removal for error conditions. (closes + issue #11333) Reported by: Laureano Patches: + res_features_v2.c.patch uploaded by Laureano (license 265) + ........ + +2007-11-26 17:49 +0000 [r89596] Russell Bryant <russell@digium.com> + + * main/pbx.c, /: Merged revisions 89594 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) | + 3 lines Add channel locking to a function that needed to be doing + it. This is just a little something I noticed while working on a + completely unrelated issue. ........ + +2007-11-26 17:46 +0000 [r89595] Steve Murphy <murf@digium.com> + + * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c: closes + issue #11341; made changes to make utils again right with the + MTX_PROFILE world. + +2007-11-26 17:38 +0000 [r89593] Joshua Colp <jcolp@digium.com> + + * /, pbx/pbx_config.c: Merged revisions 89592 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6 + lines Use ast_free to free memory, or else we shall implode if + MALLOC_DEBUG is enabled. (closes issue #11347) Reported by: ys + Patches: pbx.pbx_config.c.diff uploaded by ys (license 281) + ........ + +2007-11-26 17:26 +0000 [r89591] Steve Murphy <murf@digium.com> + + * main/hashtab.c: closes issue #11356; Many thanks to snuffy for + his code review and changes to cut down duplication. I tested + this against hashtest, and it passes. I reviewed the changes, and + they look reasonable. I had to remove a few const decls to make + things compile on my workstation, + +2007-11-26 17:25 +0000 [r89590] Russell Bryant <russell@digium.com> + + * Makefile: make sure we check to see if the configure script has + been executed on a new checkout or after a distclean + +2007-11-26 17:23 +0000 [r89589] Joshua Colp <jcolp@digium.com> + + * /, apps/app_mixmonitor.c: Merged revisions 89587 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov + 2007) | 6 lines Close the audio file before sending it to the + post processing application. (closes issue #11357) Reported by: + reformed Patches: mixmonitor.patch uploaded by reformed (license + 330) ........ + +2007-11-26 17:21 +0000 [r89588] Kevin P. Fleming <kpfleming@digium.com> + + * /, main/app.c, apps/app_controlplayback.c: Merged revisions 89586 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007) + | 2 lines when parsing application options that take arguments, + don't indicate that the option was supplied unless a + non-zero-length argument was found for it ........ + +2007-11-26 16:24 +0000 [r89583] Steve Murphy <murf@digium.com> + + * main/pbx.c, CHANGES, configs/extensions.conf.sample: Thanks to + pnlarsson for noting the spelling error in the cli commands. + Also, added some verbage about the new algorithm to CHANGES. + +2007-11-26 16:20 +0000 [r89582] Joshua Colp <jcolp@digium.com> + + * main/utils.c: Revert change for 11348 until it can be looked at + even more. + +2007-11-26 15:50 +0000 [r89581] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 89580 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov + 2007) | 6 lines Revert vmu->email back to an empty string if it + was empty when imap_store_file was called. This prevents sending + a duplicate e-mail. (closes issue #11204, reported by spditner, + patched by me) ........ + +2007-11-26 15:36 +0000 [r89570-89578] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 89577 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6 + lines If channel allocation fails because the alert pipe could + not be created also free the scheduler context. (closes issue + #11355) Reported by: eliel Patches: main.channel.c.patch uploaded + by eliel (license 64) ........ + + * main/utils.c: Make the behavior of using /dev/urandom for random + numbers the same as random(). (closes issue #11348) Reported by: + sperreault Patches: ast_random2.diff uploaded by sperreault + (license 252) + + * channels/chan_sip.c: Instead of printing out one codec in sip + show channels print out all of the native ones (this is for + video). (closes issue #11366) Reported by: ovi + + * /, apps/app_meetme.c: Merged revisions 89571 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4 + lines When unloading app_meetme destroy any auto created contexts + created by SLA. (closes issue #11367) Reported by: eliel ........ + + * apps/app_controlplayback.c: Don't crash if the 'o' option of + ControlPlayback is used without any value. (closes issue #11375) + Reported by: johan + +2007-11-25 21:12 +0000 [r89564-89566] Olle Johansson <oej@edvina.net> + + * channels/chan_usbradio.c: Formatting changes + + * main/channel.c, include/asterisk/channel.h: Try to get channel.h + and channel.c aligned in regards to ast_set_callerid as well as + change name of variables to follow the rest of the naming. + +2007-11-25 17:50 +0000 [r89560-89561] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/res_odbc.h, res/res_config_odbc.c, /, + res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions + 89559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) + | 14 lines We previously attempted to use the ESCAPE clause to + set the escape delimiter to a backslash. Unfortunately, this does + not universally work on all databases, since on databases which + natively use the backslash as a delimiter, the backslash itself + needs to be delimited, but on other databases that have no + delimiter, backslashing the backslash causes an error. So the + only solution that I can come up with is to create an option in + res_odbc that explicitly specifies whether or not backslash is a + native delimiter. If it is, we use it natively; if not, we use + the ESCAPE clause to make it one. Reported by: elguero Patch by: + tilghman (Closes issue #11364) ........ + + * channels/chan_sip.c: Typo (someone needs to test compile before + committing his changes) + +2007-11-25 12:18 +0000 [r89551-89557] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: More doxygen changes + + * channels/chan_sip.c: Housekeeping + + * channels/chan_sip.c: Formatting, doxygen updates + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: - + Deprecate "call-limit" in chan_sip. No other channel driver + enforces call-limits and we now have the groupcount system to + implement call-limits in the dialplan. You can use the "setvar" + option in realtime/sip.conf to set limits per device. - Implement + "callcounter" as a new option to enable the call counting we need + to report device status to queue, manager and SIP subscriptions. + The call counter setting is now enabled in the code by setting + the device call-limit to 999. When we remove the call limit, we + can simply enable this with a boolean setting. + + * channels/chan_sip.c, include/asterisk/channel.h: Housekeeping... + - Fix typo in chan_sip - Remove changes to caller ID structure, + moving it to branch (russellb) + +2007-11-24 21:00 +0000 [r89547] Steve Murphy <murf@digium.com> + + * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c, + configs/extensions.conf.sample: closes issue #11363; where the + pattern _20x. buried in an included context, didn't match 2012; + There were a small set of problems to fix: 1. I needed NOT to + score patterns unless you are at the end of the data string. 2. + Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize + the patterns in the trie to caps. 3. When a pattern ends with dot + or exclamation, CANMATCH/MATCHMORE should always report this + pattern, no matter the length. With this commit, I also supplied + the wish of Luigi, where the user can select which pattern + matching algorithm to use, the old (legacy) pattern matcher, or + the new, trie based matcher. The OLD matcher is the default. A + new [general] section variable, extenpatternmatchnew, is added to + the extensions.conf, and the example config has it set to false. + If true, the new matcher is used. In all other respects, the + context/exten structs are the same; the tries and hashtabs are + formed, but in the new mode the tries are not used. A new CLI + command 'dialplan set extenpatternmatch true/false' is provided + to allow switching at run time. I beg users that are forced to + return to the old matcher to please report the reason in the bug + tracker. Measured the speed benefit of the new matcher against an + impossibly large context with 10,000 extensions: the new matcher + is 374 times faster. + +2007-11-24 17:07 +0000 [r89546] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_adsi.c: Merged revisions 89545 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89545 | tilghman | 2007-11-24 10:59:59 -0600 (Sat, 24 Nov 2007) + | 5 lines Free some frames that would otherwise leak on error. + Reported by: Laureano Patch by: Laureano,tilghman (Closes issue + #11351) ........ + +2007-11-24 16:53 +0000 [r89544] Steve Murphy <murf@digium.com> + + * main/app.c: Added <sys/file.h> include to allow trunk to compile. + Hope this doesn't louse thing up. + +2007-11-24 13:57 +0000 [r89542-89543] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_h323.c: remove a DEBUG_THREADS message that + accesses private lock fields. If needed, the code to extract this + information should be implemented in some generic header or + library and the function called here. (closed bug #11362) + + * main/acl.c, main/http.c, main/app.c: remove some unnecessary + includes + +2007-11-24 06:24 +0000 [r89535-89541] Tilghman Lesher <tlesher@digium.com> + + * /, main/app.c, apps/app_voicemail.c: Merged revisions 89540 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007) + | 9 lines Currently, zero-length voicemail messages cause a + hangup in VoicemailMain. This change fixes the problem, with a + multi-faceted approach. First, we do our best to avoid these + messages from being created in the first place, and second, if + that fails, we detect when the voicemail message is zero-length + and avoid exiting at that point. Reported by: dtyoo Patch by: + gkloepfer,tilghman (Closes issue #11083) ........ + + * main/manager.c, /: Merged revisions 89536 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89536 | tilghman | 2007-11-23 11:18:26 -0600 (Fri, 23 Nov 2007) + | 10 lines Up until this point, the XML output of the manager has + been technically invalid, due to the repetition of certain + parameters in a single event. This caused various issues for XML + parsers, some of which refused to parse at all, given the + invalidity of the rendered XML. So this commit fixes the XML + output, ensuring that each entity parameter has a unique name, + thus ensuring valid XML. Reported by: msetim Patch by: tilghman + (Closes issue #10220) ........ + + * res/res_config_odbc.c, /: Merged revisions 89534 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89534 | tilghman | 2007-11-23 11:05:10 -0600 (Fri, 23 + Nov 2007) | 5 lines Use ESCAPE clause for the first parameter, + not just 2nd-Nth parameters. Reported by: apsaras Patch by: + tilghman (Closes issue #11353) ........ + +2007-11-23 15:54 +0000 [r89532-89533] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_oss.c: put in the necessary hooks for video support + in the console. This is a NOP as far as the current code is + concerned, but there is already support in ./configure and the + Makefiles for the various libraries used by console_video.c (not + yet in the tree) so addition is trivial. + + * channels/chan_sip.c: set rtpmap video info according to what is + read from SDP; make the format explicit in a debug message; print + the audio instead of aggregated peer capability in a debugging + msg. + +2007-11-23 09:40 +0000 [r89531] Olle Johansson <oej@edvina.net> + + * include/asterisk/channel.h: Let's start with implementing the + base architecture for UTF8 caller ID's so we can handle multiple + formats properly. This is not carved in stone, but a proposal to + start with. We need to add support for transliterations as well + as UTF8 handling, propably with libiconv. Murf is looking into + that for the dialplan. + +2007-11-23 09:03 +0000 [r89530] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/image.h, formats/format_jpeg.c: formatting + cleanup on the header, normalization of the assignment of + descriptor fields. + +2007-11-23 02:37 +0000 [r89529] Russell Bryant <russell@digium.com> + + * configs/agents.conf.sample, /: Merged revisions 89527 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89527 | russell | 2007-11-22 12:29:41 -0500 (Thu, 22 Nov 2007) | + 3 lines mvanbaak pointed out a spelling error in this sample + configuration file. While I was at it, I went ahead and tweaked + it a little bit more. ........ + +2007-11-22 07:10 +0000 [r89514-89526] Luigi Rizzo <rizzo@icir.org> + + * doc/CODING-GUIDELINES: new info on the management of headers + + * apps/app_echo.c, apps/app_sendtext.c, apps/app_verbose.c, + apps/app_milliwatt.c: more header removal + + * include/asterisk/channel.h: formatting cleanup + + * include/asterisk.h, apps/app_read.c, apps/app_record.c, + apps/app_echo.c, apps/app_readexten.c, + include/asterisk/channel.h, apps/app_system.c, + apps/app_transfer.c, res/ael/pval.c, include/asterisk/app.h, + apps/app_dumpchan.c, include/asterisk/module.h, apps/app_url.c, + include/asterisk/pbx.h, apps/app_senddtmf.c, pbx/pbx_config.c, + apps/app_mixmonitor.c, apps/app_stack.c, apps/app_verbose.c, + apps/app_milliwatt.c, apps/app_cdr.c, apps/app_while.c: shuffle a + little bit the content of header files to reduce dependencies. In + this commit: - move the ast_register/unregister_app functions to + module.h to avoid the need to include pbx.h for the simpler apps; + - move the ast_group structure to channel.h to remove the + dependency of app.h on linkedlists.h Note, this is a long process + that I am doing in small steps. The main difficulty is that now + for each subsystem we have a single header (e.g. channel.h) + included by the subsystem provider (usually one file, e.g. + channel.c) and by its clients (dozens of them, e.g. we have some + 70+ apps and 30+ functions). This requires the clients to include + all the extra headers required by the provider (eg. lock.h, + linkedlists.h, definitions of substructures...) even though many + of the clients would be just happy with opaque struct + declarations and function prototypes. The long term plan is to + eventually rectify this structure so that the compilation can + become faster, and also APIs are more stable. + + * funcs/func_md5.c, funcs/func_module.c, funcs/func_blacklist.c, + apps/app_url.c, funcs/func_sha1.c, funcs/func_global.c: remove + some useless includes + + * include/asterisk/audiohook.h, apps/app_dictate.c, + apps/app_readexten.c, apps/app_directory.c, apps/app_senddtmf.c, + apps/app_mixmonitor.c, apps/app_stack.c, + apps/app_controlplayback.c: more removal of redundant headers + + * apps/app_read.c, apps/app_echo.c, apps/app_record.c, + apps/app_userevent.c, apps/app_image.c, apps/app_system.c, + apps/app_verbose.c, apps/app_milliwatt.c, apps/app_playback.c, + apps/app_while.c: remove redundant headers + + * main/file.c, main/netsock.c: more removal of fcntl.h and other + system headers + + * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c, + codecs/codec_speex.c, codecs/codec_alaw.c, codecs/codec_adpcm.c, + res/res_crypto.c, codecs/codec_g726.c, apps/app_test.c, + formats/format_ogg_vorbis.c, codecs/codec_gsm.c, res/res_agi.c, + apps/app_mp3.c, main/app.c, codecs/codec_ulaw.c, + codecs/codec_ilbc.c: remove a number of #include <fcntl.h> which + are either useless or done elsewhere + + * formats/format_sln.c, formats/format_wav.c, + formats/format_ogg_vorbis.c, include/asterisk/_private.h, + formats/format_wav_gsm.c, formats/format_ilbc.c, + include/asterisk/file.h, formats/format_vox.c, + formats/format_pcm.c, main/file.c, formats/format_h263.c, + formats/format_g723.c, formats/format_h264.c, + include/asterisk/frame.h, formats/format_jpeg.c, + formats/format_g726.c, formats/format_gsm.c, + formats/format_g729.c: implement the split of file.h and + mod_format.h + + * include/asterisk/mod_format.h (added): Add a specific header for + providers of file and format handling routines, moving here + structs and function declarations formerly in file.h + +2007-11-21 23:54 +0000 [r89513] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, channels/chan_sip.c, channels/chan_skinny.c, + res/res_features.c, apps/app_queue.c, channels/chan_iax2.c: + closes issue #11285, where an unload of a module that creates a + dialplan context, causes a crash when you do a 'dialplan show' of + that context. This is because the registrar string is defined in + the module, and the stale pointer is traversed. The reporter + offered a patch that would always strdup the registrar string, + which is practical, but I preferred to destroy the created + contexts in each module where one is created. That seemed more + symmetric. There were only 6 place in asterisk where this is + done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, + and app_queue. The two apps destroyed the context, but left the + contexts. All is fixed now and unloads should be dialplan + friendly. + +2007-11-21 23:24 +0000 [r89511-89512] Luigi Rizzo <rizzo@icir.org> + + * funcs/func_rand.c, cdr/cdr_sqlite3_custom.c, apps/app_readfile.c, + channels/chan_local.c, apps/app_record.c, funcs/func_strings.c, + apps/app_sayunixtime.c, apps/app_test.c, + apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c, + apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c, + channels/chan_iax2.c, apps/app_exec.c, pbx/pbx_loopback.c, + pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_dumpchan.c, + apps/app_zapscan.c, apps/app_zapras.c, pbx/pbx_realtime.c, + channels/chan_alsa.c, apps/app_amd.c, apps/app_url.c, + apps/app_externalivr.c, cdr/cdr_odbc.c, apps/app_dial.c, + funcs/func_timeout.c, apps/app_page.c, apps/app_privacy.c, + channels/chan_agent.c, apps/app_disa.c, apps/app_morsecode.c, + channels/iax2-provision.c, funcs/func_cut.c, + apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c, + apps/app_playback.c, funcs/func_curl.c, channels/chan_misdn.c, + apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c, + channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c, + pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c, + apps/app_voicemail.c, channels/chan_unistim.c, + channels/chan_vpb.cc, apps/app_meetme.c, apps/app_authenticate.c, + apps/app_readexten.c, funcs/func_vmcount.c, + channels/chan_gtalk.c, cdr/cdr_pgsql.c, apps/app_followme.c, + cdr/cdr_radius.c, apps/app_controlplayback.c, cdr/cdr_csv.c, + channels/chan_phone.c, funcs/func_enum.c, apps/app_osplookup.c, + funcs/func_odbc.c, apps/app_mp3.c, apps/app_minivm.c, + apps/app_rpt.c, channels/chan_mgcp.c, apps/app_parkandannounce.c, + apps/app_while.c, apps/app_adsiprog.c, apps/app_nbscat.c, + funcs/func_version.c, funcs/func_db.c, channels/chan_zap.c, + apps/app_read.c, channels/chan_sip.c, apps/app_festival.c, + apps/app_waitforsilence.c, funcs/func_lock.c, pbx/pbx_lua.c, + apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c, + channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c, + channels/chan_jingle.c, channels/chan_usbradio.c, + apps/app_channelredirect.c, apps/app_flash.c, + apps/app_directed_pickup.c, funcs/func_blacklist.c, + channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_sms.c, + channels/chan_nbs.c, apps/app_senddtmf.c, funcs/func_callerid.c, + apps/app_verbose.c, apps/app_stack.c, pbx/pbx_gtkconsole.c: + remove another set of redundant #include "asterisk/options.h" + + * main/udptl.c, main/autoservice.c, main/frame.c, res/res_snmp.c, + main/say.c, res/res_features.c, main/devicestate.c, main/utils.c, + res/res_musiconhold.c, res/res_jabber.c, main/indications.c, + main/enum.c, res/res_config_sqlite.c, main/config.c, + main/loader.c, main/term.c, main/cli.c, main/io.c, + main/channel.c, main/cdr.c, main/dial.c, res/res_smdi.c, + res/res_config_odbc.c, main/manager.c, res/res_agi.c, + main/http.c, main/logger.c, res/res_realtime.c, main/app.c, + main/image.c, main/dns.c, main/db.c, res/res_speech.c, + main/sched.c, main/pbx.c, res/res_config_pgsql.c, main/dnsmgr.c, + main/translate.c, res/res_crypto.c, res/res_adsi.c, + main/jitterbuf.c, main/acl.c, formats/format_ogg_vorbis.c, + res/res_ael_share.c, res/res_monitor.c, main/rtp.c, + main/netsock.c, main/srv.c, main/hashtab.c, main/privacy.c, + main/adsistub.c, main/abstract_jb.c, main/file.c, + main/callerid.c, main/astmm.c, main/audiohook.c, + formats/format_g726.c, main/asterisk.c, res/res_odbc.c, + main/dsp.c: remove a bunch of useless #include "options.h" + +2007-11-21 22:37 +0000 [r89509-89510] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Remove unneccessary explicit case for BRI + + * channels/chan_zap.c: Take some debug code out :-) + +2007-11-21 22:20 +0000 [r89508] Luigi Rizzo <rizzo@icir.org> + + * main/cygload.c: add a missing return + +2007-11-21 22:07 +0000 [r89507] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Add BRI support to chan_zap + +2007-11-21 21:30 +0000 [r89506] Luigi Rizzo <rizzo@icir.org> + + * utils/Makefile, configure, configure.ac: enable support for stack + backtrace for stuff built in utils/ (this was present in the main + tree but forgotten here). + +2007-11-21 20:38 +0000 [r89505] Steve Murphy <murf@digium.com> + + * main/pbx.c: closes issue #11290; the proposed patch was a good + guess, and would solve the bug to some extent, but was really + masking the real issue, that there were bad entries in the table. + This fix removes the condition that the hashtab updates be done + on exten removal only when the pattern_tree was present, which is + silly. The operations that apply to the pattern tree are instead + made conditional. Also, threw back in routines that kpfleming + deleted because of probs in the 64-bit world. Tested on both 32 + and 64-bit machines (compile). Tested the reload problem with + over 20 reloads, and no problems. If you find more problems, + please reopen 11290. + +2007-11-21 20:22 +0000 [r89504] Terry Wilson <twilson@digium.com> + + * res/res_features.c: Simplify comparison in parking fix + +2007-11-21 19:28 +0000 [r89494-89496] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 89495 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89495 | mmichelson | 2007-11-21 13:27:51 -0600 (Wed, 21 Nov + 2007) | 3 lines Fix a small error I made in my previous commit + ........ + + * /, apps/app_queue.c: Merged revisions 89493 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov + 2007) | 5 lines Changing an inaccurate debug message to be less + inaccurate. Under the circumstances, this message would always + report that there were 0 members available, even though that may + not be true. ........ + +2007-11-21 19:20 +0000 [r89492] Terry Wilson <twilson@digium.com> + + * /, res/res_features.c: Merged revisions 89491 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89491 | twilson | 2007-11-21 12:59:27 -0600 (Wed, 21 Nov 2007) | + 4 lines If a channel gets masqueraded in the middle of a park, + don't play the announcement to the masqueraded channel, and dial + back to the original channel on timeout. ........ + +2007-11-21 18:52 +0000 [r89490] Russell Bryant <russell@digium.com> + + * main/dsp.c: Remove obsolete OLD_DSP_ROUTINES code. Also, remove + the FAX_DETECT define and only do the calculations if fax + detection is enabled on the dsp. (closes issue #11331) Reported + by: dimas Patches: dsp.patch uploaded by dimas (license 88) + +2007-11-21 18:38 +0000 [r89489] Tilghman Lesher <tlesher@digium.com> + + * apps/app_read.c, UPGRADE.txt, CHANGES: Change Read to set + READSTATUS as an indication of the result Also, some cleanup to + CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman + (Closes issue #11004) + +2007-11-21 18:24 +0000 [r89488] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c: fix a small gramatical error in a comment + +2007-11-21 18:19 +0000 [r89487] Mark Michelson <mmichelson@digium.com> + + * main/utils.c: There existed about a 1 in 4 billion chance that + reading from /dev/urandom would return LONG_MIN (1 in 9 + quintillion if using 64-bit longs). Since there is no positive + equivalent of LONG_MIN, the result of labs() in this case is + unpredictable. This fixes that situation. (closes issue #11336, + reported and patched by sperreault) + +2007-11-21 16:24 +0000 [r89484] Russell Bryant <russell@digium.com> + + * channels/chan_unistim.c: Fix some code that was supposed to + ensure that a buffer was terminated, but was writing to the wrong + byte. Also, remove some non-thread safe test code. (closes issue + #11317) Reported by: IgorG Patches: unistim-2.patch uploaded by + IgorG (license 20) - additional changes by me + +2007-11-21 16:08 +0000 [r89483] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c: I introduced a deadlock avoidance into 1.4, which I + attempted to port to trunk as well. Unfortunately, since trunk + uses read/write locks for the context lock, it means that I have + actually *introduced* a deadlock condition since they are not + recursive. Removing this change for now and will look into + introducing a different one. + +2007-11-21 16:07 +0000 [r89480-89482] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk.h, include/asterisk/compat.h, utils/ael_main.c, + utils/conf2ael.c: move these forward declarations back to + asterisk.h where they belong... even though asterisk.h includes + compat.h, these declarations have nothing to do with the being + platform-compatible and are directly related to being part of + Asterisk + + * channels/chan_usbradio.c: get this to actually compile... + + * main/pbx.c: remove some debugging code that doesn't compile on + 64-bit platforms + +2007-11-21 15:17 +0000 [r89478-89479] Steve Murphy <murf@digium.com> + + * res/res_features.c: OOOps! All the debug stuff I inserted was + accidentally committed. I hereby revert it. + + * main/hashtab.c, res/res_features.c: closes issue #11265; Thanks + to snuffy for his work on neatening up the code and removing + duplicated code. + +2007-11-21 08:28 +0000 [r89475-89477] Luigi Rizzo <rizzo@icir.org> + + * channels/gentone-ulaw.c (removed): remove this file, it is not + used anywhere. + + * main/astmm.c: add missing paths.h + + * configure, include/asterisk/autoconfig.h.in, configure.ac: add + check for video4linux + +2007-11-21 01:09 +0000 [r89474] Steve Murphy <murf@digium.com> + + * main/pbx.c: A free in add_pri was ultimately the source of the + grief we were having with parking. This set of changes fixes that + problem, and introduces some more error messages, and puts debugs + into ifdefs for what could be short-term usage. Txs to Terry W. + for his help, guidance, and especially patience. + +2007-11-21 00:23 +0000 [r89472-89473] Luigi Rizzo <rizzo@icir.org> + + * main/sha1.c, agi/eagi-test.c, utils/smsq.c, utils/hashtest2.c, + main/minimime/mm.h, utils/check_expr.c: more header + removal/normalization + + * configure, include/asterisk/autoconfig.h.in, configure.ac: X11 + checks (at least some - for other platforms with unusual X11 + locations you might need to add more directories) + +2007-11-21 00:21 +0000 [r89470] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c, CHANGES: Merge changes from + team/russell/sla_trunk_moh ... * Added the ability to specify the + music on hold class used to play into the conference when there + is only one member and the M option is used. * Added the ability + to specify a music on hold class to play instead of ringing for + the SLATrunk application. (patched by me, and tested internally) + +2007-11-21 00:20 +0000 [r89469] Luigi Rizzo <rizzo@icir.org> + + * makeopts.in: complete support for X11 + +2007-11-20 23:29 +0000 [r89467-89468] Tilghman Lesher <tlesher@digium.com> + + * apps/app_meetme.c, cdr/cdr_sqlite.c, pbx/pbx_lua.c: Make trunk + build again + + * main/say.c: Add support for new recorded character sounds Closes + issue #5208 + +2007-11-20 23:16 +0000 [r89465-89466] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c, + apps/app_dictate.c, apps/app_test.c, apps/app_ices.c, + apps/app_followme.c, channels/chan_iax2.c, main/config.c, + main/loader.c, main/cli.c, cdr/cdr_csv.c, main/channel.c, + main/manager.c, pbx/pbx_spool.c, include/asterisk/compat.h, + res/res_agi.c, apps/app_minivm.c, main/logger.c, main/http.c, + main/app.c, main/image.c, apps/app_directory.c, main/db.c, + cdr/cdr_custom.c, apps/app_adsiprog.c, apps/app_dial.c, + include/asterisk/utils.h, include/asterisk.h, main/pbx.c, + channels/chan_sip.c, res/res_crypto.c, + include/asterisk/channel.h, res/res_monitor.c, + include/asterisk/paths.h, main/file.c, apps/app_sms.c, + include/asterisk/ael_structs.h, pbx/pbx_config.c, + apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c: + move asterisk/paths.h outside asterisk.h and into those files who + really need it. + + * main/pbx.c, include/asterisk.h, main/frame.c, main/dnsmgr.c, + main/threadstorage.c, main/devicestate.c, + include/asterisk/_private.h (added), main/astobj2.c, + main/loader.c, main/term.c, main/cli.c, main/channel.c, + main/manager.c, main/logger.c, build_tools/strip_nonapi, + main/event.c, main/asterisk.c, main/db.c: move internal function + declarations to include/asterisk/_private.h + +2007-11-20 19:29 +0000 [r89464] Russell Bryant <russell@digium.com> + + * configure, configure.ac: i got a little carried away with commas + ... + +2007-11-20 19:28 +0000 [r89463] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/module.h, build_tools/make_buildopts_h, + main/loader.c: switch compile-time option checking to string + storage mode in this branch too + +2007-11-20 19:11 +0000 [r89460] Russell Bryant <russell@digium.com> + + * configure, configure.ac: fix the zaptel configure script check + +2007-11-20 18:20 +0000 [r89459] Luigi Rizzo <rizzo@icir.org> + + * acinclude.m4: the 'version' is now $7 not $6 (wait a bit before + regenerating configure, i have more changes) + +2007-11-20 17:59 +0000 [r89458] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c, /: Merged revisions 89457 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89457 | mmichelson | 2007-11-20 11:50:31 -0600 (Tue, 20 Nov + 2007) | 9 lines According to comments in main/pbx.c, it is + essential that if we are going to lock the conlock as well as the + hints lock, it must be locked in that respective order. In order + to prevent a potential deadlock, we need to lock the conlock + prior to locking the hints lock in ast_hint_state_changed (see + the call stack example on issue #11323 for how this can happen). + (closes issue #11323, reported by eelcob, suggestion for patch by + eelcob, patch by me) ........ + +2007-11-20 17:11 +0000 [r89454-89455] Luigi Rizzo <rizzo@icir.org> + + * makeopts.in: prepare to support console_video + + * apps/Makefile, Makefile.moddir_rules, pbx/Makefile, res/Makefile, + channels/Makefile: Fix building of modules under cygwin. After + this commit we can actually load modules under windows, and we + can start debugging more interesting problems related to the load + order and functionality of modules. + +2007-11-20 16:11 +0000 [r89453] Mark Michelson <mmichelson@digium.com> + + * configs/sip.conf.sample: Changed occurrences of "busy-level" to + "busylevel" in sip.conf.sample in light of commit 89441. Thanks + to pj for pointing out the need for this (closes issue #11307, + reported by pj) + +2007-11-20 15:39 +0000 [r89452] Luigi Rizzo <rizzo@icir.org> + + * configure, configure.ac, acinclude.m4: add an argument for extra + headers to AC_EXT_LIB_CHECK, and on passing simplify the code. + Too bad that every time we need to regenerate configure... + +2007-11-20 15:30 +0000 [r89451] Steve Murphy <murf@digium.com> + + * /, doc/tex/queues-with-callback-members.tex: Merged revisions + 89450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1 + line closes issue #11324; break statements missing in switch + cases. ........ + +2007-11-20 15:00 +0000 [r89449] Joshua Colp <jcolp@digium.com> + + * main/translate.c: Minor documentation tweak and if an incorrect + parameter is given to core show translation return the usage + information. (closes issue #11316) Reported by: eliel Patches: + translate.c.patch uploaded by eliel (license 64) + +2007-11-20 14:54 +0000 [r89448] Luigi Rizzo <rizzo@icir.org> + + * configure, acinclude.m4: comment a bit the code in acinclude.m4 + There is still a lot of code to clean up there, but hopefully + this should clarify what goes on in there. + +2007-11-20 14:49 +0000 [r89447] Joshua Colp <jcolp@digium.com> + + * channels/h323/ast_h323.cxx: Include the compatibility header file + in ast_h323.cxx for compatibility reasons. (closes issue #11311) + Reported by: falves11 + +2007-11-20 14:44 +0000 [r89444-89446] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Fix sip show history. Closes issue #11312 + + * channels/chan_sip.c: Change terminology a bit for CLI commands + handling SIP channels/calls/dialogs/whatever. Closes issue #11312 + +2007-11-20 07:42 +0000 [r89443] Luigi Rizzo <rizzo@icir.org> + + * Makefile, main/Makefile, Makefile.moddir_rules: initial makefile + changes to build loadable modules under cygwin (not complete yet + - still need to sort out dependecies on res_*) + +2007-11-20 00:17 +0000 [r89442] Steve Murphy <murf@digium.com> + + * main/pbx.c: Get rid of some debug messages in pbx.c + +2007-11-19 23:24 +0000 [r89441] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c, CHANGES: Changed the "busy-level" option in + sip.conf to "busylevel" to be more parallel with the SIPPEER() + argument of the same name. The deprecation procedure is not being + used here since this is a trunk-only option. (closes issue + #11307, reported by pj, patched by me) + +2007-11-19 23:03 +0000 [r89439-89440] Russell Bryant <russell@digium.com> + + * include/asterisk/module.h: Be a bit more pedantic about the type + for holding the md5 sum for the build options. Also, doxygenify + the comment. + + * funcs/func_sysinfo.c: Make the SYSINFO documentation reflect + which options were compiled in + +2007-11-19 22:55 +0000 [r89438] Steve Murphy <murf@digium.com> + + * main/pbx.c: These changes were made in response to + niklas@tese.se's letter of 11-17-2007, where he had 20 and 201 in + two different contexts, included in the same context. In that + particular case, we were behaving the same as 1.4, but after + experimenting, I quickly found that if 20 and 201 were in the + same extension, 1.4 would return 201, and this code returns 20. + These changes now enable the current code to replicate the + behavior of 1.4 in respect to MATCHMORE in cases like this. + +2007-11-19 21:18 +0000 [r89430-89433] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_vpb.cc, channels/misdn_config.c, main/dsp.c: + another few errno.h removals + + * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c, + apps/app_meetme.c, pbx/pbx_ael.c, pbx/pbx_lua.c, + pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_externalivr.c, + apps/app_directory.c, apps/app_system.c, pbx/pbx_config.c, + apps/app_milliwatt.c: more errno.h removal + + * funcs/func_sysinfo.c: remove unnecessary headers + + * funcs/func_base64.c, funcs/func_volume.c: remove some unnecessary + includes. + +2007-11-19 20:13 +0000 [r89429] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: Change delimiter of SIPPEER to be comma + (instead of pipe) and further deprecate the old ':' delimiter + Reported by: pj Patch by: tilghman Closes issue #11305 + +2007-11-19 19:51 +0000 [r89424-89428] Luigi Rizzo <rizzo@icir.org> + + * codecs/codec_lpc10.c, codecs/codec_a_mu.c, codecs/codec_g722.c, + codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c, + codecs/codec_g726.c, codecs/codec_gsm.c, codecs/codec_ulaw.c, + codecs/codec_ilbc.c, codecs/codec_zap.c: remove some useless + includes from codecs + + * formats/format_ilbc.c, formats/format_sln.c, + formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, + formats/format_ogg_vorbis.c, formats/format_g723.c, + formats/format_h263.c, formats/format_h264.c, + formats/format_wav_gsm.c, formats/format_g726.c, + formats/format_jpeg.c, formats/format_gsm.c, + formats/format_g729.c: format handlers don't need network, lock, + channel and scheduler headers + + * include/asterisk.h, include/asterisk/compat.h, + include/asterisk/lock.h, utils/extconf.c, + include/asterisk/abstract_jb.h: move the declaration of struct + ast_channel ast_frame and ast_module to compat.h so it is always + available - hopefully this will let us reduce the number of + inclusions of channel.h and frame.h + + * main/udptl.c, main/autoservice.c, funcs/func_rand.c, + cdr/cdr_sqlite3_custom.c, main/frame.c, funcs/func_module.c, + main/threadstorage.c, main/say.c, funcs/func_env.c, + funcs/func_strings.c, main/devicestate.c, + cdr/cdr_adaptive_odbc.c, main/indications.c, main/config.c, + main/loader.c, main/term.c, main/cli.c, funcs/func_shell.c, + main/http.c, cdr/cdr_odbc.c, main/db.c, cdr/cdr_manager.c, + main/sched.c, main/pbx.c, funcs/func_timeout.c, + funcs/func_math.c, funcs/func_cut.c, main/chanvars.c, + main/netsock.c, funcs/func_curl.c, main/srv.c, main/privacy.c, + funcs/func_cdr.c, funcs/func_channel.c, main/audiohook.c, + funcs/func_iconv.c, main/alaw.c, main/asterisk.c, + funcs/func_base64.c, funcs/func_md5.c, funcs/func_sysinfo.c, + main/utils.c, funcs/func_sha1.c, cdr/cdr_pgsql.c, + funcs/func_logic.c, cdr/cdr_radius.c, main/enum.c, + funcs/func_uri.c, main/io.c, cdr/cdr_csv.c, main/ulaw.c, + main/channel.c, main/cdr.c, funcs/func_enum.c, main/dial.c, + funcs/func_groupcount.c, main/manager.c, main/tdd.c, + funcs/func_odbc.c, cdr/cdr_sqlite.c, main/logger.c, main/app.c, + main/image.c, main/dns.c, cdr/cdr_custom.c, funcs/func_version.c, + funcs/func_db.c, main/dnsmgr.c, main/translate.c, + main/slinfactory.c, funcs/func_lock.c, main/acl.c, main/rtp.c, + cdr/cdr_tds.c, funcs/func_realtime.c, main/hashtab.c, + funcs/func_blacklist.c, main/abstract_jb.c, main/cryptostub.c, + main/adsistub.c, main/file.c, main/callerid.c, main/astmm.c, + funcs/func_callerid.c, main/dsp.c: another bunch of include + removals (errno.h and asterisk/logger.h) + + * channels/chan_local.c, apps/app_record.c, + apps/app_alarmreceiver.c, apps/app_chanisavail.c, + apps/app_ices.c, apps/app_exec.c, channels/chan_iax2.c, + channels/chan_skinny.c, formats/format_pcm.c, + apps/app_dumpchan.c, apps/app_zapras.c, formats/format_h263.c, + codecs/codec_g722.c, formats/format_wav.c, apps/app_softhangup.c, + codecs/codec_g726.c, formats/format_ogg_vorbis.c, + apps/app_morsecode.c, apps/app_talkdetect.c, apps/app_db.c, + apps/app_speech_utils.c, apps/app_sendtext.c, + formats/format_g726.c, apps/app_mixmonitor.c, res/res_odbc.c, + apps/app_voicemail.c, channels/chan_vpb.cc, formats/format_sln.c, + res/res_snmp.c, apps/app_dictate.c, apps/app_authenticate.c, + apps/app_readexten.c, codecs/codec_gsm.c, apps/app_userevent.c, + channels/chan_gtalk.c, res/res_jabber.c, apps/app_setcallerid.c, + res/res_config_odbc.c, apps/app_osplookup.c, apps/app_mp3.c, + apps/app_minivm.c, res/res_realtime.c, formats/format_h264.c, + apps/app_directory.c, apps/app_rpt.c, channels/chan_mgcp.c, + apps/app_adsiprog.c, codecs/codec_lpc10.c, + res/res_config_pgsql.c, apps/app_read.c, channels/chan_sip.c, + codecs/codec_alaw.c, res/res_adsi.c, res/res_crypto.c, + channels/chan_jingle.c, apps/app_channelredirect.c, + apps/app_forkcdr.c, formats/format_vox.c, apps/app_sms.c, + formats/format_g723.c, apps/app_verbose.c, apps/app_stack.c, + apps/app_readfile.c, res/res_features.c, codecs/codec_adpcm.c, + apps/app_sayunixtime.c, apps/app_test.c, apps/app_image.c, + formats/format_wav_gsm.c, res/res_smdi.c, + include/asterisk/compat.h, apps/app_skel.c, apps/app_zapscan.c, + channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c, + formats/format_jpeg.c, formats/format_gsm.c, + apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c, + apps/app_privacy.c, codecs/codec_speex.c, apps/app_echo.c, + channels/chan_agent.c, apps/app_disa.c, + channels/iax2-provision.c, res/res_ael_share.c, + apps/app_transfer.c, res/res_monitor.c, apps/app_playback.c, + channels/chan_misdn.c, apps/app_waitforring.c, + apps/app_zapbarge.c, channels/chan_features.c, apps/app_macro.c, + apps/app_zapateller.c, res/res_indications.c, + codecs/codec_ilbc.c, apps/app_chanspy.c, channels/chan_unistim.c, + apps/app_meetme.c, res/res_musiconhold.c, apps/app_followme.c, + codecs/codec_zap.c, res/res_config_sqlite.c, + channels/misdn_config.c, apps/app_controlplayback.c, + formats/format_ilbc.c, channels/chan_phone.c, res/res_agi.c, + main/logger.c, apps/app_ivrdemo.c, apps/app_parkandannounce.c, + res/res_clioriginate.c, apps/app_while.c, include/asterisk.h, + apps/app_nbscat.c, channels/chan_zap.c, codecs/codec_a_mu.c, + res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c, + res/res_convert.c, apps/app_getcpeid.c, apps/app_system.c, + apps/app_queue.c, channels/chan_oss.c, channels/chan_usbradio.c, + apps/app_flash.c, apps/app_directed_pickup.c, + channels/chan_h323.c, codecs/codec_ulaw.c, channels/chan_nbs.c, + apps/app_senddtmf.c, formats/format_g729.c: include "logger.h" + and errno.h from asterisk.h - usage shows that they were included + almost everywhere. Remove some of the instances. + +2007-11-19 17:18 +0000 [r89422] Steve Murphy <murf@digium.com> + + * main/pbx.c: a correction to code involved in an extension removal + +2007-11-19 16:29 +0000 [r89421] Mark Michelson <mmichelson@digium.com> + + * funcs/func_sysinfo.c (added), CHANGES: Adding SYSINFO() dialplan + function for retrieval of system information + +2007-11-19 15:55 +0000 [r89417-89420] Joshua Colp <jcolp@digium.com> + + * /, res/res_features.c: Merged revisions 89419 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89419 | file | 2007-11-19 11:53:32 -0400 (Mon, 19 Nov 2007) | 6 + lines Print out the correct filename (features.conf) in the log + message when parkpos options are incorrect. (closes issue #11295) + Reported by: Laureano Patches: res_features.c.patch uploaded by + Laureano (license 265) ........ + + * /, doc/tex/localchannel.tex: Merged revisions 89416 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov + 2007) | 4 lines Clarify documentation a bit, include that a frame + has to pass through the core in order for the Local channel + optimization to happen. (closes issue #11246) Reported by: jon + ........ + +2007-11-19 14:36 +0000 [r89412] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/logger.h: revert inclusion of options.h + +2007-11-19 14:03 +0000 [r89410] Joshua Colp <jcolp@digium.com> + + * apps/app_playback.c: Change warning messages (which are really + debug messages) into debug messages. (closes issue #11288) + Reported by: IgorG Patches: saydebug-89394-1-trunk.patch uploaded + by IgorG (license 20) + +2007-11-19 09:16 +0000 [r89404-89407] Olle Johansson <oej@edvina.net> + + * CHANGES: Update CHANGES + + * channels/chan_sip.c: Adding busy-level to the SIP_PEER() dialplan + function. With this, you can control the peer in the dialplan, so + you avoid placing outbound calls when the device has reached + busy-level. Reported by pj. Closes bug #11180 + + * main/acl.c: Add some debugging to the routines that finds our + local IP address. Related to bug #9225 + + * channels/chan_sip.c: Make some notes about a problem I found with + the OPTIONs handler while working with the bug tracker. Since we + don't authenticate devices (peers/users) on OPTIONS we don't have + the proper context set for the user/peer. However, we might not + want to process an authentication for every OPTIONS, so we could + have a config option for this, "optionsforceok" to always answer + 200 OK on the request and not check device or destination, nor + add a SDP. If Asterisk sends the OPTIONs request, it doesn't care + about the reply. Some devices use OPTIONs to discover + capabilities, since we should answer like an INVITE from the + device and we need to support that properly too, which we don't + today. So much to do :-) + +2007-11-18 21:50 +0000 [r89394-89399] Joshua Colp <jcolp@digium.com> + + * build_tools/make_buildopts_h: Add OSX into the logic that uses + md5 instead of md5sum. + + * include/asterisk/compat.h: Use the easy way that rizzo mentioned, + only include malloc.h on the Windows platform. + + * include/asterisk/compat.h: Revert last commit, apparently + buildbot lied to me. + + * include/asterisk/compat.h: Change how we handle alloca to conform + with how it is suggested in the autoconf manual for + AC_FUNC_ALLOCA. FreeBSD 6 now builds again and no other platforms + should be broken by this. + + * configure, configure.ac: Change autoconf logic a bit so it says + what it is looking for in two instances where it didn't. + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/lock.h, include/asterisk/network.h: Use autoconf + logic to determine the presence of + PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP and + PTHREAD_MUTEX_RECURSIVE_NP. Enclose error message from network.h + in " + +2007-11-17 21:47 +0000 [r89393] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Add SS7 Generic address support (#11156) + +2007-11-17 19:29 +0000 [r89389-89392] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/compat.h: if alloca.h is not present, try + malloc.h + + * agi/Makefile: temporarily disable this target in mingw + + * Makefile: will i ever get precedences for windows right ? in the + meantime, use a variable to ease enabling/disabling print + subdirectories. + + * Makefile: reformulate dependencies in a more correct way + +2007-11-17 17:46 +0000 [r89388] Steve Murphy <murf@digium.com> + + * main/pbx.c, pbx/pbx_dundi.c: a quick fix to pbx_dundi.c to make + it so it will compile. Hope I did the right thing. And some + additions to removal of extens to take care of hashtab pointers + in all cases. + +2007-11-17 17:27 +0000 [r89363-89387] Luigi Rizzo <rizzo@icir.org> + + * Makefile.moddir_rules, Makefile.rules: as discussed some time ago + on the -dev list, create embedde object with a .eo suffix even if + they are coming from .cc sources. This simplifies the handling in + the build scripts. + + * include/asterisk/network.h: prefer socket.h over other variants + (winsock etc.) + + * channels/chan_local.c, main/translate.c, + channels/chan_features.c, main/http.c, main/config.c: trim more + redundant headers + + * main/acl.c: remove unnecessary includes + + * main/udptl.c, main/dnsmgr.c, channels/chan_sip.c, main/acl.c, + main/dns.c, main/rtp.c, main/netsock.c: fix breakage induced by + previous mistake + + * Makefile: wrong variable, wrong order -> broken build. + + * include/asterisk/acl.h, include/asterisk/utils.h, + include/asterisk/autoconfig.h.in, include/asterisk/rtp.h, + configure.ac, main/acl.c, include/asterisk/netsock.h, + main/utils.c, include/asterisk/manager.h, main/netsock.c, + main/manager.c, res/res_agi.c, pbx/pbx_dundi.c, + include/asterisk/udptl.h, include/asterisk/dnsmgr.h, + main/asterisk.c: start using asterisk/network.h for network + related headers. Also remove some unnecessary includes. + + * include/asterisk/network.h (added): wrapper for all generic + network headers that have different names and locations on the + various systems. + + * main/cygload.c: main is called main not amain! + + * main/Makefile: conditional targets for building the windows + version + + * Makefile: support cygwin targets + + * Makefile.moddir_rules: and this is the last one to have asterisk + compile (not run yet) natively under cygwin. + + * apps/app_sms.c: another cygwin compatibility fix. This one must + be handled in a better way in configure, also for other + architectures + + * utils/Makefile, main/Makefile, utils/extconf.c: more + cygwin/mingw32 compatibility fixes + + * include/asterisk/channel.h: use autoconf results to conditionally + compile timersub + + * include/asterisk/lock.h: compatibility fixes for cygwin + + * include/asterisk/compat.h: some version of flex produce code that + wants __STDC_VERSION__ defined, but the compiler does not always + define it. + + * Makefile: these linker flags apply to both cygwin and mingw32 + + * utils/hashtest2.c: add a return NULL to a function that is + expected to return a value so compilers that don't understand + that this code is NOTREACHED will not complain (the fault is not + much on the compiler but on the declaration of pthread_exit on + certain platforms) s/certain platform/cygwin/ if you are really + curious + + * main/loader.c: define RTLD_LOCAL for platforms that don't have + it. This is only to complete the build, clearly the linker + behaviour will be completely different and likely to cause + trouble in those cases. + + * channels/Makefile: filter out modules that do not compile under + windows (this should be handled with the dependencies generated + by configure and menuselect, but will be fixed later) + + * main/utils.c: netdb.h is used for gethostbyname, and it was not + included in some platforms. + + * main/cygload.c (added): Loader for cygwin where asterisk is + really a big dll (something like this is already in 1.2) + + * configure, include/asterisk/autoconfig.h.in, configure.ac: + timersub is a macro not a function, so write the check in a way + that detects both formats. + +2007-11-17 06:34 +0000 [r89359-89362] Russell Bryant <russell@digium.com> + + * pbx/pbx_lua.c: fix the build of pbx_lua + + * configure, include/asterisk/autoconfig.h.in, + include/asterisk/compat.h, configure.ac, include/asterisk/io.h, + include/asterisk/channel.h: Update the configure script check for + sys/poll.h to also provide the result in + include/asterisk/autoconfig.h. Also, move the conditional include + of sys/poll.h or asterisk/poll-compat.h into asterisk/config.h + instead of the two headers it existed in before. + + * build_tools/make_buildopts_h: actually let this compile, oops :( + + * build_tools/make_buildopts_h: Use the fix suggested by Tilghman + on the -dev to make cutting up the BUILDSUM friendly to non-bash + shells. I think this should work for BSD/mingw as well, but did + not yet remove the switch statement. + +2007-11-17 04:19 +0000 [r89348-89358] Luigi Rizzo <rizzo@icir.org> + + * Makefile: linker flags for mingw32 + + * configure, include/asterisk/autoconfig.h.in, configure.ac: add + detection for timersub() and winsock.h/winsock2.h + + * include/asterisk/endian.h: provide definitions for + __LITTLE_ENDIAN and __BIG_ENDIAN if not present. + + * main/Makefile, include/asterisk/io.h, include/asterisk/channel.h: + use poll as detected by configure + + * configure, configure.ac, makeopts.in: use autoconf to check for + the existence of sys/poll.h + + * build_tools/make_buildopts_h: this script is run on the build + system, not on the host. + + * Makefile.moddir_rules: compatibility fix for mingw32 + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + acinclude.m4, makeopts.in: acinclude.m4: add a function to help + checking sdl-config, gtk-config and the like (this could be used + for gtk and gtk2 as well) Other files: add tests for sdl, + sdl_image and avcodec and regenerate configure and + autoconfig.h.in + + * include/asterisk/autoconfig.h.in, configure.ac: add check for the + presence of glob + + * channels/chan_jingle.c, channels/chan_unistim.c, + funcs/func_enum.c, channels/chan_local.c, channels/chan_misdn.c, + channels/chan_skinny.c, funcs/func_odbc.c, channels/chan_h323.c, + utils/ael_main.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c, + apps/app_db.c, channels/chan_mgcp.c: more removal of duplicate + #include lines + + * main/udptl.c, funcs/func_module.c, res/res_features.c, + funcs/func_lock.c, res/res_adsi.c, funcs/func_strings.c, + channels/chan_agent.c, pbx/dundi-parser.c, main/rtp.c, + pbx/pbx_loopback.c, funcs/func_blacklist.c, + channels/chan_features.c, apps/app_dumpchan.c, res/res_agi.c, + main/logger.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, + apps/app_rpt.c, main/asterisk.c, apps/app_parkandannounce.c: + remove a bunch of duplicate includes Reproduce with grep -r + #include . | grep -v .svn | grep -v Binary | sort | uniq -c | + sort -nr + +2007-11-16 23:44 +0000 [r89347] Terry Wilson <twilson@digium.com> + + * res/res_features.c: Fix broken parking dial-back + +2007-11-16 23:33 +0000 [r89346] Steve Murphy <murf@digium.com> + + * main/pbx.c: My goodness, haven't handled an extension deletion. + Add code to ast_context_remove_extension2() to remove an + extension from the trie. Done by marking it deleted. The + scoreboard won't update for it any more. Also, a couple of calls + to insert hashtab had a spurious ->exten, which was removed. + +2007-11-16 23:28 +0000 [r89341-89345] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/paths.h, include/asterisk.h: paths are already + in include/asterisk/paths.h so don't duplicate them in + include/asterisk.h + + * include/asterisk/utils.h, include/asterisk/lock.h: whitespace + only change - adjust indentation and add some comments on the + content of these two files. utils.h (which is included in over + 150 files) contains a lot of unrelated functions which require + the inclusion of a large number of other headers. At some point + we should partition its content in a better way. + +2007-11-16 22:33 +0000 [r89340] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 89339 via svnmerge ........ r89339 | russell + | 2007-11-16 16:26:44 -0600 (Fri, 16 Nov 2007) | 5 lines + Temporarily revert revision 89325, which added md5 magic for + keeping track of what build options were used. We agreed that we + should remove this before making a 1.4 release, and then we can + put it back in. Then, we can take a month or so to play around + with it to get it how we want it. ........ + +2007-11-16 21:23 +0000 [r89333-89338] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/logger.h: logger.h does not need options.h + + * include/asterisk/utils.h, channels/chan_sip.c, + include/asterisk/astobj.h, include/asterisk/compat.h, + include/asterisk/channel.h, include/asterisk/strings.h, + utils/extconf.c, include/asterisk/frame.h, + include/asterisk/stringfields.h, include/asterisk/endian.h: + remove redundant #include "asterisk/compat.h", but make sure that + asterisk/compiler.h is included everywhere + + * main/acl.c, main/asterisk.c: remove duplicate headers. Properly + check for netdb.h (there is actually tens of places to fix) + + * Makefile.rules: put back default optimization to -O6 (previously + changed by mistake) + + * main/frame.c, main/threadstorage.c, apps/app_alarmreceiver.c, + apps/app_ices.c, channels/chan_iax2.c, apps/app_exec.c, + channels/chan_skinny.c, main/strcompat.c, pbx/pbx_ael.c, + apps/app_zapras.c, formats/format_h263.c, cdr/cdr_odbc.c, + include/asterisk/sha1.h, main/db.c, cdr/cdr_manager.c, + main/pbx.c, funcs/func_timeout.c, formats/format_wav.c, + apps/app_softhangup.c, codecs/codec_g726.c, funcs/func_cut.c, + apps/app_talkdetect.c, apps/app_db.c, funcs/func_channel.c, + main/privacy.c, funcs/func_iconv.c, pbx/pbx_config.c, + main/asterisk.c, res/res_odbc.c, include/asterisk/stringfields.h, + apps/app_voicemail.c, formats/format_sln.c, + apps/app_authenticate.c, apps/app_readexten.c, + apps/app_userevent.c, codecs/codec_gsm.c, Makefile.rules, + apps/app_setcallerid.c, include/asterisk/astmm.h, + res/res_config_odbc.c, apps/app_osplookup.c, funcs/func_odbc.c, + apps/app_mp3.c, formats/format_h264.c, apps/app_directory.c, + main/md5.c, res/res_config_pgsql.c, main/dnsmgr.c, + funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c, + res/res_crypto.c, include/asterisk/cli.h, channels/chan_jingle.c, + apps/app_forkcdr.c, funcs/func_blacklist.c, main/abstract_jb.c, + main/file.c, apps/app_sms.c, formats/format_g723.c, main/astmm.c, + apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c, + main/autoservice.c, funcs/func_module.c, codecs/codec_adpcm.c, + cdr/cdr_adaptive_odbc.c, main/devicestate.c, apps/app_image.c, + formats/format_wav_gsm.c, main/indications.c, pbx/pbx_loopback.c, + funcs/func_shell.c, include/asterisk/compat.h, apps/app_skel.c, + main/plc.c, channels/chan_alsa.c, apps/app_externalivr.c, + formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c, + main/sched.c, apps/app_dial.c, apps/app_page.c, apps/app_disa.c, + channels/iax2-provision.c, res/res_monitor.c, main/netsock.c, + apps/app_waitforring.c, main/fixedjitterbuf.c, + include/asterisk/lock.h, apps/app_chanspy.c, apps/app_cdr.c, + channels/chan_unistim.c, funcs/func_base64.c, funcs/func_md5.c, + apps/app_meetme.c, main/sha1.c, funcs/func_vmcount.c, + res/res_musiconhold.c, cdr/cdr_radius.c, apps/app_followme.c, + res/res_config_sqlite.c, main/fskmodem.c, + channels/misdn_config.c, apps/app_controlplayback.c, + cdr/cdr_csv.c, formats/format_ilbc.c, main/cdr.c, + channels/chan_phone.c, funcs/func_enum.c, main/dial.c, + main/manager.c, funcs/func_groupcount.c, cdr/cdr_sqlite.c, + main/logger.c, main/image.c, apps/app_ivrdemo.c, + res/res_clioriginate.c, apps/app_nbscat.c, codecs/codec_a_mu.c, + channels/chan_zap.c, main/slinfactory.c, res/res_convert.c, + pbx/pbx_lua.c, apps/app_queue.c, apps/app_system.c, + channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c, + channels/chan_usbradio.c, main/hashtab.c, apps/app_flash.c, + include/asterisk/strings.h, apps/app_senddtmf.c, + funcs/func_callerid.c, include/asterisk/time.h, + channels/chan_local.c, funcs/func_dialgroup.c, funcs/func_env.c, + apps/app_record.c, funcs/func_strings.c, apps/app_chanisavail.c, + pbx/pbx_spool.c, apps/app_dumpchan.c, formats/format_pcm.c, + main/http.c, main/stdtime/localtime.c, codecs/codec_g722.c, + apps/app_morsecode.c, formats/format_ogg_vorbis.c, + channels/iax2-parser.c, apps/app_speech_utils.c, + include/asterisk/logger.h, main/srv.c, apps/app_sendtext.c, + funcs/func_cdr.c, include/asterisk/md5.h, utils/hashtest2.c, + utils/ael_main.c, main/audiohook.c, apps/app_mixmonitor.c, + formats/format_g726.c, channels/chan_vpb.cc, apps/app_dictate.c, + channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c, + res/res_jabber.c, funcs/func_uri.c, main/io.c, + include/asterisk/abstract_jb.h, main/channel.c, + apps/app_minivm.c, res/res_realtime.c, main/dns.c, + apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, + codecs/codec_lpc10.c, apps/app_read.c, codecs/codec_alaw.c, + res/res_adsi.c, include/asterisk/plc.h, + apps/app_channelredirect.c, formats/format_vox.c, + main/cryptostub.c, main/callerid.c, pbx/pbx_dundi.c, + funcs/func_devstate.c, funcs/func_rand.c, apps/app_readfile.c, + cdr/cdr_sqlite3_custom.c, main/say.c, res/res_features.c, + apps/app_sayunixtime.c, apps/app_test.c, main/config.c, + main/loader.c, main/term.c, main/cli.c, res/res_smdi.c, + include/asterisk/astobj.h, apps/app_zapscan.c, apps/app_amd.c, + pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c, + include/asterisk/utils.h, apps/app_privacy.c, + codecs/codec_speex.c, apps/app_echo.c, channels/chan_agent.c, + funcs/func_math.c, res/res_ael_share.c, pbx/dundi-parser.c, + apps/app_transfer.c, include/asterisk/manager.h, + apps/app_playback.c, main/chanvars.c, apps/app_zapbarge.c, + channels/chan_misdn.c, funcs/func_curl.c, + channels/chan_features.c, apps/app_macro.c, codecs/codec_ilbc.c, + res/res_indications.c, apps/app_zapateller.c, main/dlfcn.c, + include/asterisk/slinfactory.h, utils/hashtest.c, main/utils.c, + funcs/func_sha1.c, codecs/codec_zap.c, main/enum.c, + include/asterisk/file.h, main/tdd.c, funcs/func_volume.c, + res/res_agi.c, main/app.c, apps/app_parkandannounce.c, + cdr/cdr_custom.c, apps/app_while.c, funcs/func_db.c, + res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c, + main/translate.c, include/asterisk/config.h, main/jitterbuf.c, + main/acl.c, apps/app_getcpeid.c, funcs/func_global.c, main/rtp.c, + funcs/func_extstate.c, apps/app_directed_pickup.c, + main/adsistub.c, channels/chan_h323.c, codecs/codec_ulaw.c, + main/event.c, channels/chan_nbs.c, pbx/pbx_gtkconsole.c, + formats/format_g729.c: Start untangling header inclusion in a way + that does not affect build times - tested, there is no + measureable difference before and after this commit. In this + change: use asterisk/compat.h to include a small set of system + headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, + stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the + inclusion is conditional on HAVE_FOO_H as determined by autoconf. + Normally, source files should not include any of the above system + headers, and instead use either "asterisk.h" or + "asterisk/compat.h" which does it better. For the time being I + have left alone second-level directories (main/db1-ast, etc.). + +2007-11-16 19:51 +0000 [r89331-89332] Mark Michelson <mmichelson@digium.com> + + * main/manager.c: Fixing a problem pointed out by Qwell + + * main/manager.c: Added some locks that should have been around + astman_send_error, at least according to the comments. (closes + issue #11258, reported and patched by eliel) + +2007-11-16 19:26 +0000 [r89329-89330] Steve Murphy <murf@digium.com> + + * main/pbx.c: This corrects a hashtab removal, given a bad argument + + * main/pbx.c, res/res_features.c: This fixes a problem with pattern + ranges; and corrects a situation in res_features, where an + extension would be created with the name Zap/51, as an example. + THe / is bad because it would tend to mean that the 51 is to be + cid matched. + +2007-11-16 18:48 +0000 [r89328] Luigi Rizzo <rizzo@icir.org> + + * build_tools/make_buildopts_h: both md5sum and variable + substitutions such as ${BUILDSUM:0:8} are not available in + FreeBSD. For the time being, put in a workaround so we can build + the system, and wait for the result of the discussion on whether + we can store the md5 as a string rather than 4 ints (if so, we + won't need more complex tricks with awk or sed for splitting the + md5). 1.4 will be fixed when we decide the issue. + +2007-11-16 17:11 +0000 [r89327] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Adding confirmation playback when + forwarding voicemail messages. This will attempt to play the + name(s) of the person(s) to whom you are forwarding the message + prior to prompting for prepending. If no name is found, the + extension is read back verbatim. (closes issue #9046, reported + and patched by jaroth) + +2007-11-16 16:56 +0000 [r89326] Kevin P. Fleming <kpfleming@digium.com> + + * /, include/asterisk/module.h, build_tools/make_buildopts_h, + main/loader.c: Merged revisions 89325 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89325 | kpfleming | 2007-11-16 10:47:46 -0600 (Fri, 16 Nov 2007) + | 4 lines To help combat problems where people build external + modules (asterisk-addons or others) and then change the build + options of the Asterisk build in a way that makes the + incompatible without warning, this commit introduces an MD5 + signature of the important build-time options and includes that + signature into modules when they are built. When the loader loads + one of these modules and notices the problem, it will emit a + warning to console and refuse to initialize the module, as doing + so could cause the system to be unstable or even crash. If you + upgrade to this version of Asterisk, you must rebuild *all* of + your modules that came from other sources before trying to run + this version. If you are using Digium's G.729 binary codec + module, you will need v33 or newer. ........ + +2007-11-16 15:44 +0000 [r89324] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 89323 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89323 | mmichelson | 2007-11-16 09:28:22 -0600 (Fri, 16 Nov + 2007) | 5 lines Make realtime queues accessible from the + QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and + patched by atis, with small modifications from me) ........ + +2007-11-16 10:07 +0000 [r89322] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/config.h, main/config.c: add a small new + function to retrieve variables from a config once we have a + pointer to the category. + +2007-11-16 10:06 +0000 [r89321] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c: fixed #10631, about one way audio. thanks + IgorG again. + +2007-11-16 09:51 +0000 [r89320] Luigi Rizzo <rizzo@icir.org> + + * channels/chan_oss.c: move the inner part of config file parsing + to a separate function, so it can be reused in the implementation + of cli commands when they have a similar syntax. + +2007-11-16 08:54 +0000 [r89319] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c: fixed compilation of chan_misdn, #11269, + thanks IgorG. + +2007-11-15 23:50 +0000 [r89299-89312] Tilghman Lesher <tlesher@digium.com> + + * main/utils.c, include/asterisk/stringfields.h: If we're going to + be passing a negative value for the size of a stringfield, in + order to indicate something, then using an UNSIGNED parameter is + bad, mmmmmkay? + + * Makefile, /: Merged revisions 89302 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89302 | tilghman | 2007-11-15 12:37:38 -0600 (Thu, 15 Nov 2007) + | 2 lines Start Asterisk in Debian at a more reasonable time + (since zaptel is at level 20) ........ + + * /, channels/misdn/isdn_lib.c: Merged revisions 89301 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15 + Nov 2007) | 2 lines Fix an uninitialized memory read found by + valgrind ........ + + * apps/app_zapscan.c: Fix trunk breakage due to chan->lock being + renamed. + + * /, channels/chan_iax2.c: Merged revisions 89298 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007) + | 5 lines Yet another memory corruption issue. Reported by: atis + Patch by: tilghman Fixes issue #10923 ........ + +2007-11-15 17:27 +0000 [r89297] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: Merged revisions 89296 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) | + 8 lines Update the SLAStation application to account for the case + where the SLA thread has a call out to the station, but the user + has pressed a line button to answer the call instead of picking + up the handset. If they do, the phone sends out a new INVITE. So, + the SLAStation app must check to see if it is picking up a + ringing trunk, and ensure that the other stations stop ringing. + (reported internally, patched by me, tested by mogorman) ........ + +2007-11-15 16:50 +0000 [r89294-89295] Steve Murphy <murf@digium.com> + + * main/pbx.c: Get rid of a previously missed ast_log call for + debug, no longer nec. + + * main/pbx.c: Perhaps I went overboard on initializing things. I + can remove unnecc. stuff later. A few bug fixes. Killing small + bugs on the way to killing bigger ones. Removed locking on + hashtabs; there's plenty of locks already being taken. A small + bug in the root_tree hashtab compare func. + +2007-11-15 16:20 +0000 [r89293] Luigi Rizzo <rizzo@icir.org> + + * main/channel.c, apps/app_channelredirect.c, main/manager.c, + res/res_features.c, apps/app_softhangup.c, + include/asterisk/channel.h, include/asterisk/lock.h, + apps/app_senddtmf.c: access channel locks through + ast_channel_lock/unlock/trylock and not through ast_mutex + primitives. To detect all occurrences, I have renamed the lock + field in struct ast_channel so it is clear that it shouldn't be + used directly. There are some uses in res/res_features.c (see + details of the diff) that are error prone as they try and lock + two channels without caring about the order (or without + explaining why it is safe). + +2007-11-15 15:39 +0000 [r89290-89291] Joshua Colp <jcolp@digium.com> + + * UPGRADE.txt: Fix typo in UPGRADE.txt. 'increase' should have been + used, not 'increasing'. + + * channels/chan_sip.c, channels/chan_h323.c, + channels/misdn_config.c: And file said... let trunk build again! + Accomplished by some more constification, and marking a function + in chan_sip as purposely unused until it is fixed up. + +2007-11-15 14:58 +0000 [r89287-89289] Mark Michelson <mmichelson@digium.com> + + * main/manager.c, /: Merged revisions 89288 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89288 | mmichelson | 2007-11-15 08:57:28 -0600 (Thu, 15 Nov + 2007) | 3 lines Undoing previous commit since I realize it was + wrong ........ + + * main/manager.c, /: Merged revisions 89286 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89286 | mmichelson | 2007-11-15 08:54:10 -0600 (Thu, 15 Nov + 2007) | 4 lines Adding a missing mutex unlock. (closes issue + 11256, reported and patched by ys) ........ + +2007-11-15 12:21 +0000 [r89278-89285] Olle Johansson <oej@edvina.net> + + * channels/chan_sip.c: Always relying on the responses when + crossing NAT's are not a good solution, it breaks communication. + Rizzo - you need to implement a configuration option for this + code. It's good, but maybe should be off by default. + + * /, channels/chan_sip.c: Merged revisions 89281 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6 + lines Don't send re-invites during pending INVITE transactions. + Patch by one47 - thanks! Closes issue #9305 ........ + + * /, channels/chan_sip.c: Merged revisions 89280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5 + lines Improve support for multipart messages. Code by gasparz, + changes by me (mostly formatting). Thanks, gasparz! Closes issue + #10947 ........ + + * channels/chan_sip.c: Exit early instead of deciding to exit after + processing the message. + + * channels/chan_sip.c, configs/sip.conf.sample: Add support for + application/dtmf SIP INFO dtmf handling. Yep, another way of + handling DTMF in SIP. Totally undocumented, but implemented in + enough devices so we have to support it. Code by sergee, small + changes by oej. Closes issue #11049 + +2007-11-15 01:42 +0000 [r89277] Steve Murphy <murf@digium.com> + + * main/pbx.c: Had trouble playing with parking; spent a long time + trying to reason out MATCHMORE mode. made these updates and xfers + on zaptel lines seem to work ok now + +2007-11-15 00:01 +0000 [r89273-89276] Tilghman Lesher <tlesher@digium.com> + + * /, main/app.c: Merged revisions 89275 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89275 | tilghman | 2007-11-14 17:23:58 -0600 (Wed, 14 Nov 2007) + | 5 lines When a recording ends with '#', we are improperly + trimming an extra 200ms from the recording. Reported by: sim + Patch by: tilghman Closes issue #11247 ........ + + * main/channel.c: Typo + + * main/channel.c: Add callerid to the Hangup manager event. + Reported by: outtolunc Patch by: outtolunc Closes issue #11248 + +2007-11-14 18:05 +0000 [r89271-89272] Steve Murphy <murf@digium.com> + + * main/pbx.c: Rescaled the weights of the patterns to give + something more independent of pattern length; and make . less + likely to win. Question: which should win for 14102241145-- + _1xxxxxxx. or _XXXXXXXXXXX -- right now, the pure X pattern will + win. + + * main/pbx.c: A further problem highlighted by 11233 has been + resolved; a certain combination of patterns in a certain order, + led to a malformed trie, due to a ptr not being initialized in + the loop. Also, some tree printing prettifications. + +2007-11-14 15:13 +0000 [r89269-89270] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_phone.c, channels/chan_zap.c, res/res_jabber.c, + res/res_config_sqlite.c, main/config.c, res/res_odbc.c: One more + typo in config.c; and missed conversions due to the constifying + of ast_variable_new parameters + + * main/config.c: Typo + +2007-11-14 13:18 +0000 [r89268] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/acl.h, channels/chan_sip.c, + include/asterisk/config.h, channels/chan_agent.c, res/res_adsi.c, + main/acl.c, pbx/dundi-parser.c, apps/app_queue.c, + channels/chan_iax2.c, main/enum.c, channels/chan_oss.c, + apps/app_playback.c, main/config.c, pbx/dundi-parser.h, + include/asterisk/abstract_jb.h, main/manager.c, + channels/chan_skinny.c, apps/app_minivm.c, main/abstract_jb.c, + main/logger.c, pbx/pbx_dundi.c, apps/app_directory.c, + apps/app_voicemail.c: make the 'name' and 'value' fields in + ast_variable const char * This prevents modifying the strings in + the stored variables, and catched a few instances where this was + actually done. Given the differences between trunk and 1.4 (and + the fact that this is effectively an API change) it is better to + fix 1.4 independently. These are chan_sip.c::sip_register() + chan_skinny.c:: near line 2847 config.c:: near line 1774 + logger.c::make_components() res_adsi.c:: near line 1049 I may + have missed some instances for modules that do not build here. + +2007-11-14 03:22 +0000 [r89263-89266] Russell Bryant <russell@digium.com> + + * main/hashtab.c, include/asterisk/hashtab.h: Fix up various coding + guidelines issues ... - handle memory allocation failures - add + an ast_ prefix to a publicly exported function - put curly braces + in the right places - add a bunch of spaces where they should be + be used + + * res/res_clioriginate.c: - Use the ARRAY_LEN macro in a couple + places - return errors from load_module / unload_module + + * apps/app_dial.c: Use BEGIN_OPTIONS / END_OPTIONS to make the + syntax highlighting in my editor happy + + * apps/app_queue.c: Instead of reserving 800 bytes for periodic + announcements, use an array of ast_str pointers and only alloate + space for the strings as needed. + +2007-11-14 01:16 +0000 [r89262] Joshua Colp <jcolp@digium.com> + + * main/srv.c, /: Merged revisions 89260 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89260 | file | 2007-11-13 21:15:12 -0400 (Tue, 13 Nov 2007) | 4 + lines Return the proper value when the srv_callback function + executes properly. (closes issue #11240) Reported by: jtodd + ........ + +2007-11-14 01:15 +0000 [r89261] Russell Bryant <russell@digium.com> + + * apps/app_queue.c: Convert most of the strings in the call_queue + struct to use stringfields. + +2007-11-14 00:54 +0000 [r89259] Kevin P. Fleming <kpfleming@digium.com> + + * main/channel.c, main/pbx.c: use simpler technique for removing + known entries from lists + +2007-11-14 00:33 +0000 [r89258] Russell Bryant <russell@digium.com> + + * main/image.c: - Simplify removing an item from a list - move a + verbose message to after the item is added to the list - make use + of the ARRAY_LEN macro in one spot + +2007-11-13 23:43 +0000 [r89256-89257] Steve Murphy <murf@digium.com> + + * main/pbx.c: This hopefully will fix the re-opened 11233. Hadn't + covered the case of a context with no patterns. (blush) + + * main/pbx.c: closes issue #11233 -- where some fine points in the + algorithm to build the tree needed to be corrected. Many thanks + for the test case, jtodd + +2007-11-13 21:08 +0000 [r89255] Jason Parker <jparker@digium.com> + + * /: Blocked revisions 89254 via svnmerge (closes issue #11238) + ........ r89254 | qwell | 2007-11-13 15:07:08 -0600 (Tue, 13 Nov + 2007) | 4 lines Fix building on newer systems which require a + third arg to open() when using O_CREAT. Issue 11238, reported by + puzzled. ........ + +2007-11-13 21:01 +0000 [r89250-89253] Russell Bryant <russell@digium.com> + + * include/asterisk/lock.h: This fixes a build error on my mac. It + also works on my linux box. Let me know if it breaks any other + platform ... + + * res/res_features.c: Fix a typo pointed out by outtolunc, thanks + :) + + * channels/chan_sip.c: - Convert initialization of a struct to C99 + style instead of GNU style - Fix a minor spelling error in a + comment + + * res/res_features.c, CHANGES: Update the ParkedCall application to + grab the first available parked call if no parked extension is + provided as an argument. (closes issue #10803) Reported by: + outtolunc Patches: res_features-parkedcall-any.diff4 uploaded by + outtolunc (license 237) - modified by me to work a bit + differently ... + +2007-11-13 19:48 +0000 [r89249] Jason Parker <jparker@digium.com> + + * /, res/res_features.c: Merged revisions 89248 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11237) ........ r89248 | qwell | 2007-11-13 13:47:45 -0600 + (Tue, 13 Nov 2007) | 7 lines Revert change from revision 67064. + It is documented behavior that if a parking extension already + exists while using PARKINGEXTEN, dialplan execution will + continue. If blind transferring to a Park with PARKINGEXTEN, you + must keep this in mind, and handle the failure yourself. Issue + 11237, reported by jon. ........ + +2007-11-13 17:41 +0000 [r89247] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 89246 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007) + | 2 lines If we set a value for qualify, we should actually pay + attention to it, instead of overriding the value ........ + +2007-11-13 16:03 +0000 [r89242] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_mixmonitor.c: Merged revisions 89241 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13 + Nov 2007) | 5 lines Reverting commit made in revision 89205 since + it is unnecessary. Thanks to Kevin for pointing this out ........ + +2007-11-13 14:03 +0000 [r89240] Tilghman Lesher <tlesher@digium.com> + + * /, main/utils.c: Merged revisions 89239 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89239 | tilghman | 2007-11-13 07:51:53 -0600 (Tue, 13 Nov 2007) + | 4 lines Debugging is running into the 16-lock limit. Increase + to avoid. (This define is only effective when debugging is turned + on, so there's no effect for most installations.) ........ + +2007-11-13 01:19 +0000 [r89206-89207] Mark Michelson <mmichelson@digium.com> + + * apps/app_mixmonitor.c: There is the potential to copy + uninitialized memory into the mixmonitor->post_process string. + This fix prevents that. + + * /, apps/app_mixmonitor.c: Merged revisions 89205 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12 + Nov 2007) | 5 lines Some sanity checking for MixMonitor. If only + 1 argument is given, then the args.options and args.post_process + strings are uninitialized and could contain garbage. This change + handles this situation properly by only using arguments that we + have parsed. ........ + +2007-11-13 00:19 +0000 [r89202-89203] Jason Parker <jparker@digium.com> + + * Makefile: oops, somebody left out the directory here... + + * channels/chan_unistim.c, res/res_features.c, main/ast_expr2f.c, + include/asterisk/config.h, res/res_convert.c, res/res_crypto.c, + pbx/pbx_lua.c, include/asterisk/cli.h, include/asterisk/pbx.h, + res/res_config_sqlite.c, res/res_monitor.c, + include/asterisk/stringfields.h, res/res_clioriginate.c: Doxygen + fixes. Also fix a common typo I kept seeing (arguement) in + various files. Closes issue #11222, patch by snuffy (with + arguement > argument by me). + +2007-11-12 23:33 +0000 [r89196-89201] Steve Murphy <murf@digium.com> + + * utils/hashtest.c: Don't forget the ASTERISK_VERSION for the sake + of the mtx_prof stuff. + + * include/asterisk/hashtab.h: Thanks to snuffy for this doxygen + update to hashtab.h; closes issue #11223 + + * main/hashtab.c, include/asterisk/hashtab.h: Thanks to snuff-work, + who brought up that these fixes might need to be made. + +2007-11-12 20:48 +0000 [r89195] Jason Parker <jparker@digium.com> + + * main/pbx.c, /: Merged revisions 89194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89194 | qwell | 2007-11-12 14:46:52 -0600 (Mon, 12 Nov 2007) | 1 + line Fix a typo pointed out by De_Mon on #asterisk-dev ........ + +2007-11-12 20:29 +0000 [r89193] Tilghman Lesher <tlesher@digium.com> + + * /: Blocked revisions 89191 via svnmerge ........ r89191 | + tilghman | 2007-11-12 14:16:18 -0600 (Mon, 12 Nov 2007) | 5 lines + If two config writes collide, file corruption could result. Use a + mkstemp() file, instead. Reported by: paravoid Patch by: tilghman + Closes issue #10781 ........ + +2007-11-12 20:16 +0000 [r89190] Kevin P. Fleming <kpfleming@digium.com> + + * utils/Makefile, utils/hashtest.c: (closes issue #11221) Reported + by: eliel Patches: utils.Makefile.patch uploaded by eliel + (modified by me) (license 64) + +2007-11-12 18:44 +0000 [r89186] Steve Murphy <murf@digium.com> + + * main/pbx.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, + funcs/func_logic.c, apps/app_exec.c, apps/app_queue.c, + apps/app_mixmonitor.c, cdr/cdr_manager.c: Based on a note in + asterisk-dev by Brian Capouch, I determined I too agressive in + not initializing arrays passed to pbx_substitute_variables_xxxx; + I reviewed the code (again) and hopefully found every possible + spot where substitute_variables is called conditionally, and made + sure the char array involved was set to a null string. + +2007-11-12 17:44 +0000 [r89185] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /, channels/chan_sip.c: Merged revisions 89184 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007) + | 5 lines Fix two cases of memory corruption caused by background + threads. Reported by: atis Patch by: tilghman Fixes issue #10923 + ........ + +2007-11-12 13:36 +0000 [r89178-89179] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged + revisions 89173 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | + 1 line if we're NT and no number was dialed and overlapdial is + set, we wait for the ISDN timeout instead of starting our own + timer. added a comment for the misdn.conf.sample for the + overlapdial config option. ........ + + * channels/misdn/isdn_lib_intern.h, channels/chan_misdn.c, /, + channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: + Merged revisions 89172 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) | + 1 line added restart all interfaces Restart_Indicator, to + automatically send a RESTART after the L2 of a PTP Port comes up. + Also fixed some places where we have send a RELEASE without need + for it. ........ + +2007-11-12 13:26 +0000 [r89177] Joshua Colp <jcolp@digium.com> + + * channels/chan_unistim.c, utils/hashtest.c: Fix building on + FreeBSD by including/not including some headers. (closes issue + #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys + (license 281) + +2007-11-12 13:22 +0000 [r89174-89176] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged + revisions 89171 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) | + 1 line fixed a state/event issue with overlapdial=yes when no + extension matched. removed the general sending of a + RELEASE_COMPLETE when we receive a RELEASE, this is done by + mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with + mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to + upgrade to at least mISDNuser-1.1.6 (when using the NT mode at + all) ........ + + * /, channels/misdn/isdn_lib.c: Merged revisions 89170 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12 + Nov 2007) | 1 line fixed the support for CW and therefore for the + reject_cause option. ........ + + * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, + channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample, + channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged + revisions 89169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | + 1 line aded ntkeepcalls option, to avoid droÃpping calls when the + L2 goes down on a PTP link. There are some pbx which do turn off + the L1 for a very short while and restart it immediately. + normally T310 should be started and after 10 seconds or so the + calls should be dropped, this is a simple fix wihtout this timer. + ........ + +2007-11-09 18:57 +0000 [r89130-89132] Jason Parker <jparker@digium.com> + + * configs/usbradio.conf.sample (added): Add usbradio.conf.sample + from branches/1.4/configs - r84162. It was mistakenly deleted in + 1.4 without ever being merged to trunk. Reported by eliel on + #asterisk-dev. + + * cdr/cdr_sqlite3_custom.c, configs/cdr_sqlite3_custom.conf + (removed), configs/cdr_sqlite3_custom.conf.sample (added): Fix a + few potential deadlocks in cdr_sqlite3_custom. (also rename + sample config to .sample) Closes issue #11208, patch by Laureano. + +2007-11-09 16:00 +0000 [r89129] Steve Murphy <murf@digium.com> + + * res/ael/pval.c, utils/Makefile, main/pbx.c, main/hashtab.c + (added), main/Makefile, utils/hashtest.c (added), pbx/pbx_ael.c, + include/asterisk/hashtab.h (added), main/config.c: This is the + perhaps the biggest, boldest, most daring change I've ever + committed to trunk. Forgive me in advance any disruption this may + cause, and please, report any problems via the bugtracker. The + upside is that this can speed up large dialplans by 20 times (or + more). Context, extension, and priority matching are all fairly + constant-time searches. I introduce here my hashtables + (hashtabs), and a regression for them. I would have used the + ast_obj2 tables, but mine are resizeable, and don't need the + object destruction capability. The hashtab stuff is well tested + and stable. I introduce a data structure, a trie, for extension + pattern matching, in which knowledge of all patterns is + accumulated, and all matches can be found via a single traversal + of the tree. This is per-context. The trie is formed on the first + lookup attempt, and stored in the context for future lookups. + Destruction routines are in place for hashtabs and the pattern + match trie. You can see the contents of the pattern match trie by + using the 'dialplan show' cli command when 'core set debug' has + been done to put it in debug mode. The pattern tree traversal + only traverses those parts of the tree that are interesting. It + uses a scoreboard sort of approach to find the best match. The + speed of the traversal is more a function of the length of the + pattern than the number of patterns in the tree. The tree also + contains the CID matching patterns. See the source code comments + for details on how everything works. I believe the approach + general enough that any issues that might come up involving fine + points in the pattern matching algorithm, can be solved by just + tweaking things. We shall see. The current pattern matcher is + fairly involved, and replicating every nuance of it is difficult. + If you find and report problems, I will try to resolve than as + quickly as I can. The trie and hashtabs are added to the existing + context and exten structs, and none of the old machinery has been + removed for the sake of the multitude of functions that use them. + In the future, we can (maybe) weed out the linked lists and save + some space. + +2007-11-08 23:53 +0000 [r89124-89126] Jason Parker <jparker@digium.com> + + * /, main/say.c: Merged revisions 89125 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11203) ........ r89125 | qwell | 2007-11-08 17:52:35 -0600 + (Thu, 08 Nov 2007) | 4 lines Properly say the seconds here.. + Issue 11203, fix described by vma. ........ + + * pbx/pbx_lua.c: Add check_hangup() method to pbx_lua, which can be + used to check whether it is time to hangup a channel. Closes + issue #11202, patch by mnicholson + +2007-11-08 22:33 +0000 [r89122-89123] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: app_voicemail failed to build when + compiling with IMAP_STORAGE Now it does not. + + * main/threadstorage.c: AST_LIST_REMOVE_CURRENT takes only one + argument. Thanks to snuffy for pointing this out on IRC + +2007-11-08 21:27 +0000 [r89121] Joshua Colp <jcolp@digium.com> + + * funcs/func_env.c: Make func_env build again. + +2007-11-08 21:01 +0000 [r89120] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 89119 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov + 2007) | 7 lines Rework of the commit I made yesterday to use the + already built-in ast_uri_decode function as opposed to my + home-rolled one. Also added comments. Thanks to oej for pointing + me in the right direction ........ + +2007-11-08 20:39 +0000 [r89118] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_features.c: convert this code to a more efficient + idiom + +2007-11-08 18:49 +0000 [r89116-89117] Jason Parker <jparker@digium.com> + + * res/res_smdi.c: Change a warning to a notice. Issue #11195, patch + by eliel + + * /, configs/cdr_adaptive_odbc.conf.sample, + configs/res_odbc.conf.sample: Merged revisions 89115 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11195) ........ r89115 | qwell | 2007-11-08 12:45:15 -0600 + (Thu, 08 Nov 2007) | 4 lines Avoid warnings on load when using + sample configuration files. Issue 11195, patch by eliel. ........ + +2007-11-08 17:32 +0000 [r89113-89114] Tilghman Lesher <tlesher@digium.com> + + * apps/app_readfile.c, funcs/func_env.c: Add the FILE() dialplan + function and deprecate ReadFile. + + * channels/chan_features.c: Fix missed conversion to linkedlists + macro change + +2007-11-08 16:51 +0000 [r89112] Mark Michelson <mmichelson@digium.com> + + * /: Blocking changes from previous 1.4 commit + +2007-11-08 09:21 +0000 [r89108-89110] Luigi Rizzo <rizzo@icir.org> + + * apps/app_voicemail.c: use %f instead of %lf (the 'l' is ignored + anyways). + + * main/audiohook.c: use %d and cast to int instead of %zd for + size_t object, this helps portability. + + * channels/chan_unistim.c: initialize a variable to silence + compiler. The type of warnings emitted depends on the + optimization level, at the lower levels the compiler doesn't + always understand what the programmer has in mind. In this case I + could not understand it either. + +2007-11-08 05:36 +0000 [r89106-89107] Kevin P. Fleming <kpfleming@digium.com> + + * main/srv.c, /: Merged revisions 89105 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89105 | kpfleming | 2007-11-08 00:26:47 -0500 (Thu, 08 Nov 2007) + | 2 lines fix a glaring bug in the new SRV record handling that + would cause incorrect weight sorting ........ + + * main/autoservice.c, main/frame.c, apps/app_meetme.c, + res/res_features.c, funcs/func_strings.c, main/devicestate.c, + res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c, + codecs/codec_zap.c, res/res_jabber.c, main/indications.c, + main/astobj2.c, main/config.c, main/loader.c, main/cli.c, + main/cdr.c, main/channel.c, main/manager.c, res/res_agi.c, + main/logger.c, main/app.c, main/image.c, res/res_speech.c, + main/sched.c, main/pbx.c, main/translate.c, res/res_crypto.c, + channels/chan_agent.c, utils/astman.c, apps/app_queue.c, + channels/iax2-parser.c, main/srv.c, + include/asterisk/linkedlists.h, main/file.c, pbx/pbx_dundi.c, + main/event.c, main/audiohook.c, res/res_odbc.c, main/asterisk.c, + apps/app_voicemail.c: improve linked-list macros in two ways: - + the *_CURRENT macros no longer need the list head pointer + argument - add AST_LIST_MOVE_CURRENT to encapsulate the + remove/add operation when moving entries between lists + +2007-11-08 05:00 +0000 [r89104] Tilghman Lesher <tlesher@digium.com> + + * /, doc/valgrind.txt: Merged revisions 89103 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007) + | 2 lines Typo ........ + +2007-11-08 02:28 +0000 [r89096-89102] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 89101 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 + lines Do not add a sip: to the beginning of the To URI unless + needed. (closes issue #10756) Reported by: goestelecom ........ + + * /, channels/chan_sip.c: Merged revisions 89099 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 + lines Improve the devicestate logic for multiple devices. If any + are available then the extension is considered available. (closes + issue #10164) Reported by: nic_bellamy Patches: + sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299) + ........ + + * /, channels/chan_sip.c: Merged revisions 89097 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8 + lines Add support for allowing one outgoing transaction. This + means if a response comes back out of order chan_sip will still + handle it. I dream of a chan_sip with real transaction support. + (closes issue #10946) Reported by: flefoll (closes issue #10915) + Reported by: ramonpeek (closes issue #9567) Reported by: + atca_pres ........ + + * /, channels/chan_sip.c: Merged revisions 89095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4 + lines If callerid is configured in sip.conf use that for checking + the presence of an extension in the dialplan. (closes issue + #11185) Reported by: spditner ........ + +2007-11-07 23:47 +0000 [r89094] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_queue.c: Merged revisions 89093 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007) + | 7 lines The member refcount must be incremented, to avoid using + it after deallocation. A huge thanks go to lvl- for patiently + providing the necessary valgrind output that was necessary to + finding this problem of memory corruption. Reported by: lvl- + Patch by: tilghman Closes issue #11174 ........ + +2007-11-07 23:18 +0000 [r89091-89092] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: If imapfolder has been specified in + voicemail.conf, we should not connect to INBOX... ever. It may + not exist. (closes issue #11151, reported by selsky, patched by + me) + + * /, channels/chan_sip.c: Merged revisions 89090 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov + 2007) | 6 lines This patch makes it possible for SIP phones to + dial extensions defined with '#' characters in extensions.conf + AND maintain their escaped characters when forming URI's (closes + issue #10681, reported by cahen, patched by me, code review by + file) ........ + +2007-11-07 22:09 +0000 [r89089] Steve Murphy <murf@digium.com> + + * /, res/res_jabber.c, cdr/cdr_tds.c: Merged revisions 89088 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1 + line In response to 10578, I just ran 1.4 thru valgrind; some of + the config leakage I've already fixed, but it doesn't hurt to + double check. I found and fixed leaks in res_jabber, cdr_tds, + pbx_ael. Nothing major, tho. ........ + +2007-11-07 17:45 +0000 [r89086] Joshua Colp <jcolp@digium.com> + + * channels/h323/ast_h323.cxx: Minor change so chan_h323 builds + again. + +2007-11-07 13:12 +0000 [r89082-89084] Luigi Rizzo <rizzo@icir.org> + + * Makefile: remove enter/exit comments when handling subdirectory. + If we really want them we can remove the --no-print-directory + + * main/loader.c: remove a debugging message which i forgot in. + + * Makefile: match changes in menuselect's Makefile + +2007-11-07 04:21 +0000 [r89077-89081] Tilghman Lesher <tlesher@digium.com> + + * apps/app_playback.c: Suppress erroneous warnings on load. + Reported by: eliel Patch by: eliel Closes issue #11177 + + * /, configs/extensions.ael.sample: Merged revisions 89079 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89079 | tilghman | 2007-11-06 22:07:49 -0600 (Tue, 06 Nov 2007) + | 5 lines Suppress AEL warnings on load. Reported by: eliel Patch + by: eliel Closes issue #11178 ........ + + * channels/chan_zap.c, configs/zapata.conf.sample: Provide the + ability to directly manipulate the TON/NPI bits in the + dialstring. Reported by: thetatag Patch by: + thetatag/stevens/tilghman Closes issue #5331 + + * contrib/utils/eagi_proxy.c (added): Add contributed EAGI proxy, + which provides FastAGI functionality for EAGI, while also + buffering the audio stream. Reported by: devil_slayer Patch by: + devil_slayer Closes issue #8921 + +2007-11-07 00:16 +0000 [r89076] Russell Bryant <russell@digium.com> + + * main/astmm.c: Fix another CLI command so it doesn't run the real + code when called for initialization. + +2007-11-07 00:04 +0000 [r89075] Mark Michelson <mmichelson@digium.com> + + * doc/tex/imapstorage.tex: Adding documentation regarding + imapfolder, imapgreetings, and greetingsfolder options in + voicemail.conf (closes issue #11133, reported by selsky, patched + by blitzrage) + +2007-11-07 00:00 +0000 [r89073-89074] Russell Bryant <russell@digium.com> + + * include/asterisk/agi.h, res/res_agi.c, CHANGES: Print out the + channel name as a prefix to the "agi debug" output. This makes + AGI debugging on busy systems much easier. (closes issue #10730) + Reported by: junky Patches: agi_debug_chan.diff uploaded by junky + (license 177) 20070923_10730.diff uploaded by mvanbaak (license + 7) + + * apps/app_meetme.c, CHANGES: Added the ability to do "meetme + concise" with the "meetme" CLI command. This extends the concise + capabilities of this CLI command to include listing all + conferences, instead of an addition to the other sub commands for + the "meetme" command. (closes issue #11078) Reported by: jthomas + Patches: meetme-concise.patch uploaded by jthomas (license 293) + +2007-11-06 23:08 +0000 [r89072] Joshua Colp <jcolp@digium.com> + + * main/pbx.c: Fix up some PBX logic that became broken. The code + would exit prematurely when it should have been collecting more + digits. (closes issue #11175) Reported by: pj + +2007-11-06 22:51 +0000 [r89071] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_jingle.c, channels/chan_phone.c, + codecs/codec_g722.c, main/frame.c, channels/chan_sip.c, + channels/chan_skinny.c, main/translate.c, channels/chan_h323.c, + main/file.c, channels/chan_gtalk.c, include/asterisk/frame.h, + main/rtp.c, channels/chan_mgcp.c, include/asterisk/translate.h: + Commit some cleanups to the format type code. - Remove the + AST_FORMAT_MAX_* types, as these are consuming 3 out of our + available 32 bits. - Add a native slin16 type, so that 16kHz + codecs can translate without losing resolution. (This doesn't + affect anything immediately, until another codec has wb support.) + +2007-11-06 22:36 +0000 [r89070] Mark Michelson <mmichelson@digium.com> + + * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the + queue strategy wrandom (closes issue #10942, reported and patched + by julianjm, documentation changes by me) + +2007-11-06 22:15 +0000 [r89069] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c, doc/tex/channelvariables.tex, CHANGES: Added + the S() and L() options to the MeetMe application. These are + pretty much identical to the S() and L() options to Dial(). They + let you set timeouts for the conference, as well as have warning + sounds played to let the caller know how much time is left, and + when it is running out. (closes issue #8030) Reported by: areski + Patches: meetme_timeout_timelimit_v2.patch uploaded by areski + (license 29) + +2007-11-06 22:05 +0000 [r89068] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Added CLI and manager commands for changing a + queue member's penalty (closes issue #9374, reported and + initially patched by wuwu, intermediate patch by eliel, and final + patch by me) + +2007-11-06 22:01 +0000 [r89067] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Add some more locking as well as API update + for libss7 for new transport types + +2007-11-06 21:08 +0000 [r89062] Steve Murphy <murf@digium.com> + + * /, main/config.c: Merged revisions 89036 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1 + line closes issue #8786 - where the [catname](!) and + [catname](othercat1,othercat2,...) notation gets dropped across a + ConfigUpdate (or any other thing that would cause a config file + to be written). While I was at it, I also cleaned up some of the + destroy routines to free up comments, which was not being done. + Made sure the new struct I introduced is also cleaned up properly + at destruction time. My code handles multiple template + inclusions. Many thanks to ssokol for his patch, which, while not + literally used in the final merge, served as a foundation for the + fix. ........ + +2007-11-06 20:55 +0000 [r89057] Joshua Colp <jcolp@digium.com> + + * main/channel.c: Remove native bridging check for DTMF based + transfers. Thanks to the last batch of RTP changes it is no + longer required for the media stream to go through Asterisk if + DTMF is going over signalling. It will simply reinvite back as + needed. (closes issue #11172) Reported by: ibc + +2007-11-06 20:32 +0000 [r89055] Mark Michelson <mmichelson@digium.com> + + * res/res_features.c: Instead of trying to callback a local channel + on a failed attended transfer, call the device that made the + transfer instead. This makes for much smoother calling back when + queues are involved. (closes issue #11155, reported by IPetrov) + Tremendous thanks to Russell for pulling me out of my block I was + having on this one + +2007-11-06 20:22 +0000 [r89052-89054] Russell Bryant <russell@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 89053 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06 + Nov 2007) | 3 lines Fix init_classes() so that classes that + actually do have files loaded aren't treated as empty, and + immediately destroyed ... ........ + + * main/astmm.c: Fix the memory show allocations CLI command so that + it doesn't spew out all of the current memory allocations when + you start Asterisk, when the command's handler gets called for + initialization. + +2007-11-06 19:40 +0000 [r89051] Steve Murphy <murf@digium.com> + + * main/ast_expr2f.c, main/ast_expr2.fl: Hoping to avoid a crash in + OSX for a problem blitzrage found + +2007-11-06 19:23 +0000 [r89050] Olle Johansson <oej@edvina.net> + + * main/fskmodem.c: Formatting. Illegaly using some spare spaces + from Russell's space-bucket. + +2007-11-06 19:16 +0000 [r89049] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 89045 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06 + Nov 2007) | 2 lines We went to the trouble of creating a method + of tracking failed trylocks, then never turned it on (oops). + ........ + +2007-11-06 19:10 +0000 [r89048] Olle Johansson <oej@edvina.net> + + * main/tdd.c, include/asterisk/tdd.h: Additional TDD changes + (preparing for SIP changes - adding TDD support to SIP) + +2007-11-06 19:10 +0000 [r89047] Jason Parker <jparker@digium.com> + + * /, codecs/codec_zap.c: Merged revisions 89046 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4 + lines Correctly set the total number of channels from a zaptel + transcoder board. SPD-49, patch by Matthew Nicholson. ........ + +2007-11-06 19:04 +0000 [r89044] Mark Michelson <mmichelson@digium.com> + + * apps/app_readfile.c, res/res_features.c, apps/app_sayunixtime.c, + apps/app_test.c, apps/app_chanisavail.c, res/res_musiconhold.c, + apps/app_exec.c, apps/app_followme.c, apps/app_minivm.c, + apps/app_mp3.c, apps/app_amd.c, apps/app_while.c, main/pbx.c, + apps/app_nbscat.c, channels/chan_sip.c, apps/app_festival.c, + apps/app_softhangup.c, apps/app_waitforsilence.c, + channels/chan_agent.c, apps/app_morsecode.c, apps/app_getcpeid.c, + apps/app_playback.c, res/res_monitor.c, apps/app_speech_utils.c, + apps/app_forkcdr.c, apps/app_waitforring.c, + apps/app_directed_pickup.c, apps/app_macro.c, apps/app_sms.c, + res/res_indications.c, apps/app_chanspy.c, apps/app_mixmonitor.c, + apps/app_stack.c: "show application <foo>" changes for clarity. + (closes issue #11171, reported and patched by blitzrage) Many + thanks! + +2007-11-06 19:04 +0000 [r89043] Olle Johansson <oej@edvina.net> + + * /, main/tdd.c: Merged revisions 89042 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2 + lines Bug fixes to tdd support in zaptel. ........ (Small changes + for trunk) + +2007-11-06 18:44 +0000 [r89041] Jason Parker <jparker@digium.com> + + * channels/chan_jingle.c, include/asterisk/jabber.h, + channels/chan_gtalk.c, res/res_jabber.c: Allow gtalk and jingle + to use TLS connections again. Closes issue #9972 + +2007-11-06 18:23 +0000 [r89038] Russell Bryant <russell@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 89037 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06 + Nov 2007) | 11 lines If someone were to delete the files used by + an existing MOH class, and then issue a reload, further use of + that class could result in a crash due to dividing by zero. This + set of changes fixes up some places to prevent this from + happening. (closes issue #10948) Reported by: jcomellas Patches: + res_musiconhold_division_by_zero.patch uploaded by jcomellas + (license 282) Additional changes added by me. ........ + +2007-11-06 17:10 +0000 [r89034] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 89032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 + lines Make it so that if a peer is determined to be unreachable + using qualify their devicestate will report back unavailable. + (closes issue #11006) Reported by: pj ........ + +2007-11-06 17:05 +0000 [r89031] Luigi Rizzo <rizzo@icir.org> + + * main/loader.c: Fix embedding of modules on FreeBSD: the + constructor for the list of modules was run after the + constructors for the embedded modules (which appended entries to + the list). As a result, the list appeared empty when it was time + to use it. On linux the order of execution of constructor was + evidently different (it may depend on the ordering of modules in + the ELF file). This is only a workaround - there may be other + situations where the execution of constructors causes problems, + so if we manage to find a more general solution this workaround + can go away. + +2007-11-06 16:29 +0000 [r88974-88995] Joshua Colp <jcolp@digium.com> + + * channels/chan_zap.c, /, configs/zapata.conf.sample: Merged + revisions 88994 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6 + lines Fix improbable but possible memory leaks in chan_zap. + (closes issue #11166) Reported by: eliel Patches: + chan_zap.c.patch uploaded by eliel (license 64) ........ + + * channels/chan_agent.c: Update chan_agent documentation. Change a + | to , as that is now the required way. (closes issue #11167) + Reported by: eliel Patches: chan_agent.c.patch uploaded by eliel + (license 64) + +2007-11-06 15:01 +0000 [r88973] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_unistim.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: Set up detection + of IP_PKTINFO in autoconf for chan_unistim + +2007-11-06 14:17 +0000 [r88932-88937] Russell Bryant <russell@digium.com> + + * channels/chan_unistim.c: convert uses of LOG_DEBUG to use + ast_debug() + + * channels/chan_unistim.c, configs/unistim.conf.sample: Add + jitterbuffer support to chan_unistim. (closes issue #11168) + Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by + IgorG (license 20) + + * main/pbx.c, /, channels/busy.h, channels/ringtone.h, + include/asterisk/pbx.h: Merged revisions 88805 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | + 12 lines After seeing crashes related to channel variables, I + went looking around at the ways that channel variables are + handled. In general, they were not handled in a thread-safe way. + The channel _must_ be locked when reading or writing from/to the + channel variable list. What I have done to improve this situation + is to make pbx_builtin_setvar_helper() and friends lock the + channel when doing their thing. Asterisk API calls almost all + lock the channel for you as necessary, but this family of + functions did not. (closes issue #10923, reported by atis) + (closes issue #11159, reported by 850t) ........ + + * /, include/asterisk/lock.h: Merged revisions 88931 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06 + Nov 2007) | 8 lines Remove some checks to see if locks are + initialized from the non-DEBUG_THREADS versions of the lock + routines. These are incorrect for a number of reasons: - It + breaks the build on mac. - If there is a problem with locks not + getting initialized, then the proper fix is to find that place + and fix the code so that it does get initialized. - If additional + debug code is needed to help find the problem areas, then this + type of things should _only_ be put in the DEBUG_THREADS + wrappers. ........ + +2007-11-06 08:17 +0000 [r88898-88913] Luigi Rizzo <rizzo@icir.org> + + * channels/Makefile: explain that the host environment must be used + to build gentone; Remove unset variables, they would be + misleading. + + * Makefile: don't export variables that can be retrieved from + makeopts in child subdirs + +2007-11-06 02:53 +0000 [r88863] Kevin P. Fleming <kpfleming@digium.com> + + * /, include/asterisk/srv.h: Merged revisions 88862 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r88862 | kpfleming | 2007-11-05 20:52:05 -0600 (Mon, 05 + Nov 2007) | 2 lines update comment to match the state of the code + ........ + +2007-11-05 23:31 +0000 [r88827] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 88826 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov + 2007) | 6 lines Reworked deadlock avoidance in __ast_read. + Restored audio to callback agents. (closes issue #11071, reported + by callguy, patched by me, tested by callguy and Ted Brown) + ........ + +2007-11-05 21:36 +0000 [r88770] Luigi Rizzo <rizzo@icir.org> + + * Makefile, utils/Makefile: Move AUDIO_LIBS outside the top level + Makefile. This too is used only in one place. + +2007-11-05 21:35 +0000 [r88769] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 88768 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | + 8 lines When traversing the list of channel variables here in + transmit_invite(), the asterisk channel must be locked, as this + data may change at any time. (I have seen numerous reports of + crashes related to the handling of channel variables. There are a + couple of issues on the bug tracker related to it, but it has + also been noted on IRC and mailing lists. So, I am finding and + fixing some places where channel variables are handled + improperly.) ........ + +2007-11-05 21:27 +0000 [r88767] Luigi Rizzo <rizzo@icir.org> + + * Makefile, main/Makefile: Move the last instance of AST_LIBS to + the only place it is used, namely main/Makefile . I am unclear + where decisions on the build environment (CFLAGS, LDFLAGS, LIBS + and so on) should be made - right now they are split here and + there. As a first step in cleaning up this situation, i am trying + to at least collect all instances of each variable in one place. + +2007-11-05 21:23 +0000 [r88766] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 88765 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) | + 2 lines Fix up some indentation. ........ + +2007-11-05 20:50 +0000 [r88764] Luigi Rizzo <rizzo@icir.org> + + * Makefile.moddir_rules: comment out an unused variable. Remove it + in a few days if no problems arise. + +2007-11-05 20:44 +0000 [r88710-88740] Russell Bryant <russell@digium.com> + + * main/srv.c, /, include/asterisk/srv.h: Merged revisions 88719 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) | + 7 lines Merge changes from + asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV + record support in Asterisk was broken. There was no guarantee on + what record Asterisk would choose to actually use. This set of + changes improves the situation by ensuring that Asterisk will + choose the highest priority record. ........ + + * main/channel.c, /: Merged revisions 88709 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) | + 20 lines Merge the last bit of changes from + asterisk/team/russell/readq-1.4 The issue here is that the + channel frame readq handling got broken when the code was + converted to use the linked list macros. It caused corruption of + the list head and tail pointers. So, I fixed up the usage of the + linked list macros and in passing, simplified the code. I also + documented what the code is doing, as it was a bit difficult to + figure out at first. This bug showed itself with crashes showing + messed up head/tail pointers for the readq. However, there are a + couple of crashes that aren't quite as obvious, but I think may + be related. So, if your bug gets closed by this commit, but you + still have a problem, please reopen or create a new bug report. + (closes issue #10936) (closes issue #10595) (closes issue #10368) + (closes issue #11084) (closes issue #10040) (closes issue #10840) + ........ + +2007-11-05 19:22 +0000 [r88675] Luigi Rizzo <rizzo@icir.org> + + * Makefile: Cleanup the installation of samples, avoiding + repetitions. I am preserving the behaviour on *.adsi files, i.e. + overwrite anything there without making a backup. However I am + not sure that this is the intended behaviour. + +2007-11-05 18:52 +0000 [r88673] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 88671 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7 + lines If a SIP channel is put on hold multiple times do not keep + incrementing the onHold value. (closes issue #11085) Reported by: + francesco_r Tested by: blitzrage (closes issue #10474) Reported + by: acennami ........ + +2007-11-05 18:22 +0000 [r88653] Tilghman Lesher <tlesher@digium.com> + + * CHANGES: Change wording to that suggested by MasterYoda + +2007-11-05 18:00 +0000 [r88652] Luigi Rizzo <rizzo@icir.org> + + * Makefile: simplify (hopefully) the printing of $(MAKE) in aligned + output. + +2007-11-05 17:52 +0000 [r88651] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 88624 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) | + 5 lines Fix up datastore handling in ast_do_masquerade(). The + code is intended to move any channel datastores from the old + channel to the new one. However, it did not use the linked list + macros properly to accomplish the task. The existing code would + only work if there was only a single datastore on the old + channel. ........ + +2007-11-05 17:44 +0000 [r88587-88615] Luigi Rizzo <rizzo@icir.org> + + * Makefile: print messages when entering/leaving a directory so we + know where we are (sometimes it is obvious, sometimes it is not). + + * Makefile.moddir_rules: merge two rules with the same right hand; + document a bit what is done here. + +2007-11-05 17:21 +0000 [r88586] Jason Parker <jparker@digium.com> + + * /, channels/chan_sip.c: Merged revisions 88585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11163) ........ r88585 | qwell | 2007-11-05 11:19:41 -0600 + (Mon, 05 Nov 2007) | 4 lines Make sure we destroy the config + structure on configuration failure. Issue 11163, patch by eliel. + ........ + +2007-11-05 17:00 +0000 [r88584] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile.rules: use a variable name that actually indicates what + it is for + +2007-11-05 16:41 +0000 [r88553] Luigi Rizzo <rizzo@icir.org> + + * Makefile.rules: Put extra compiler flags into a variable so they + are not repeated too many times. On passing, add some comments + and fix indentation a bit. On passing, i suspect that the + following pattern is wrong %.eoo: %.o but in case it will be + fixed in a later commit. + +2007-11-05 16:30 +0000 [r88540] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_odbc.c: Merged revisions 88539 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88539 | tilghman | 2007-11-05 10:20:13 -0600 (Mon, 05 Nov 2007) + | 4 lines Don't check used pooled connections for connection + status, as it will cause issues for prepared queries. Reported + by: Nick Gorham (via -dev list) Patch by: tilghman ........ + +2007-11-05 15:15 +0000 [r88525] Luigi Rizzo <rizzo@icir.org> + + * main/db.c: remove a cygwin-specific function remap that does not + work. + +2007-11-05 13:11 +0000 [r88510] Joshua Colp <jcolp@digium.com> + + * channels/chan_unistim.c: Fix memory leaks and deadlocks in + chan_unistim. (closes issue #11158) Reported by: eliel Patches: + chan_unistim.c.patch uploaded by eliel (license 64) + +2007-11-04 22:42 +0000 [r88454-88490] Luigi Rizzo <rizzo@icir.org> + + * /: block merging of not-applicable patch + + * main/channel.c, main/pbx.c, apps/app_meetme.c, + channels/chan_sip.c, res/res_features.c, main/utils.c, + channels/chan_iax2.c, include/asterisk/stringfields.h: Simplify + the implementation and the API for stringfields; details and + examples are in include/asterisk/stringfields.h. Not applicable + to older branches except for 1.4 which will receive a fix for the + routines that free memory pools. + +2007-11-03 14:19 +0000 [r88437] Tilghman Lesher <tlesher@digium.com> + + * main/term.c: Revert commit #86119. Some users intentionally do + not want colorized terminals, so this was a misfeature. + +2007-11-03 04:55 +0000 [r88422] James Golovich <james@gnuinter.net> + + * main/db.c: Set CLI command to the correct name. Rev 85460 + introduced two 'database show' commands when this one should have + been 'database showkey' + +2007-11-02 22:36 +0000 [r88368-88409] Russell Bryant <russell@digium.com> + + * channels/chan_unistim.c: fix some issues with crashing on unload, + when it didn't completely load cleanly + + * channels/chan_unistim.c: Convert the CLI commands to the new + format + + * pbx/pbx_lua.c: propagate the DECLINE return value back to the + loader + + * pbx/pbx_lua.c: Don't kill asterisk if extensions.lua is not + present. + + * main/cli.c: Show the channel unique ID in the "show channel + concise" output (closes issue #11148, requested by falves11, + patched by me) + + * channels/chan_unistim.c (added), CREDITS, + configs/unistim.conf.sample (added), CHANGES, doc/unistim.txt + (added): Merge the code from asterisk/team/group/chan_unistim: + This introduces a new channel driver, chan_unistim, that supports + the Unistim VoIP protocol for Nortel phones. The following models + have been confirmed to work: i2002, i2004 and i2050. (closes + issue #8864) Reported by: c_hans Patches: chan_unistim.patch + uploaded by c (license 304) ustm_no_conf.diff uploaded by junky + (license 177) Tested by: c_hans, dbowerman, math, junky, loloski + +2007-11-02 20:51 +0000 [r88329-88367] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 88366 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4 + lines Make subscribecontext behave as advertised. It will now + look for the presence of a hint in the given context (be it + subscribecontext or context). (closes issue #10702) Reported by: + slavon ........ + + * /, channels/chan_sip.c: Merged revisions 88328 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6 + lines If an INFO request within a dialog is received with a + content length of 0 simply send back a 200 OK. It is valid to do + this and the remote side is probably using it to make sure the + signalling is still alive. (closes issue #5747) Reported by: + chandi Patches: infofix-81430-1.patch uploaded by IgorG (license + 20) ........ + +2007-11-02 20:13 +0000 [r88327] Russell Bryant <russell@digium.com> + + * doc/tex/Makefile: Fix replacing the version number when it has a + '/' in it, like SVN-group-chan_unistim-r88326M-/trunk + +2007-11-02 17:34 +0000 [r88287] Tilghman Lesher <tlesher@digium.com> + + * pbx/pbx_lua.c: Oops, some dev-mode changes for ISO C90 + +2007-11-02 16:54 +0000 [r88284] Jason Parker <jparker@digium.com> + + * /, main/say.c: Merged revisions 88283 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11147) ........ r88283 | qwell | 2007-11-02 11:51:08 -0500 + (Fri, 02 Nov 2007) | 4 lines We need to make sure to specify a + language to ast_fileexists, otherwise it may fail for anything + besides en Issue 11147, fix discovered by both citats and myself + (independently), with input from Corydon76 ........ + +2007-11-02 16:26 +0000 [r88209-88267] Tilghman Lesher <tlesher@digium.com> + + * CHANGES: Add a few bytes on LUA + + * main/pbx.c, utils/build-extensions-conf.lua (added), + build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c + (added), configs/extensions.lua.sample (added), + include/asterisk/pbx.h, makeopts.in: Add pbx_lua as a method of + doing extensions Reported by: mnicholson Patch by: mnicholson + Closes issue #11140 + + * main/config.c: Don't re-cache the filename, but check to see if + it already exists Reported by: jamesgolovich Patch by: + jamesgolovich Closes issue #11144 + + * /, include/asterisk/lock.h: Merged revisions 88210 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r88210 | tilghman | 2007-11-02 08:03:03 -0500 (Fri, 02 + Nov 2007) | 5 lines Fix build on Solaris Reported by: snuffy + Patch by: ys Closes issue #11143 ........ + + * main/pbx.c: 'h' extension doesn't execute past first priority + Reported by: dimas Patch by: dimas Closes bug #11146 + +2007-11-02 03:09 +0000 [r88197] Joshua Colp <jcolp@digium.com> + + * cdr/cdr_odbc.c: Restore building under 64-bit platforms. + +2007-11-01 23:26 +0000 [r88184] Jason Parker <jparker@digium.com> + + * channels/chan_jingle.c, configure, + include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/jabber.h, channels/chan_gtalk.c, makeopts.in: + Remove traces of gnutls, since we no longer use/need it. + +2007-11-01 23:26 +0000 [r88182-88183] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c: Modify WaitExten to include an optional dialtone + Closes issue #10783 + + * UPGRADE.txt, cdr/cdr_odbc.c: Convert cdr_odbc to use res_odbc + managed connections Closes issue #10614 + +2007-11-01 22:26 +0000 [r88166] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, + funcs/func_strings.c, funcs/func_cut.c, funcs/func_logic.c, + apps/app_exec.c, apps/app_queue.c, apps/app_playback.c, + res/ael/pval.c, pbx/pbx_loopback.c, funcs/func_odbc.c, + apps/app_minivm.c, res/res_agi.c, main/logger.c, + pbx/pbx_realtime.c, apps/app_macro.c, pbx/pbx_dundi.c, + utils/extconf.c, include/asterisk/pbx.h, pbx/pbx_config.c, + apps/app_mixmonitor.c, apps/app_rpt.c, cdr/cdr_custom.c, + cdr/cdr_manager.c: This commits the performance mods that give + the priority processing engine in the pbx, a 25-30% speed boost. + The two updates used, are, first, to merge the + ast_exists_extension() and the ast_spawn_extension() where they + are called sequentially in a loop in the code, into a slightly + upgraded version of ast_spawn_extension(), with a few extra args; + and, second, I modified the substitute_variables_helper_full, so + it zeroes out the byte after the evaluated string instead of + demanding you pre-zero the buffer; I also went thru the code and + removed the code that zeroed this buffer before every call to the + substitute_variables_helper_full. The first fix provides about a + 9% speedup, and the second the rest. These figures come from the + 'PIPS' benchmark I describe in blogs, conf. reports, etc. + +2007-11-01 22:19 +0000 [r88164-88165] Jason Parker <jparker@digium.com> + + * /: Crap, accidentally copied the props. Thanks for pointing this + out mvanbaak. The odds are quite high that this will break + automerge on every team branch. + + * /, include/asterisk/jabber.h, res/res_jabber.c: Switch res_jabber + to use openssl rather than gnutls. Closes issue #9972, patch by + phsultan. Copied from branch at + http://svn.digium.com/svn/asterisk/team/phsultan/res_jabber-openssl/ + +2007-11-01 17:25 +0000 [r88117] Tilghman Lesher <tlesher@digium.com> + + * /, doc/valgrind.txt (added): Merged revisions 88116 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r88116 | tilghman | 2007-11-01 12:17:56 -0500 (Thu, 01 + Nov 2007) | 2 lines Add some notes on using valgrind ........ + +2007-11-01 16:22 +0000 [r88079] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, /: Merged revisions 88078 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4 + lines Make sure we set the poll fds to NULL after free()ing it. + Part of issue 11017, patch by tzafrir. ........ + +2007-11-01 15:56 +0000 [r88062-88077] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c, pbx/pbx_dundi.c: Change some uses of free() + to ast_free(). (No functional differences.) (closes issue #11138) + Reported by: eliel Patches: pbx_dundi.c.patch uploaded by eliel + (license 64) chan_sip.c.patch uploaded by eliel (license 64) + + * utils/Makefile: Remove another copied source file on "make + clean". (closes issue #11137) Reported by: IgorG Patches: + addonclean-87971-1.patch uploaded by IgorG (license 20) + +2007-11-01 13:30 +0000 [r88027] Joshua Colp <jcolp@digium.com> + + * /, apps/app_meetme.c: Merged revisions 88026 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r88026 | file | 2007-11-01 10:27:37 -0300 (Thu, 01 Nov 2007) | 2 + lines Fix up commit for my Zap channel with spies in Meetme fix. + (thanks Tony Mountifield!) ........ + +2007-11-01 06:12 +0000 [r88007-88010] Tilghman Lesher <tlesher@digium.com> + + * main/utils.c: Conditionally free lock_info->thread_name to avoid + a useless warning Reported by: snuffy Patch by: snuffy Closes + issue #11125 + + * apps/app_meetme.c, channels/chan_iax2.c: Janitor: use ast_free to + pair calls of ast_malloc and ast_calloc Reported by: eliel Patch + by: eliel Closes issue #11135 + + * cdr/cdr_adaptive_odbc.c: Fix memory leak Reported by: eliel Fixed + by: tilghman Closes issue #11136 + +2007-11-01 01:55 +0000 [r87953-87971] Joshua Colp <jcolp@digium.com> + + * /, apps/app_meetme.c: Merged revisions 87970 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4 + lines If a Zap channel contains a spy or a spy is added take it + out of the conference in kernel space and make it go through + Asterisk so the spy gets audio from both sides. (closes issue + #10060) Reported by: mparker ........ + + * main/pbx.c: Drop any more references to type in the Exception + dialplan function. (closes issue #11134) Reported by: blitzrage + Patches: exception_patch.txt uploaded by blitzrage (license 10) + +2007-10-31 21:23 +0000 [r87889-87909] Jason Parker <jparker@digium.com> + + * /, res/res_jabber.c: Merged revisions 87908 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11131) ........ r87908 | qwell | 2007-10-31 16:23:11 -0500 + (Wed, 31 Oct 2007) | 4 lines Make sure we free some allocated + memory before returning. Issue 11131, patch by eliel. ........ + + * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged + revisions 87906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11130) (closes issue #11132) ........ r87906 | qwell | + 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try + to allocate memory that we're just going to re-allocate later + anyways. Issues 11130 and 11132, patch by eliel. ........ + + * formats/format_sln.c, codecs/codec_adpcm.c, codecs/codec_gsm.c, + formats/format_wav_gsm.c, res/res_musiconhold.c, + codecs/codec_zap.c, formats/format_ilbc.c, res/res_smdi.c, + formats/format_pcm.c, formats/format_h263.c, + formats/format_h264.c, formats/format_jpeg.c, + formats/format_gsm.c, res/res_speech.c, res/res_clioriginate.c, + codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c, + formats/format_wav.c, codecs/codec_speex.c, codecs/codec_alaw.c, + res/res_adsi.c, res/res_convert.c, codecs/codec_g726.c, + formats/format_ogg_vorbis.c, res/res_ael_share.c, + formats/format_vox.c, codecs/codec_ulaw.c, formats/format_g723.c, + res/res_indications.c, codecs/codec_ilbc.c, + formats/format_g726.c, formats/format_g729.c: More changes to + change return values from load_module functions. (issue #11096) + Patches: codec_adpcm.c.patch uploaded by moy (license 222) + codec_alaw.c.patch uploaded by moy (license 222) + codec_a_mu.c.patch uploaded by moy (license 222) + codec_g722.c.patch uploaded by moy (license 222) + codec_g726.c.diff uploaded by moy (license 222) codec_gsm.c.patch + uploaded by moy (license 222) codec_ilbc.c.patch uploaded by moy + (license 222) codec_lpc10.c.patch uploaded by moy (license 222) + codec_speex.c.patch uploaded by moy (license 222) + codec_ulaw.c.patch uploaded by moy (license 222) + codec_zap.c.patch uploaded by moy (license 222) + format_g723.c.patch uploaded by moy (license 222) + format_g726.c.patch uploaded by moy (license 222) + format_g729.c.patch uploaded by moy (license 222) + format_gsm.c.patch uploaded by moy (license 222) + format_h263.c.patch uploaded by moy (license 222) + format_h264.c.patch uploaded by moy (license 222) + format_ilbc.c.patch uploaded by moy (license 222) + format_jpeg.c.patch uploaded by moy (license 222) + format_ogg_vorbis.c.patch uploaded by moy (license 222) + format_pcm.c.patch uploaded by moy (license 222) + format_sln.c.patch uploaded by moy (license 222) + format_vox.c.patch uploaded by moy (license 222) + format_wav.c.patch uploaded by moy (license 222) + format_wav_gsm.c.patch uploaded by moy (license 222) + res_adsi.c.patch uploaded by eliel (license 64) + res_ael_share.c.patch uploaded by eliel (license 64) + res_clioriginate.c.patch uploaded by eliel (license 64) + res_convert.c.patch uploaded by eliel (license 64) + res_indications.c.patch uploaded by eliel (license 64) + res_musiconhold.c.patch uploaded by eliel (license 64) + res_smdi.c.patch uploaded by eliel (license 64) + res_speech.c.patch uploaded by eliel (license 64) + +2007-10-31 18:53 +0000 [r87888] Steve Murphy <murf@digium.com> + + * /: Merged revisions 87849 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87849 | murf | 2007-10-31 11:49:39 -0600 (Wed, 31 Oct 2007) | 1 + line closes issue #11108 -- where the 'dialplan save' cli command + saves a file where the semicolon is not escaped. Fixed this; User + also wanted comments to be preserved across dialplan save, but + this is impossible at this point in time, because comments are + not stored in the dialplan. They are 'compiled' out of + extensions.conf. The only way to preserve those comments is to + use the config file reader/writer that the GUI uses to allow + online user edits. extensions.conf is first and foremost, a + config file, and is read in by the normal config-file reading + routines. Then, it is processed into a dialplan (context/exten + structs). (in the case of trunk, tho, no mods needed to be made + -- works OK there -- just make sure you use ',' to sep app args!) + ........ + +2007-10-31 18:09 +0000 [r87854] Tilghman Lesher <tlesher@digium.com> + + * Makefile, /: Merged revisions 87852 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87852 | tilghman | 2007-10-31 13:03:53 -0500 (Wed, 31 Oct 2007) + | 2 lines Create samples for ALL of the available options in + asterisk.conf ........ + +2007-10-31 18:03 +0000 [r87833-87851] Joshua Colp <jcolp@digium.com> + + * apps/app_mixmonitor.c: Add volume adjustment in. + + * apps/app_mixmonitor.c: Restore operation of the option that only + writes when the channel is bridged. + + * apps/app_chanspy.c: Add volume adjustment to spy audiohook in + app_chanspy. + +2007-10-31 16:13 +0000 [r87817] Tilghman Lesher <tlesher@digium.com> + + * CREDITS: Formatting cleanups, remove obsolete contributions + (modules no longer in Asterisk), and obfuscate email addresses + enough to stop most spam harvesters. + +2007-10-31 16:07 +0000 [r87815] Joshua Colp <jcolp@digium.com> + + * include/asterisk/channel.h: Remove old whisper remnants from + channel.h + +2007-10-31 15:46 +0000 [r87811] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c: Optimize pbx_substitute_variables + +2007-10-31 04:20 +0000 [r87776] Steve Murphy <murf@digium.com> + + * res/ael/pval.c, /: Merged revisions 87775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1 + line Included some verbage in the check_includes func, to inform + the user that included contexts that have no match in the AEL, + might be OK, as AEL cannot check in the extensions.conf or the + in-memory contexts, as they may not be there at the time of the + check. ........ + +2007-10-30 23:08 +0000 [r87724-87740] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 87739 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r87739 | tilghman | 2007-10-30 18:02:22 -0500 (Tue, 30 + Oct 2007) | 5 lines Fix for uninitialized mutexes on *BSD + Reported by: ys Fixed by: ys Closes issue #11116 ........ + + * apps/app_exec.c: If no '?' is found in the arguments, don't + attempt to continue. Reported by: blitzrage Fixed by: tilghman + Closes issue #11111 + +2007-10-30 21:22 +0000 [r87687] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 87686 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) | + 11 lines Merge the changes from team/russell/iax2_poke_fix and + iax2-poke-fix-trunk There was a race condition related to the + handling of POKEing peers. Essentially, a reference to a peer is + held by the scheduler when there are pending callbacks, but the + reference count didn't reflect it. So, it was possible for a peer + to hit a reference count of zero and have its destructor begin to + be called at the same time that the scheduler thread ran a POKE + related callback. If that happened, a crash would likely occur. + (closes issue #11082, closes issue #11094) ........ + +2007-10-30 20:30 +0000 [r87626-87651] Jason Parker <jparker@digium.com> + + * /, channels/Makefile: Merged revisions 87650 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1 + line Only try to clean out h323/ if the h323/Makefile exists. + ........ + + * main/pbx.c: Update documentation to give an example of how to use + the return status of RaiseException Closes issue #11117, patch by + blitzrage (yay blitzrage) + +2007-10-30 17:07 +0000 [r87573-87608] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c: The priority gets incremented after raising an + exception, so the priority should be set to 0 + + * main/pbx.c: Jumped the gun a bit in the RaiseException app. It + would always return -1 since it checked for the existence of + something that will never exist. + +2007-10-30 16:15 +0000 [r87572] Joshua Colp <jcolp@digium.com> + + * /, res/res_features.c: Merged revisions 87571 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87571 | file | 2007-10-30 13:13:39 -0300 (Tue, 30 Oct 2007) | 4 + lines Add two more checks before printing out a warning message + about bridging. If either channel has hungup of course the bridge + will have failed. (closes issue #10009) Reported by: dimas + ........ + +2007-10-30 15:47 +0000 [r87568] Jason Parker <jparker@digium.com> + + * /, main/editline/np/vis.c: Merged revisions 87567 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11113) ........ r87567 | qwell | 2007-10-30 10:45:35 -0500 + (Tue, 30 Oct 2007) | 4 lines Fix build of editline on Solaris. + Issue 11113, patch by snuffy. ........ + +2007-10-30 15:11 +0000 [r87535] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 87534 via svnmerge ........ r87534 | file | + 2007-10-30 12:10:13 -0300 (Tue, 30 Oct 2007) | 2 lines Return 1.4 + to a state where it builds. Changing the arguments to a function + and not changing where they are used is bad, mmmk? ........ + +2007-10-29 22:44 +0000 [r87462-87498] Kevin P. Fleming <kpfleming@digium.com> + + * utils/Makefile, utils, utils/hashtest2.c: UGH... while trying to + fix #10995, I found all kinds of cruft in this Makefile. It + should all be gone now, and as a side effect hashtest2 now builds + with --enable-dev-mode enabled without a host of errors + + * agi/Makefile, utils/Makefile, codecs/g722/Makefile, + main/editline/Makefile.in, Makefile.moddir_rules, + codecs/ilbc/Makefile, codecs/lpc10/Makefile, + main/db1-ast/Makefile: clean up assembler and preprocessor files + if they are here too + + * utils, agi, codecs, apps, cdr, codecs/ilbc, formats, funcs, + codecs/lpc10, main/db1-ast, codecs/g722, main/editline, main, + codecs/gsm, main/minimime, pbx, res, channels: ignore + preprocessor and assembler files if they are present + + * Makefile, /: Merged revisions 87460 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87460 | kpfleming | 2007-10-29 17:04:29 -0500 (Mon, 29 Oct 2007) + | 2 lines don't put '-pipe' into ASTCFLAGS if '-save-temps' is + already there (used when debugging preprocessor issues) because + the compiler will whine about each compile command ........ + +2007-10-29 21:34 +0000 [r87397-87428] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c: If a caller is listen-only, then don't bother + with doing talker detection. (closes issue #10911, reported by + junky, patched by me) + + * /, main/utils.c, include/asterisk/lock.h: Merged revisions 87396 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87396 | russell | 2007-10-29 15:22:07 -0500 (Mon, 29 Oct 2007) | + 5 lines Add some more details to the output of "core show locks". + When a thread is waiting for a lock, this will now show the + details about who currently has it locked. (inspired by issue + #11100) ........ + +2007-10-29 20:13 +0000 [r87395] Mark Michelson <mmichelson@digium.com> + + * UPGRADE.txt, apps/app_queue.c: Adding the more flexible + QUEUE_MEMBER function to replace the QUEUE_MEMBER_COUNT function. + A deprecation notice will be issued the first time + QUEUE_MEMBER_COUNT is used. + +2007-10-29 20:02 +0000 [r87394] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Drop the RTCP Read too short message to debug. There + are some phones out there that send a sort of keep alive packet + in the RTCP that trigger this every 5 seconds. + +2007-10-29 19:56 +0000 [r87393] Jason Parker <jparker@digium.com> + + * apps/app_record.c: Make sure we set flags to a 0 value before + trying to use it. Pointed out by seanbright while I was debugging + issue 11109. + +2007-10-29 19:47 +0000 [r87392] Russell Bryant <russell@digium.com> + + * /, main/astmm.c: Merged revisions 87373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87373 | russell | 2007-10-29 14:21:06 -0500 (Mon, 29 Oct 2007) | + 5 lines Remove a lock that doesn't make any sense. The regions + lock needs to be held when traversing the list of allocated + chunks so that they can be printed out to the CLI. (Thanks to + eliel on #asterisk-dev for pointing this out!) ........ + +2007-10-29 17:22 +0000 [r87343] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 87342 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6 + lines Fix issue where if both sides of the dialog cancelled the + dialog at the same time chan_sip could kepe retransmitting a + response for no reason. (closes issue #9566) Reported by: + atca_pres Patches: bug9566.patch uploaded by oej ........ + +2007-10-29 17:19 +0000 [r87341] Jason Parker <jparker@digium.com> + + * /: Blocked revisions 87340 via svnmerge (Closes issue #11104) + ........ r87340 | qwell | 2007-10-29 12:13:04 -0500 (Mon, 29 Oct + 2007) | 4 lines Allow some function modules to compile under dev + mode. Issue 11104, patch by andrew. ........ + +2007-10-29 16:38 +0000 [r87295-87327] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Remove duplicate stdlib.h include. (closes + issue #11105) Reported by: eliel Patches: app_voicemail.c.patch + uploaded by eliel (license 64) + + * channels/chan_misdn.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: Add autoconf + checks for extra suppserv definitions that are not present in + releases yet. chan_misdn should now build against the latest + release. (closes issue #11103) Reported by: IgorG + + * /, main/utils.c: Merged revisions 87294 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87294 | file | 2007-10-29 11:23:49 -0300 (Mon, 29 Oct 2007) | 6 + lines Fix issue with ast_unescape_semicolon going into an endless + loop. (closes issue #10550) Reported by: ramonpeek Patches: + unescape-85177-1.patch uploaded by IgorG (license 20) ........ + +2007-10-28 14:16 +0000 [r87263-87264] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_dialgroup.c (added): Add a simple dialgroup function. + By taking one of the simpler uses of Queue away from Queue, we + simplify the lives of people who do not need all the bells and + whistles. Also, this is part of the functions that people need to + reimplement Queue in the dialplan, as a set of logic, rather than + as a single app with hundreds of options. + + * /, funcs/func_odbc.c, funcs/func_strings.c, funcs/func_cut.c, + funcs/func_realtime.c: Merged revisions 87262 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87262 | tilghman | 2007-10-28 08:46:55 -0500 (Sun, 28 Oct 2007) + | 7 lines Add autoservice to several more functions which might + delay in their responses. Also, make sure that func_odbc + functions have a channel on which to set variables. Reported by + russell Fixed by tilghman Closes issue #11099 ........ + +2007-10-27 15:41 +0000 [r87233-87247] Russell Bryant <russell@digium.com> + + * configure, configure.ac: Update the configure script for the last + libss7 API change + + * funcs/func_shell.c, funcs/func_lock.c: Make sure a channel exists + before attempting to start or stop channel autoservice in + func_lock and func_shell. + +2007-10-27 00:48 +0000 [r87231-87232] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Add Circuit Group Queury message code + + * channels/chan_zap.c: Make sure we turn on the DSP when we answer + the call + +2007-10-26 22:21 +0000 [r87217] Mark Michelson <mmichelson@digium.com> + + * CHANGES: Forgot to update CHANGES when I committed the linear + queue strategy. Thank you Russell, for pointing this out! + +2007-10-26 21:37 +0000 [r87202] Jason Parker <jparker@digium.com> + + * channels/chan_local.c, channels/chan_zap.c, + channels/chan_agent.c, channels/chan_features.c, + res/res_crypto.c, res/res_realtime.c, res/res_monitor.c: + Correctly use defined return values in (some) load_module + functions. (issue #11096) Patches: chan_agent.c.patch uploaded by + eliel (license 64) chan_local.c.patch uploaded by eliel (license + 64) chan_features.c.patch uploaded by eliel (license 64) + chan_zap.c.patch uploaded by eliel (license 64) + res_monitor.c.patch uploaded by eliel (license 64) + res_realtime.c.patch uploaded by eliel (license 64) + res_crypto.c.patch uploaded by eliel (license 64) + +2007-10-26 17:39 +0000 [r87187] Steve Murphy <murf@digium.com> + + * res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael.tab.c, + res/ael/ael.y, pbx/pbx_ael.c, res/ael/ael_lex.c, + res/ael/ael.tab.h, utils/ael_main.c, + pbx/ael/ael-test/ref.ael-test16, res/ael/ael.flex, + utils/conf2ael.c, pbx/ael/ael-test/ref.ael-test19: Merged + revisions 87168 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87168 | murf | 2007-10-26 10:34:02 -0600 (Fri, 26 Oct 2007) | 1 + line closes issue #11086 where a user complains that references + to following contexts report a problem; The problem was REALLy + that he was referring to empty contexts, which were being + ignored. Reporter stated that empty contexts should be OK. I + checked it out against extensions.conf, and sure enough, empty + contexts ARE ok. So, I removed the restriction from AEL. This, + though, highlighted a problem with multiple contexts of the same + name. This should be OK, also. So, I added the extend keyword to + AEL, and it can preceed the 'context' keyword (mixed with + 'abstract', if nec.). This will turn off the warnings in AEL if + the same context name is used 2 or more times. Also, I now call + ast_context_find_or_create for contexts now, instead of just + ast_context_create; I did this because pbx_config does this. The + 'extend' keyword thus becomes a statement of intent. AEL can now + duplicate the behavior of pbx_config, ........ + +2007-10-26 15:19 +0000 [r87153-87154] Mark Michelson <mmichelson@digium.com> + + * configs/queues.conf.sample, apps/app_queue.c: Added queue + strategy "linear". This strategy is useful for those who always + wish for their phones to be rung in a specific order. (closes + issue #7279, reported and initially patched by diLLec, patch + reworked by me) + + * configs/queues.conf.sample: Remove information about the + roundrobin strategy from trunk's queues.conf.sample since it no + longer exists + +2007-10-26 14:00 +0000 [r87103-87121] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_curl.c, /: Merged revisions 87120 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87120 | tilghman | 2007-10-26 08:54:30 -0500 (Fri, 26 Oct 2007) + | 7 lines The addition of autoservice to func_curl additionally + made func_curl dependent on the existence of a channel, with no + real reason. This should make func_curl once again work without a + channel. Reported by jmls. Fixed by tilghman. Closes issue #11090 + ........ + + * /: Blocked revisions 87067 via svnmerge ........ r87067 | + tilghman | 2007-10-25 17:53:06 -0500 (Thu, 25 Oct 2007) | 4 lines + Backport alternate encoding of newline delimiters from trunk to + 1.4, as approved by Russell Reported by blitzrage Closes issue + #10903 ........ + + * include/asterisk/app.h, funcs/func_strings.c, funcs/func_cut.c, + main/app.c: Use the same delimited character as the FILTER + function in FIELDQTY and CUT. + +2007-10-25 23:11 +0000 [r87070] Kevin P. Fleming <kpfleming@digium.com> + + * main/channel.c, /, include/asterisk/linkedlists.h: Merged + revisions 87069 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r87069 | kpfleming | 2007-10-25 18:03:11 -0500 (Thu, 25 Oct 2007) + | 2 lines appending one list to another should leave the first + list empty, and not require the user to do that ........ + +2007-10-25 18:59 +0000 [r87040] Russell Bryant <russell@digium.com> + + * apps/app_meetme.c: Add support for a muted user to request to + talk. The '2' option in the user menu will adjust this status if + a user is muted. The talk request status will be reflected in the + CLI commands as well as the manager interface. (closes issue + #9418) Reported by: imesper Patches: app_meetme_v2.patch uploaded + by imesper (license 275) + +2007-10-25 16:21 +0000 [r87024] Steve Murphy <murf@digium.com> + + * main/ast_expr2.y, res/res_config_sqlite.c, main/ast_expr2.c: + closes issue #11045 - each file needs to define + ASTERISK_FILE_VERSION, if you are going to set MTX_PROFILE in the + compiler flags; the problem was that the fixes were getting made + to the generated .c file, and erased the next time someone + regenerated that file from the corresponding .y or .flex file. + Moral of story: keep your eyes open and make mods to the .y (or + flex input file) and re-run bison (or flex) as the Makefile + directs for that file, and then check in both. Also, + res_config_sqlite was kinda missed, and has the same issue. + +2007-10-24 21:26 +0000 [r86985] Mark Michelson <mmichelson@digium.com> + + * configs/queues.conf.sample, apps/app_queue.c: Adding the general + option "shared_lastcall" to queues so that a member's wrapuptime + may be used across multiple queues. (closes issue #9777, reported + and patched by eliel) + +2007-10-24 20:59 +0000 [r86983] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, /: Merged revisions 86982 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #11079) ........ r86982 | qwell | 2007-10-24 15:56:47 -0500 + (Wed, 24 Oct 2007) | 5 lines Correctly respect hidecalleridname + configuration option. Simplify code slightly in the process. + Issue 11079, reported by ddv2005 ........ + +2007-10-24 13:21 +0000 [r86900-86967] Steve Murphy <murf@digium.com> + + * pbx/ael/ael-test/ref.ael-ntest22, pbx/ael/ael-test/ref.ael-test2, + pbx/ael/ael-test/ref.ael-test3, res/ael/ael_lex.c, + pbx/ael/ael-test/ref.ael-test4, res/ael/ael.flex: closes issue + #11005, where #include uses the current dir instead of the config + dir (/etc/asterisk) for relative path includes for AEL + + * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 86936 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86936 | murf | 2007-10-23 22:14:28 -0600 (Tue, 23 Oct 2007) | 1 + line closes issue #11037 -- unable to specify app:spec in hint + arguments ........ + + * /, funcs/func_logic.c: Merged revisions 86902 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86902 | murf | 2007-10-23 15:18:08 -0600 (Tue, 23 Oct 2007) | 1 + line closes issue #11052 -- where nothing after the ? will allow + un-initialized variable values to corrupt and crash asterisk on + 64-bit platforms ........ + + * /, main/ast_expr2f.c: Merged revisions 86880 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86880 | murf | 2007-10-23 14:20:54 -0600 (Tue, 23 Oct 2007) | 1 + line This should get rid of a really, really irritating warning + generated by some 64-bit platforms from libc, where free(0) is + frowned upon ........ + + * /, main/Makefile: Merged revisions 86881 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86881 | murf | 2007-10-23 14:22:25 -0600 (Tue, 23 Oct 2007) | 1 + line this update to Makefile corrects how ast_expr2f.c should be + generated ........ + +2007-10-22 21:37 +0000 [r86835-86839] Russell Bryant <russell@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 86836 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r86836 | russell | 2007-10-22 16:36:12 -0500 (Mon, 22 + Oct 2007) | 9 lines If lock tracking is not enabled, then we can + not attempt to log any mutex failures. If so, we could end up in + infinite recursion. The only lock that is affected by this is a + mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes issue + #11044) Reported by: ys Patches: lock.h.diff uploaded by ys + (license 281) ........ + + * apps/app_playback.c: Convert some spaces to tabs and make it so + the CLI command is only registered once instead of 3 times. + (closes issue #11053) Reported by: seanbright Patches: + app_playback.patch uploaded by seanbright (license 71) + +2007-10-22 20:05 +0000 [r86820] Jason Parker <jparker@digium.com> + + * main/udptl.c, channels/chan_local.c, main/frame.c, + res/res_features.c, main/threadstorage.c, channels/chan_iax2.c, + main/astobj2.c, main/config.c, main/cli.c, + channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c, + channels/chan_alsa.c, main/db.c, main/pbx.c, + channels/chan_agent.c, channels/iax2-provision.c, + apps/app_playback.c, channels/chan_misdn.c, + channels/chan_features.c, res/res_indications.c, + pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c, + res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c, + main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c, + res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c, + main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c, + res/res_agi.c, apps/app_minivm.c, main/logger.c, + res/res_realtime.c, main/image.c, apps/app_rpt.c, + channels/chan_mgcp.c, res/res_clioriginate.c, + res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, + channels/chan_sip.c, res/res_limit.c, main/translate.c, + res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h, + apps/app_queue.c, channels/chan_oss.c, main/rtp.c, + channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c, + channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c, + funcs/func_devstate.c: Switch from AST_CLI (formerly NEW_CLI) to + AST_CLI_DEFINE, since the former didn't make much sense + +2007-10-22 17:40 +0000 [r86790] Tilghman Lesher <tlesher@digium.com> + + * /, main/astmm.c: Merged revisions 86787 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86787 | tilghman | 2007-10-22 12:38:13 -0500 (Mon, 22 Oct 2007) + | 2 lines Minor FreeBSD build fix ........ + +2007-10-22 16:36 +0000 [r86755-86757] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 86756 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86756 | file | 2007-10-22 13:35:22 -0300 (Mon, 22 Oct 2007) | 4 + lines After reading online I have confirmed that Record-Route + headers should be copied to 1xx responses as well. (closes issue + #10113) Reported by: makoto ........ + + * /, apps/app_controlplayback.c: Merged revisions 86754 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86754 | file | 2007-10-22 13:15:18 -0300 (Mon, 22 Oct 2007) | 4 + lines Make sure res is a positive value before performing the + check to determine whether the user stopped it or not. (closes + issue #11023) Reported by: cfc ........ + +2007-10-22 15:57 +0000 [r86734-86751] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 86750 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86750 | russell | 2007-10-22 10:52:48 -0500 (Mon, 22 Oct 2007) | + 8 lines Don't leak a frame in the case that an END frame is + received and the time since the BEGIN is less than that of the + defined minimum DTMF duration. (closes issue #11051) Reported by: + casper Patches: channel.c.86664.diff uploaded by casper (license + 55) ........ + + * channels/chan_zap.c: There is a really fun game that you can play + before committing code, and it's called "make". :) + + * /, include/asterisk/lock.h: Merged revisions 86726 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r86726 | russell | 2007-10-22 10:43:30 -0500 (Mon, 22 + Oct 2007) | 4 lines Update the static mutex initializer to + include the initialization of the internal mutex used to protect + the lock debugging data. (closes issue #11044, patch suggested by + Ivan) ........ + +2007-10-22 14:59 +0000 [r86697] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, configs/zapata.conf.sample: resetinterval + defaulting to something other than 'never' doesn't seem to + accomplish any good and causes problems for plenty of people... + +2007-10-22 14:58 +0000 [r86696] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 86694 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86694 | mmichelson | 2007-10-22 09:48:46 -0500 (Mon, 22 Oct + 2007) | 5 lines Account for the fact that sometimes headers may + be terminated with \r\n instead of just \n (closes issue #11043, + reported by yehavi) ........ + +2007-10-22 14:56 +0000 [r86695] Kevin P. Fleming <kpfleming@digium.com> + + * main/loader.c: merging patches that don't compile is bad... + mmkay? + +2007-10-22 14:28 +0000 [r86631-86664] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 86663 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86663 | file | 2007-10-22 11:27:03 -0300 (Mon, 22 Oct 2007) | 6 + lines Move log message to before the frame it references is + freed. (closes issue #11050) Reported by: slavon Patches: + channel.c.86662.diff uploaded by casper (license 55) ........ + + * /, pbx/pbx_dundi.c: Merged revisions 86661 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86661 | file | 2007-10-22 11:05:26 -0300 (Mon, 22 Oct 2007) | 6 + lines Fix tab completion for dundi show peer. (closes issue + #11041) Reported by: jsmith Patches: + asterisk-dundicomplete.diff.txt uploaded by jamesgolovich + (license 176) ........ + + * /, main/acl.c, main/loader.c: Merged revisions 86630 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r86630 | file | 2007-10-22 10:33:23 -0300 (Mon, 22 Oct + 2007) | 6 lines Fixes for building under OpenSolaris. (closes + issue #11047) Reported by: snuffy Patches: 11047-fixes.diff + uploaded by snuffy (license 35) ........ + +2007-10-22 10:18 +0000 [r86616-86617] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged + revisions 86598 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86598 | crichter | 2007-10-22 11:21:15 +0200 (Mo, 22 Okt 2007) | + 1 line we send DISCONNECT instead of RELEASE/RELEASE_COMPLETE if + the dialplan does not match after an overlap call. Also added + out_cause=1 ........ + + * channels/misdn/isdn_lib.h, channels/chan_misdn.c, + channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: + started to add some basic support for supplementary services like + CallForwarding and so forth + +2007-10-21 22:52 +0000 [r86585] Russell Bryant <russell@digium.com> + + * /, include/asterisk/cli.h, main/asterisk.c, main/cli.c: Merged + revisions 85532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85532 | russell | 2007-10-13 00:24:33 -0500 (Sat, 13 Oct 2007) | + 8 lines Properly handle the case where read() may return the text + for more than one CLI command at once for a remote console. + (closes issue #10888) Reported by: jamesgolovich Patches: + asterisk-climultiple.diff.txt uploaded by jamesgolovich (license + 176) ........ + +2007-10-20 19:56 +0000 [r86572] Matthew Fredrickson <creslin@digium.com> + + * configs/zapata.conf.sample: Improved comments and organization + for zapata.conf (#10904) + +2007-10-19 18:46 +0000 [r86549] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Add better support for blocking and + unblocking of CICs (#10965) + +2007-10-19 18:29 +0000 [r86534-86536] Jason Parker <jparker@digium.com> + + * main/udptl.c, channels/chan_local.c, main/frame.c, + res/res_features.c, main/threadstorage.c, channels/chan_iax2.c, + main/astobj2.c, main/config.c, main/cli.c, + channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c, + channels/chan_alsa.c, main/db.c, main/pbx.c, + channels/chan_agent.c, channels/iax2-provision.c, + apps/app_playback.c, channels/chan_misdn.c, + channels/chan_features.c, res/res_indications.c, + pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c, + res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c, + main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c, + res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c, + main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c, + res/res_agi.c, apps/app_minivm.c, main/logger.c, + res/res_realtime.c, main/image.c, apps/app_rpt.c, + channels/chan_mgcp.c, res/res_clioriginate.c, + res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, + channels/chan_sip.c, res/res_limit.c, main/translate.c, + res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h, + apps/app_queue.c, channels/chan_oss.c, main/rtp.c, + channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c, + channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c, + funcs/func_devstate.c: Convert NEW_CLI to AST_CLI. Closes issue + #11039, as suggested by seanbright. + + * channels/chan_usbradio.c, res/res_config_pgsql.c, + channels/chan_misdn.c, channels/chan_h323.c, + res/res_indications.c, channels/chan_iax2.c, codecs/codec_zap.c, + res/res_config_sqlite.c, main/config.c, main/rtp.c: More changes + to NEW_CLI. Also fixes a few cli messages and some minor + formatting. (closes issue #11001) Reported by: seanbright + Patches: newcli.1.patch uploaded by seanbright (license 71) + newcli.2.patch uploaded by seanbright (license 71) newcli.4.patch + uploaded by seanbright (license 71) newcli.5.patch uploaded by + seanbright (license 71) newcli.6.patch uploaded by seanbright + (license 71) newcli.7.patch uploaded by seanbright (license 71) + +2007-10-19 16:40 +0000 [r86470-86503] Joshua Colp <jcolp@digium.com> + + * /, main/app.c: Merged revisions 86502 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86502 | file | 2007-10-19 13:38:29 -0300 (Fri, 19 Oct 2007) | 4 + lines When returning a DTMF digit from ast_control_streamfile + cast it as a char so that 0 does not overlap with the success + return code. (closes issue #11023) Reported by: cfc ........ + + * /, channels/chan_sip.c: Merged revisions 86471 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6 + lines Fix two issues with domains and transfers. If a port was + given in the hostname it was treated as part of the hostname. If + domains were configured but external domains were not enabled all + transfers would be considered remote. (closes issue #11027) + Reported by: ramonpeek Patches: 11027-1.diff uploaded by + ramonpeek (license 266) ........ + + * /, channels/chan_sip.c: Merged revisions 86469 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86469 | file | 2007-10-19 12:08:12 -0300 (Fri, 19 Oct 2007) | 4 + lines Set port number in received as information for + registrations as well. (closes issue #11028) Reported by: brad-x + ........ + +2007-10-19 01:56 +0000 [r86439] TransNexus OSP Development <support@transnexus.com> + + * apps/app_osplookup.c: Fixed a buffer size issue. + +2007-10-18 22:03 +0000 [r86407-86408] Jason Parker <jparker@digium.com> + + * Makefile, /: Merged revisions 86405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes + issue #11029) ........ r86405 | qwell | 2007-10-18 16:58:44 -0500 + (Thu, 18 Oct 2007) | 4 lines Add documentation for options in + asterisk.conf Issue 11029, patch by eserra ........ + + * /: Blocked revisions 86406 via svnmerge ........ r86406 | qwell | + 2007-10-18 17:01:02 -0500 (Thu, 18 Oct 2007) | 1 line Correct + documentation. I removed the wrong line.. ........ + +2007-10-18 21:19 +0000 [r86373] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 86371-86372 via svnmerge ........ r86371 | + russell | 2007-10-18 16:14:15 -0500 (Thu, 18 Oct 2007) | 2 lines + Add support for setting the maximum trunk size for IAX2 trunking + ........ r86372 | russell | 2007-10-18 16:16:47 -0500 (Thu, 18 + Oct 2007) | 2 lines Revert erroneous commit. ........ + +2007-10-18 18:40 +0000 [r86350] Mark Michelson <mmichelson@digium.com> + + * channels/chan_zap.c: Fixing a segfault from tab-completing a "zap + restart" CLI command. (patch made by seanbright, pointed out in + #asterisk-dev on IRC) + +2007-10-18 18:06 +0000 [r86331] Russell Bryant <russell@digium.com> + + * main/channel.c, /, include/asterisk/channel.h: Merged revisions + 86330 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86330 | russell | 2007-10-18 13:03:10 -0500 (Thu, 18 Oct 2007) | + 10 lines The channel needs to stay locked while running timer + callbacks, as they access and modify channel data that may change + elsewhere. I went through every timer callback in the source tree + to make sure that none of them did any additional locking that + could introduce deadlocks, and all is well. (closes issue #10765) + Reported by: Ivan Patches: ast_1_4_11_svn_patch_channel_rc.diff + uploaded by Ivan (license 229) ........ + +2007-10-18 17:40 +0000 [r86298-86329] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 86328 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86328 | mmichelson | 2007-10-18 12:38:26 -0500 (Thu, 18 Oct + 2007) | 5 lines If a non-existent file is specified to be played + either as a periodic announcement or as a hold/position + announcement, the caller would be kicked out of the queue. No + longer does this happen. ........ + + * apps/app_queue.c: Changed some spaces to tabs + +2007-10-18 15:57 +0000 [r86297] Russell Bryant <russell@digium.com> + + * /, codecs/codec_zap.c: Merged revisions 86296 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86296 | russell | 2007-10-18 10:45:55 -0500 (Thu, 18 Oct 2007) | + 3 lines Execute the RELEASE operation on transcoder channels in + the destroy callback. (patch from jsloan) ........ + +2007-10-18 07:23 +0000 [r86277-86278] Tilghman Lesher <tlesher@digium.com> + + * main/acl.c: Code cleanup of acl.c Reported by dimas Closes issue + #10784 + + * res/res_musiconhold.c: On reload, re-read the files in the + specified moh directory (closes issue #10536) + +2007-10-18 04:41 +0000 [r86238] Russell Bryant <russell@digium.com> + + * /, main/utils.c: Merged revisions 86237 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86237 | russell | 2007-10-17 23:40:52 -0500 (Wed, 17 Oct 2007) | + 9 lines Revert a change that I made for issue #10979 which, as + has been pointed out to me in issue #11018, doesn't really make + sense. There is no reason to have the base64 decode function + force a '\0' terminated buffer, when the result is almost always + binary, anyway. In fact, this caused some breakage, as some code + in res_crypto passed in a buffer exactly the right size to get + its binary result, which got stomped on by this patch. (closes + issue #11018, reported by dimas) ........ + +2007-10-17 21:41 +0000 [r86208] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 86202 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86202 | mmichelson | 2007-10-17 16:39:05 -0500 (Wed, 17 Oct + 2007) | 6 lines Changing the strategy field of the call_queue + struct to be signed instead of unsigned, since the code attempts + to set the strategy to -1 if you specify a bogus strategy. While + this isn't a huge issue in 1.4, it could be a problem for someone + who, say, tries to use the roundrobin strategy in trunk (despite + all the deprecation warnings in 1.4). ........ + +2007-10-17 21:16 +0000 [r86195-86197] Tilghman Lesher <tlesher@digium.com> + + * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Simplify + some preprocessor logic by using #elif + + * CHANGES, configs/meetme.conf.sample: Document the changes made + earlier today to meetme + +2007-10-17 20:06 +0000 [r86180-86182] Steve Murphy <murf@digium.com> + + * utils/hashtest2.c, utils/check_expr.c, utils/clicompat.c: and + then, I noticed the clicompat stuff. + + * utils/check_expr.c: more stub routines to allow linkage in + stand-alone environment, with thread debugs turned on + + * utils/hashtest2.c: more stub routines to allow linkage in + stand-alone environment, with thread debugs turned on + +2007-10-17 18:01 +0000 [r86150] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 86149 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) | + 8 lines If Asterisk is in the middle of shutting down, respond to + OPTIONS with 503 Unavailable. (closes issue #10994) Reported by: + eserra Patches: sip-options-503.patch uploaded by eserra (license + 45) ........ + +2007-10-17 17:06 +0000 [r86119] Tilghman Lesher <tlesher@digium.com> + + * main/term.c: Support color on certain platforms, even when + started at boot (before TERM is set) Closes issue #9048 + +2007-10-17 17:00 +0000 [r86118] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 86117 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86117 | file | 2007-10-17 13:58:03 -0300 (Wed, 17 Oct 2007) | 4 + lines Whoops, forgot to remove the original sip_scheddestroy. + (closes issue #11010) Reported by: vadim ........ + +2007-10-17 16:09 +0000 [r86104] Jason Parker <jparker@digium.com> + + * channels/chan_usbradio.c, channels/xpmr/xpmr.c: Allow + chan_usbradio to compile again. Closes issue #11014, patch by + seanbright. + +2007-10-17 15:39 +0000 [r86079] Tilghman Lesher <tlesher@digium.com> + + * /, main/asterisk.c: Merged revisions 86066 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86066 | tilghman | 2007-10-17 10:23:51 -0500 (Wed, 17 Oct 2007) + | 3 lines When runuser/rungroup is specified, a remote console + could only be attained by root (Closes issue #9999) ........ + +2007-10-17 15:30 +0000 [r86067] Joshua Colp <jcolp@digium.com> + + * channels/chan_usbradio.c: Change dependency for chan_usbradio to + asound. Let's keep everything uniform. (closes issue #11013) + Reported by: seanbright + +2007-10-17 15:13 +0000 [r86065] Tilghman Lesher <tlesher@digium.com> + + * apps/app_meetme.c: Enhancements to realtime (closes issue #9609) + +2007-10-17 15:09 +0000 [r86064] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 86063 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86063 | file | 2007-10-17 12:06:36 -0300 (Wed, 17 Oct 2007) | 4 + lines Don't schedule dialog destruction if a MESSAGE is received + using an existing dialog. (closes issue #11010) Reported by: + vadim ........ + +2007-10-16 23:36 +0000 [r86029-86033] Mark Michelson <mmichelson@digium.com> + + * /, configs/queues.conf.sample: Merged revisions 86032 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r86032 | mmichelson | 2007-10-16 18:35:31 -0500 (Tue, 16 Oct + 2007) | 3 lines Since monitor-join is deprecated now, remove the + example from the sample queues.conf file ........ + + * apps/app_queue.c: Removed the monitor-join option. If one wishes + to mix audio, they should instead use monitor-type=mixmonitor. + (related to issue #10885) + + * /: Blocked revisions 86028 via svnmerge ........ r86028 | + mmichelson | 2007-10-16 17:49:10 -0500 (Tue, 16 Oct 2007) | 6 + lines Adding deprecated warning to monitor-join option, since the + plan is to no longer support this in favor of monitor-type = + mixmonitor (related to issue #10885) ........ + +2007-10-16 22:36 +0000 [r85995-85998] Russell Bryant <russell@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 85997 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r85997 | russell | 2007-10-16 17:36:16 -0500 (Tue, 16 + Oct 2007) | 1 line really picky formatting tweak ... ........ + + * /, include/asterisk/lock.h: Merged revisions 85994 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r85994 | russell | 2007-10-16 17:14:36 -0500 (Tue, 16 + Oct 2007) | 16 lines Some locking errors exposed the fact that + the lock debugging code itself was not thread safe. How ironic! + Anyway, these changes ensure that the code that is accessing the + lock debugging data is thread-safe. Many thanks to Ivan for + finding and fixing the core issue here, and also thanks to those + that tested the patch and provided test results. (closes issue + #10571) (closes issue #10886) (closes issue #10875) (might close + some others, as well ...) Patches: (from issue #10571) + ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license + 229) - a few small changes by me ........ + +2007-10-16 21:51 +0000 [r85959-85992] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fixing the build. + + * apps/app_read.c: Fixing app_read so that if a timeout of less + than 1 ms is specified, assume that 1 ms is desired. (closes + issue #11000, reported and patched by michael-fig, with a warning + line added by me) + + * /, apps/app_queue.c: Merged revisions 85958 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85958 | mmichelson | 2007-10-16 16:14:34 -0500 (Tue, 16 Oct + 2007) | 5 lines Trying to remove a non-dynamic queue member via + dynamic means can lead to some interesting (read nasty) + situations. This patch clears up the issue by making only dynamic + queue members removable via dynamic methods. ........ + +2007-10-16 20:55 +0000 [r85957] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Don't hangup the call for SS7 if we get an + alarm + +2007-10-16 20:32 +0000 [r85944] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: This fixes SIP subscriptions in trunk. Don't + improperly memset() over an ast_str. This was leftover from + before it got changed to use ast_str. (closes issue #11003, + reported by pj) (closes issue #10770, reported by yehavi) + (patched by me) + +2007-10-16 19:47 +0000 [r85943] Tilghman Lesher <tlesher@digium.com> + + * /, main/stdtime/localtime.c: Merged revisions 85921 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r85921 | tilghman | 2007-10-16 14:41:40 -0500 (Tue, 16 + Oct 2007) | 4 lines Also set up gmtoff (this is used in the %z + gnu extension to strftime) Reported and fixed by jcmoore Closes + issue #11002 ........ + +2007-10-16 19:12 +0000 [r85897] Russell Bryant <russell@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 85896 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85896 | russell | 2007-10-16 14:10:01 -0500 (Tue, 16 Oct 2007) | + 2 lines Remove a pointless lock. ........ + +2007-10-16 16:40 +0000 [r85853-85883] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Fix IMAP compilation error. (closes issue + #10986, reported and patched by snuffy) + + * /: Blocking changes from previous commit + +2007-10-16 15:15 +0000 [r85819-85851] Joshua Colp <jcolp@digium.com> + + * /, funcs/func_vmcount.c: Merged revisions 85850 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85850 | file | 2007-10-16 11:52:22 -0300 (Tue, 16 Oct 2007) | 4 + lines Check to make sure a value has been given to the VMCOUNT + dialplan function. (closes issue #10996) Reported by: marsosa + ........ + + * main/threadstorage.c: Permit building under DEBUG_THREADLOCALS. + Thanks snuff. + + * /, main/threadstorage.c: Merged revisions 85818 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85818 | file | 2007-10-16 11:19:39 -0300 (Tue, 16 Oct 2007) | 6 + lines Fix memory allocation issue in threadstorage. (closes issue + #10995) Reported by: snuffy Patches: new-patch.diff uploaded by + snuffy (license 35) ........ + +2007-10-16 10:38 +0000 [r85777-85787] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_jingle.c, channels/chan_gtalk.c: Fix CLI help + output + + * channels/chan_jingle.c: Added two CLI functions, taken from + chan_gtalk : - jingle reload ; - jingle show channels. + + * channels/chan_jingle.c: Make an audio path under the following + call configuration : SIP Phone 1 --- [chan_sip]Asterisk + 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP + Phone 2 Modifications : - set bridge type to partial ; - process + media candidates from the remote peer properly. Now we have + Jingle audio, at least between two Asterisk Jingle clients. + +2007-10-15 23:20 +0000 [r85764] Jason Parker <jparker@digium.com> + + * configs/dundi.conf.sample, channels/chan_sip.c, + channels/chan_h323.c, main/acl.c, UPGRADE.txt, + channels/iax2-provision.c, doc/tex/qos.tex, pbx/pbx_dundi.c, + channels/chan_iax2.c, channels/chan_mgcp.c: Switch dundi to new + tos config format. Remove old unused defines for old style. + Closes issue 10860, patch by IgorG. + +2007-10-15 22:03 +0000 [r85751] Tilghman Lesher <tlesher@digium.com> + + * /: Blocked revisions 85687 via svnmerge ........ r85687 | + tilghman | 2007-10-15 15:29:35 -0500 (Mon, 15 Oct 2007) | 5 lines + Don't execute a gosub if the arguments is zero-len (not just + NULL) Reported by davevg Fixed by me Closes issue #10985 ........ + +2007-10-15 21:11 +0000 [r85718-85721] Russell Bryant <russell@digium.com> + + * /, apps/app_queue.c: Merged revisions 85720 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85720 | russell | 2007-10-15 16:10:02 -0500 (Mon, 15 Oct 2007) | + 3 lines Ensure that no pending state changes are leaked when the + device state change thread gets stopped on module unload. + ........ + + * /: Blocked revisions 85717 via svnmerge ........ r85717 | russell + | 2007-10-15 15:59:27 -0500 (Mon, 15 Oct 2007) | 7 lines + Previously, app_queue created a thread to handle every single + device state change. I changed this a while ago in trunk for + performance reasons. However, bug 8407 points out that it is + actually a race condition, causing device state changes to get + processed in random order. So, I backported my changes from trunk + to 1.4. (closes issue #8407, patch provided by tim_ringenbach, + committed patch by me) ........ + + * /, main/say.c: Merged revisions 85686 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85686 | russell | 2007-10-15 15:21:27 -0500 (Mon, 15 Oct 2007) | + 7 lines Add a small fix for the tw version of saying dates. + (closes issue #7827) Reported by: sharkey Patches: say.nits.patch + uploaded by sharkey (license 172) ........ + +2007-10-15 20:16 +0000 [r85685] Jason Parker <jparker@digium.com> + + * Makefile, /: Merged revisions 85684 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #10938) ........ r85684 | qwell | 2007-10-15 15:15:51 -0500 + (Mon, 15 Oct 2007) | 5 lines Properly use DESTDIR in 'config' + target. Do not try to run chkconfig or similar if using DESTDIR. + Issue 10938, patch by cabal95. ........ + +2007-10-15 20:09 +0000 [r85648-85683] Russell Bryant <russell@digium.com> + + * doc/tex/channelvariables.tex: add TOUCH_MONITOR_PREF to the + channel var docs + + * res/res_features.c, CHANGES: Added support for reading the + TOUCH_MONITOR_PREFIX channel variable. It allows you to configure + a prefix for auto-monitor recordings. (closes issue #6353) + Reported by: ivanfm Patches: asterisk_automon_v4.patch uploaded + by ivanfm (original patch) - updated patch: + 6353-touch_monitor_prefix.diff uploaded by qwell (license 4) + + * /, main/utils.c: Merged revisions 85649 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85649 | russell | 2007-10-15 14:22:45 -0500 (Mon, 15 Oct 2007) | + 2 lines Be pedantic about handling memory allocation failure. + ........ + + * /, main/utils.c: Merged revisions 85647 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85647 | russell | 2007-10-15 14:11:38 -0500 (Mon, 15 Oct 2007) | + 5 lines The loop in the handler for the "core show locks" could + potentially block for some amount of time. Be a little bit more + careful and prepare all of the output in an intermediary buffer + while holding a global resource. Then, after releasing it, send + the output to ast_cli(). ........ + +2007-10-15 17:51 +0000 [r85633] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_strings.c: Document my changes from Friday + +2007-10-15 16:59 +0000 [r85605] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 85604 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) | + 6 lines Make the default for the srvlookup option to be yes. It + doesn't really make sense for it to default to off. The default + configuration file has it on, and proper RFC behavior, as + indicated by a comment in the code, is for it to be on. So, let's + have it on by default to make lives easier. (closes issue #10954, + suggested by jtodd) ........ + +2007-10-15 16:41 +0000 [r85578] Joshua Colp <jcolp@digium.com> + + * /, configs/features.conf.sample: Merged revisions 85571 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85571 | file | 2007-10-15 13:39:59 -0300 (Mon, 15 Oct 2007) | 4 + lines Document that DTMF based features only work when two + channels are bridged together. (closes issue #10773) Reported by: + pbayley ........ + +2007-10-15 16:36 +0000 [r85562] Russell Bryant <russell@digium.com> + + * /, include/asterisk/strings.h: Merged revisions 85561 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85561 | russell | 2007-10-15 11:34:13 -0500 (Mon, 15 Oct 2007) | + 4 lines Make a few changes so that characters in the upper half + of the ISO-8859-1 character set don't get stripped when reading + configuration. (closes issue #10982, dandre) ........ + +2007-10-15 16:23 +0000 [r85560] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 85559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85559 | file | 2007-10-15 13:22:02 -0300 (Mon, 15 Oct 2007) | 4 + lines Bring both DTMF begin and end frames up through to the core + for DTMF feature handling. (closes issue #10826) Reported by: + dimas ........ + +2007-10-15 15:55 +0000 [r85557-85558] Russell Bryant <russell@digium.com> + + * pbx/dundi-parser.c: Simplify buffer handling in dundi-parser.c. + This also makes the code a bit safer by removing various + assumptions about sizes. (No vulnerabilities, though) (closes + issue #10977) Reported by: dimas Patches: dundiparser.patch + uploaded by dimas (license 88) + + * /, pbx/pbx_dundi.c: Merged revisions 85556 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85556 | russell | 2007-10-15 10:40:45 -0500 (Mon, 15 Oct 2007) | + 9 lines Ensure the buffer passed to ast_canmatch_extension() is + properly initialized so that it is null terminated. (issue + #10977) Reported by: dimas Patches: pbxdundi.patch uploaded by + dimas (license 88) - small mods by me ........ + +2007-10-15 15:26 +0000 [r85555] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_jingle.c: Allow RTP structure registration + +2007-10-15 15:07 +0000 [r85553-85554] Joshua Colp <jcolp@digium.com> + + * main/frame.c: Add packetization data for G.722. (closes issue + #10900) Reported by: andrew Patches: frame.diff uploaded by + andrew (license 240) + + * /, main/rtp.c: Merged revisions 85552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85552 | file | 2007-10-15 11:55:04 -0300 (Mon, 15 Oct 2007) | 4 + lines If Monitor or a spy was added to a P2P or native bridged + channel bring the channel back to the generic bridging core so + the monitor or spy operations work. (closes issue #10943) + Reported by: julianjm ........ + +2007-10-15 13:51 +0000 [r85551] Philippe Sultan <philippe.sultan@gmail.com> + + * res/res_jabber.c: Allocate more space for the base64 output we + need to generate. Closes issue #10913, reported by tootai, who + graciously granted us access to his Asterisk server, thanks! + Daniel, feel free to reopen the bug in case you can reproduce + this on 1.4. + +2007-10-15 13:44 +0000 [r85539-85550] Russell Bryant <russell@digium.com> + + * main/cli.c: Move the CLI commands that were in builtins[] into + the cli_cli[] array of CLI commands and remove the cli_iterator + struct. This gets tab completion working again. (closes issue + #10970) Reported by: jamesgolovich Patches: + asterisk-clicomplete.diff.txt uploaded by jamesgolovich (license + 176) + + * /: Blocked revisions 85548 via svnmerge ........ r85548 | russell + | 2007-10-15 08:16:23 -0500 (Mon, 15 Oct 2007) | 7 lines Suppress + a LOG_DEBUG message if debug is not enabled. (closes issue + #10980) Reported by: casper Patches: db.c.84633.diff uploaded by + casper (license 55) ........ + + * doc/tex/jitterbuffer.tex, doc/tex/extensions.tex, + doc/tex/channelvariables.tex, doc/tex/ael.tex, + doc/tex/queues-with-callback-members.tex, doc/tex/realtime.tex, + doc/tex/dundi.tex, doc/tex/security.tex, + doc/tex/configuration.tex, doc/tex/ajam.tex, + doc/tex/cliprompt.tex, doc/tex/manager.tex, doc/tex/misdn.tex, + doc/tex/imapstorage.tex, doc/tex/privacy.tex, doc/tex/sla.tex, + doc/tex/app-sms.tex, doc/tex/billing.tex, apps/app_zapateller.c, + doc/tex/localchannel.tex, doc/tex/cdrdriver.tex, + doc/tex/queuelog.tex: Another major doc directory update from + IgorG. This patch includes - Many uses of the astlisting + environment around verbatim text to ensure that it gets properly + formatted and doesn't run off the page. - Update some things that + have been deprecated. - Add escaping as needed - and more ... + (closes issue #10978) Reported by: IgorG Patches: + texdoc-85542-1.patch uploaded by IgorG (license 20) + + * /, main/asterisk.c: Merged revisions 85545 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85545 | russell | 2007-10-15 08:05:45 -0500 (Mon, 15 Oct 2007) | + 7 lines Make sure remote consoles unmute themselves again after + reconnecting. (closes issue #10847) Reported by: atis Patches: + console_unmute_on_reconnect.patch uploaded by atis (license 242) + ........ + + * /, main/utils.c: Merged revisions 85543 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85543 | russell | 2007-10-15 07:48:10 -0500 (Mon, 15 Oct 2007) | + 8 lines Make sure that the base64 decoder returns a terminated + string. (closes issue #10979) Reported by: ys Patches: + util.c.diff uploaded by ys (license 281) - small mods by me + ........ + + * configure, configure.ac: Change the configure script to check for + a function that was recently added to libss7. + + * /, pbx/pbx_config.c: Merged revisions 85540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85540 | russell | 2007-10-14 10:24:52 -0500 (Sun, 14 Oct 2007) | + 7 lines Don't create the context for users in users.conf until we + know at least one user exists. (closes issue #10971) Reported by: + dimas Patches: pbxconfig.patch uploaded by dimas (license 88) + ........ + + * doc/tex/backtrace.tex (added): When merging the last + documentation update, I forgot to "svn add" a file. Here it is. + (closes issue #10962) + +2007-10-13 15:34 +0000 [r85537] Tilghman Lesher <tlesher@digium.com> + + * /: Blocked revisions 85536 via svnmerge ........ r85536 | + tilghman | 2007-10-13 10:26:01 -0500 (Sat, 13 Oct 2007) | 4 lines + Remove deprecated syntax from sample ael file Reported and + patched by: dimas Closes issue #10967 ........ + +2007-10-13 08:38 +0000 [r85535] James Golovich <james@gnuinter.net> + + * main/cli.c: Fix compiling cli.c due to differences with new cli + system (closes issue 0010966) + +2007-10-13 05:53 +0000 [r85534] Russell Bryant <russell@digium.com> + + * include/asterisk/logger.h, /, main/asterisk.c, main/cli.c: Merged + revisions 85533 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85533 | russell | 2007-10-13 01:48:10 -0400 (Sat, 13 Oct 2007) | + 12 lines Fix an issue with console verbosity when running + asterisk -rx to execute a command and retrieve its output. The + issue was that there was no way for the main Asterisk process to + know that the remote console was connecting in the -rx mode. The + way that James has fixed this is to have all remote consoles + muted by default. Then, regular remote consoles automatically + execute a CLI command to unmute themselves when they first start + up. (closes issue #10847) Reported by: atis Patches: + asterisk-consolemute.diff.txt uploaded by jamesgolovich (license + 176) ........ + +2007-10-12 20:06 +0000 [r85527] Mark Michelson <mmichelson@digium.com> + + * configs/queues.conf.sample, apps/app_queue.c: Allow for the + position announcement to be turned off if desired. (closes issue + #8515, reported by bruno_rocha, initial patch by bruno_rocha, + final patch by qwell) + +2007-10-12 19:41 +0000 [r85525-85526] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c, doc/tex/channelvariables.tex: Trying to + finish the last of the charge_number patch up #10916 + + * channels/chan_zap.c: Add support for receive charge number in + dialplan #10916 + +2007-10-12 18:37 +0000 [r85522-85524] Tilghman Lesher <tlesher@digium.com> + + * doc/asterisk-mib.txt, doc/PEERING, /, LICENSE: Merged revisions + 85523 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85523 | tilghman | 2007-10-12 13:30:55 -0500 (Fri, 12 Oct 2007) + | 2 lines Change Digium address ........ + + * funcs/func_strings.c: Enable ranges, hexadecimal, octal, and + special backslashed characters for the FILTER function + +2007-10-12 15:50 +0000 [r85516-85519] Russell Bryant <russell@digium.com> + + * doc/tex/odbcstorage.tex, doc/tex/extensions.tex, + doc/tex/channelvariables.tex, doc/tex/ael.tex, + doc/tex/queues-with-callback-members.tex, doc/tex/dundi.tex, + doc/tex/enum.tex, doc/tex/cliprompt.tex, doc/tex/manager.tex, + doc/tex/privacy.tex, doc/tex/sla.tex, doc/tex/app-sms.tex, + doc/tex/localchannel.tex, doc/tex/ices.tex, + doc/tex/cdrdriver.tex, doc/tex/asterisk.tex: Many doc directory + improvements, including: - Added development section + (backtrace.tex) - Correct filesystem path formating - Replace all + "|" argument separator to "," - Endless count of spaces at the + end of line - Using astlisting to make listings do not take so + much place - Take back ASTRISKVERSION on first page - Make + localchannel.tex readable by inserting extra end of lines (closes + issue #10962) Reported by: IgorG Patches: texdoc-85177-1.patch + uploaded by IgorG (license 20) + + * res/res_smdi.c, /: Merged revisions 85517 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85517 | russell | 2007-10-12 10:45:09 -0500 (Fri, 12 Oct 2007) | + 3 lines Fix a spelling error in a log message. SMDI, not SDMI. + (closes issue #10959) ........ + + * /, pbx/pbx_realtime.c: Merged revisions 85515 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85515 | russell | 2007-10-12 10:40:35 -0500 (Fri, 12 Oct 2007) | + 7 lines Fix the potential use of an uninitialized buffer in a log + message. (closes issue #10958) Reported by: dimas Patches: + realtime.patch uploaded by dimas (license 88) ........ + +2007-10-11 22:42 +0000 [r85474-85499] Matthew Fredrickson <creslin@digium.com> + + * apps/app_dial.c: Make sure we propogate ANI2 to the outbound + channel + + * funcs/func_callerid.c: See if I can fix this borked ANI2 code I + added + + * channels/chan_zap.c: Make sure we set the ANI2 field for PRI + + * funcs/func_callerid.c: Add ANI2 support to func_callerid + + * channels/chan_zap.c: Add SS7 ANI2 support tx and rx. #10916 + + * channels/chan_zap.c: Add CCR test support #10916 + +2007-10-11 19:03 +0000 [r85460] Russell Bryant <russell@digium.com> + + * main/udptl.c, main/threadstorage.c, res/res_limit.c, + main/translate.c, res/res_crypto.c, res/res_convert.c, + channels/iax2-provision.c, channels/chan_gtalk.c, + channels/chan_oss.c, main/astobj2.c, main/cli.c, main/cdr.c, + main/channel.c, apps/app_osplookup.c, channels/chan_skinny.c, + pbx/pbx_ael.c, main/file.c, pbx/pbx_dundi.c, main/image.c, + pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_rpt.c, + main/asterisk.c, main/db.c, channels/chan_mgcp.c, + res/res_clioriginate.c: Merge a ton of NEW_CLI conversions. + Thanks to everyone that helped out! :) (closes issue #10724) + Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel + (license 64) chan_oss.c.patch uploaded by eliel (license 64) + chan_mgcp.c.patch2 uploaded by eliel (license 64) + pbx_config.c.patch uploaded by seanbright (license 71) + iax2-provision.c.patch uploaded by eliel (license 64) + chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch + uploaded by seanbright (license 71) file.c.patch uploaded by + seanbright (license 71) image.c.patch uploaded by seanbright + (license 71) cli.c.patch uploaded by moy (license 222) + astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch + uploaded by moy (license 222) res_limit.c.patch uploaded by + seanbright (license 71) res_convert.c.patch uploaded by + seanbright (license 71) res_crypto.c.patch uploaded by seanbright + (license 71) app_osplookup.c.patch uploaded by seanbright + (license 71) app_rpt.c.patch uploaded by seanbright (license 71) + app_mixmonitor.c.patch uploaded by seanbright (license 71) + channel.c.patch uploaded by seanbright (license 71) + translate.c.patch uploaded by seanbright (license 71) + udptl.c.patch uploaded by seanbright (license 71) + threadstorage.c.patch uploaded by seanbright (license 71) + db.c.patch uploaded by seanbright (license 71) cdr.c.patch + uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy + (license 222) app_osplookup-rev83558.patch uploaded by moy + (license 222) res_clioriginate.c.patch uploaded by moy (license + 222) + +2007-10-11 17:17 +0000 [r85431-85444] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Let's hard code this until I fix it + + * channels/chan_zap.c: Make sure we are clean to build without + libpri + +2007-10-11 15:29 +0000 [r85398] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 85397 via svnmerge ........ r85397 | file | + 2007-10-11 12:26:20 -0300 (Thu, 11 Oct 2007) | 6 lines When + creating a new packet don't try to stop retransmission of it. It + was just allocated/created so it's impossible for it to have + already been scheduled. (closes issue #10945) Reported by: + flefoll Patches: chan_sip.c.br14.85280.xmit_reliable-patch + uploaded by flefoll (license 244) ........ + +2007-10-11 04:40 +0000 [r85357] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 85356 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85356 | tilghman | 2007-10-10 23:35:33 -0500 (Wed, 10 Oct 2007) + | 2 lines A dollar sign by itself, not indicating a start of a + variable or expression prematurely ends substitution (closes + issue #10939) ........ + +2007-10-10 16:01 +0000 [r85317] Russell Bryant <russell@digium.com> + + * include/asterisk/file.h, /: Merged revisions 85316 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r85316 | russell | 2007-10-10 10:56:23 -0500 (Wed, 10 + Oct 2007) | 6 lines I introduced a new member to the + ast_filestream struct in 1.4.12, but put it in the middle of the + struct, instead of at the end. One of the Debian folks, paravoid, + pointed out that this breaks binary compatability with modules + compiled against older headers. So, I'm moving the new member to + the end of the struct to resolve the situation. ........ + +2007-10-10 14:43 +0000 [r85281] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 85280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85280 | file | 2007-10-10 11:42:00 -0300 (Wed, 10 Oct 2007) | 4 + lines If devicestate is passed a port number strip it out. + (closes issue #10930) Reported by: ibc ........ + +2007-10-10 14:38 +0000 [r85279] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 85276 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85276 | mmichelson | 2007-10-10 09:26:31 -0500 (Wed, 10 Oct + 2007) | 5 lines A bunch of changes from sprintf to snprintf. See + security advisory AST-2002-022 ........ + +2007-10-10 14:30 +0000 [r85234-85278] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 85277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85277 | file | 2007-10-10 11:28:18 -0300 (Wed, 10 Oct 2007) | 6 + lines Add support for handling a 182 Queued response. (closes + issue #10924) Reported by: ramonpeek Patches: queued-182.diff + uploaded by ramonpeek (license 266) ........ + + * /, apps/app_voicemail.c: Merged revisions 85242 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85242 | file | 2007-10-10 11:14:56 -0300 (Wed, 10 Oct 2007) | 6 + lines Close voicemail message description file if duration did + not meet the minimum, or else we will eventually run out of file + descriptors. (closes issue #10918) Reported by: brak2718 Patches: + vm1.4.12.1.patch uploaded by brak2718 (license 279) ........ + + * main/logger.c: Process outstanding log messages before shutting + down the logger thread. (closes issue #10933) Reported by: + sperreault + +2007-10-10 06:48 +0000 [r85197] Luigi Rizzo <rizzo@icir.org> + + * bootstrap.sh: Adapt the autotools names to different versions of + FreeBSD (and open the way to better adaptation for other + platforms as well). + +2007-10-10 06:41 +0000 [r85196] Kevin P. Fleming <kpfleming@digium.com> + + * /, include/asterisk/frame.h: Merged revisions 85195 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r85195 | kpfleming | 2007-10-10 08:24:41 +0200 (Wed, 10 + Oct 2007) | 2 lines use a macro instead of an inline function, so + that backtraces will report the caller of ast_frame_free() + properly ........ + +2007-10-09 22:35 +0000 [r85177] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Patch to add one-touch parking for queues. + (closes issue #10869, reported and patched by bluecrow76) + +2007-10-09 22:21 +0000 [r85140-85176] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /, main/utils.c, include/asterisk/lock.h: Merged + revisions 85158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85158 | tilghman | 2007-10-09 16:55:06 -0500 (Tue, 09 Oct 2007) + | 5 lines This commit fixes the following issues: - Deadlock in + ast_write (issue #10406) - Deadlock in ast_read (issue #10406) - + Possible mutex initialization error in lock.h (issue #10571) + ........ + + * apps/app_dial.c, channels/chan_jingle.c, channels/chan_misdn.c, + apps/app_festival.c, apps/app_minivm.c, apps/app_zapras.c, + utils/astman.c, apps/app_adsiprog.c, utils/check_expr.c: Remove + redundant includes (patch by snuffy) (Closes issue #10922) + +2007-10-09 15:12 +0000 [r85097-85098] Russell Bryant <russell@digium.com> + + * CHANGES: Note jitterbuffer support for chan_local in CHANGES + + * channels/chan_local.c, doc/tex/localchannel.tex: Add jitterbuffer + support for chan_local. To enable it, you use the 'j' option in + the Dial command. The 'j' option _must_ be used in conjunction + with the 'n' option. This feature will allow you to use the + existing jitterbuffer implementation to put a jitterbuffer on + incoming SIP calls connecting to Asterisk applications by putting + a local channel in the middle. + +2007-10-09 14:31 +0000 [r84991-85094] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 85093 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85093 | file | 2007-10-09 11:30:16 -0300 (Tue, 09 Oct 2007) | 4 + lines Don't perform a reinvite if a transfer is in progress. + (issue #10915) Reported by: ramonpeek ........ + + * /, main/rtp.c: Merged revisions 85057 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85057 | file | 2007-10-08 17:06:33 -0300 (Mon, 08 Oct 2007) | 4 + lines Only update codec information if the channel has a + technology private structure. (issue #10915) Reported by: + ramonpeek ........ + + * res/res_limit.c, utils/hashtest2.c, utils/conf2ael.c, + main/ast_expr2.c, utils/check_expr.c: Fix up tree so that it + compiles when MTX Profiling is enabled. (closes issue #10898) + Reported by: snuffy Patches: 10898-mtx_prof.diff uploaded by + qwell (license 4) + + * /, main/rtp.c: Merged revisions 85023 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r85023 | file | 2007-10-08 12:37:46 -0300 (Mon, 08 Oct 2007) | 4 + lines Update codec information as well as address when doing hold + reinvites. (issue #10868) Reported by: mavince ........ + + * main/channel.c, /: Merged revisions 84990 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84990 | file | 2007-10-08 12:03:07 -0300 (Mon, 08 Oct 2007) | 4 + lines Don't keep trying to native bridge if either of the + channels are involved in a masquerade operation to be done. + (closes issue #10696) Reported by: tbelder ........ + +2007-10-08 03:29 +0000 [r84958] Russell Bryant <russell@digium.com> + + * /, Makefile.rules: Merged revisions 84957 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84957 | russell | 2007-10-07 22:28:34 -0500 (Sun, 07 Oct 2007) | + 6 lines Enable file dependency tracking for _all_ builds, and not + just for builds with dev-mode enabled. I have seen enough + problems caused by this that I don't think it's worth keeping. I + want to continue to encourage anybody that is interested to + continue to run Asterisk from svn. Furthermore, I do not want + their systems to break when we change a structure definition in a + header file. :) ........ + +2007-10-07 16:28 +0000 [r84891-84939] Philippe Sultan <philippe.sultan@gmail.com> + + * configs/jabber.conf.sample, include/asterisk/jabber.h, + res/res_jabber.c: Make the status and priority configurable. + Closes issue #10785, patch by Luke-Jr, thanks! + + * /, res/res_jabber.c: Merged revisions 84902 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84902 | phsultan | 2007-10-07 18:15:39 +0200 (Sun, 07 Oct 2007) + | 5 lines Presence packets from a client who's connected with our + Jabber ID are valid, therefore, those clients must be considered + as buddies. The resource string helps us make the distinction + between clients. Closes issue #10707, reported by yusufmotiwala. + ........ + + * res/res_jabber.c: Fix indentation + + * /, res/res_jabber.c: Merged revisions 84890 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84890 | phsultan | 2007-10-07 17:52:44 +0200 (Sun, 07 Oct 2007) + | 5 lines Prevent Asterisk from crashing when receiving a + presence packet without resource from a buddy that is known to + have a resource list. Revert a change I previously made, where + Asterisk could point to a freed memory location. ........ + +2007-10-05 19:48 +0000 [r84852] Tilghman Lesher <tlesher@digium.com> + + * /, main/db.c: Merged revisions 84851 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84851 | tilghman | 2007-10-05 14:42:21 -0500 (Fri, 05 Oct 2007) + | 2 lines Log exactly why we can't open the database, if we fail + (closes issue #10887) ........ + +2007-10-05 18:57 +0000 [r84819] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 84818 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84818 | file | 2007-10-05 15:55:36 -0300 (Fri, 05 Oct 2007) | 4 + lines Update the remembered RTP peer information when putting an + endpoint on hold or taking it off hold so that the RTP stack does + not initiate a needless reinvite. (closes issue #10868) Reported + by: mavince ........ + +2007-10-05 16:49 +0000 [r84743-84784] Russell Bryant <russell@digium.com> + + * channels/chan_zap.c, /: Merged revisions 84783 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84783 | russell | 2007-10-05 11:44:21 -0500 (Fri, 05 Oct 2007) | + 4 lines Do deadlock avoidance in a couple more places. You can't + lock two channels at the same time without doing extra work to + make sure it succeeds. (closes issue #10895, patch by me) + ........ + + * main/manager.c, /: Merged revisions 84742 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84742 | russell | 2007-10-04 20:39:07 -0500 (Thu, 04 Oct 2007) | + 3 lines Fix a copy/paste error in the description of UpdateConfig + that was pointed out by JerJer on #asterisk-dev ........ + +2007-10-04 22:58 +0000 [r84693-84726] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: A two-in-one patch from the bugtracker 1) Fix + some bad logic in the counting of statistics for QueueSummary + manager event. Variables were not being reset for each additional + queue, so cumulative totals were reported on each successive + queue. 2) Add a longest hold time stat to QueueSummary manager + event. + + * /, apps/app_queue.c: Merged revisions 84692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84692 | mmichelson | 2007-10-04 16:57:03 -0500 (Thu, 04 Oct + 2007) | 5 lines Don't allocate space for queue members unless + it's needed. You end up deleting dynamic members on a reload. Not + good. closes issue (#10879, reported by dazza76, patched by me) + ........ + +2007-10-04 21:38 +0000 [r84691] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, /: Merged revisions 84690 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84690 | kpfleming | 2007-10-04 16:36:56 -0500 (Thu, 04 Oct 2007) + | 2 lines callers of sig2str already add the word 'signalling' in + the appropriate place, so don't duplicate it ........ + +2007-10-04 16:56 +0000 [r84671] Tilghman Lesher <tlesher@digium.com> + + * res/res_jabber.c: Update to current coding standards, also + changing the argument delimiter to ',' (Closes issue #10876) + +2007-10-04 14:54 +0000 [r84613-84638] Joshua Colp <jcolp@digium.com> + + * /, apps/app_queue.c: Merged revisions 84637 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84637 | file | 2007-10-04 11:51:57 -0300 (Thu, 04 Oct 2007) | 4 + lines Create a duplicate of the channel's member name as the tab + completion stuff will free it. (closes issue #10884) Reported by: + adamg ........ + + * main/pbx.c: Don't register the exception function with module + information. Since it is in the core there is none and it will + explode. + +2007-10-03 23:05 +0000 [r84580-84582] Tilghman Lesher <tlesher@digium.com> + + * /, main/rtp.c: Merged revisions 84581 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84581 | tilghman | 2007-10-03 17:59:17 -0500 (Wed, 03 Oct 2007) + | 2 lines When an RFC 2833 event is sent that we don't recognize, + ignore it, don't queue a NULL digit (closes issue #10877) + ........ + + * main/pbx.c, doc/tex/extensions.tex, include/asterisk/pbx.h: + Create a universal exception handling extension, "e" (closes + issue #9785) + +2007-10-03 18:23 +0000 [r84512-84545] Steve Murphy <murf@digium.com> + + * /: blocked 84544 from trunk; it only applies to 1.4; 10870 -- the + CUT in AEL + + * res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest17, /, + pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, + pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5, + pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, + pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 84511 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84511 | murf | 2007-10-03 08:23:00 -0600 (Wed, 03 Oct 2007) | 1 + line closes issue #10834 ; where a null input to a switch + statement results in a hangup; since switch is implemented with + extensions, and the default case is implemented with a '.', and + the '.' matches 1 or more remaining characters, the case where 0 + characters exist isn't matched, and the extension isn't matched, + and the goto fails, and a hangup occurs. Now, when a default case + is generated, it also generates a single fixed extension that + will match a null input. That extension just does a goto to the + default extension for that switch. I played with an alternate + solution, where I just tack an extra char onto all the patterns + and the goto, but not the default case's pattern. Then even a + null input will still have at least one char in it. But it made + me nervous, having that extra char in , even if that's a pretty + secret and low-level issue. ........ + +2007-10-02 20:07 +0000 [r84475] Russell Bryant <russell@digium.com> + + * Makefile, /, build_tools/prep_tarball: Merged revisions 84474 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84474 | russell | 2007-10-02 15:06:07 -0500 (Tue, 02 Oct 2007) | + 5 lines * Don't build the menuselect-tree for the tarball, as it + requires running the configure script first * Change the Makefile + to note that menuselect-tree depends on the configure script. + ........ + +2007-10-02 19:02 +0000 [r84432-84440] Jason Parker <jparker@digium.com> + + * /: Blocked revisions 84437 via svnmerge ........ r84437 | qwell | + 2007-10-02 14:01:59 -0500 (Tue, 02 Oct 2007) | 1 line Fix some + odd formatting I missed.. ........ + + * /, res/res_features.c: Merged revisions 84410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #10821) ........ r84410 | qwell | 2007-10-02 13:52:55 -0500 + (Tue, 02 Oct 2007) | 4 lines Finish up on transferee channel + before return on failure. Issue 10821, patch by Ivan ........ + +2007-10-02 18:12 +0000 [r84405] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c: Add MSet for people who prefer the old, deprecated + syntax of Set (Closes issue #10549) + +2007-10-02 14:13 +0000 [r84371] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 84370 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84370 | russell | 2007-10-02 09:12:35 -0500 (Tue, 02 Oct 2007) | + 6 lines Use snprintf instead of sprintf in one place. There is no + vulnerability here due to various buffer sizes around the code, + but I still didn't like seeing a non length-limited copy of data + coming off of the wire into a stack buffer, as this would be a + problem in the future if buffer sizes elsewhere got changed or + size limitations removed ... ........ + +2007-10-02 13:58 +0000 [r84368] Joshua Colp <jcolp@digium.com> + + * main/rtp.c: Don't swap channel priority if using epoll as polling + should/will only happen off the first channel. (closes issue + #10867) Reported by: phsultan + +2007-10-01 23:33 +0000 [r84327-84331] Steve Murphy <murf@digium.com> + + * utils/check_expr.c: OK. THis a DEBUG_THREADS situation. + + * utils/check_expr.c: picky gcc versions... sigh. + + * utils/check_expr.c: This mod will allow check_expr to compile in + the presence of DEBUG_THREAD situations. At least, it does for + me. And it's less expensive than several other approaches I + tried. + + * res/ael/pval.c, /, res/ael/ael.tab.c, res/ael/ael.y, + pbx/pbx_ael.c: Merged revisions 84239 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84239 | murf | 2007-10-01 14:27:52 -0600 (Mon, 01 Oct 2007) | 1 + line closes issue #10777 -- by returning a null for the parse + tree when there's really nothing there, and making sure we don't + try to do checking on a null tree. ........ + +2007-10-01 21:54 +0000 [r84300] Jason Parker <jparker@digium.com> + + * Makefile, /, Makefile.rules, channels/Makefile: Merged revisions + 84291 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84291 | qwell | 2007-10-01 16:52:45 -0500 (Mon, 01 Oct 2007) | 6 + lines Add dist-clean support for subdirs. Change h323 to only + remove the Makefile on a dist-clean, rather than a clean. This + fixes a bug I found with trying to run make after a make clean + ........ + +2007-10-01 21:31 +0000 [r84275] Dwayne M. Hubbard <dhubbard@digium.com> + + * main/channel.c, main/manager.c, /, channels/chan_agent.c: Merged + revisions 84274 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84274 | dhubbard | 2007-10-01 16:25:37 -0500 (Mon, 01 Oct 2007) + | 1 line moved get_base_channel() code from action_redirect to + ast_channel_masquerade() for issue 7706 and BE-160 ........ + +2007-10-01 21:15 +0000 [r84207-84272] Russell Bryant <russell@digium.com> + + * /, main/utils.c, include/asterisk/lock.h: Merged revisions 84271 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84271 | russell | 2007-10-01 16:07:06 -0500 (Mon, 01 Oct 2007) | + 4 lines Fulfull a feature request from Qwell on the "core show + locks" output. It will now note the lock type for each lock that + a thread holds. (mutex, rdlock, or wrlock) ........ + + * /, res/res_agi.c: Merged revisions 84236 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84236 | russell | 2007-10-01 14:56:28 -0500 (Mon, 01 Oct 2007) | + 5 lines Add another sanity check in the AGI read loop. We really + don't care about EAGAIN unless we didn't read an entire line. If + there is a newline at the end if the read buffer, break, because + we got the whole thing. (reported and patched by bmd) ........ + + * /, include/asterisk/lock.h: Merged revisions 84206 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r84206 | russell | 2007-10-01 14:34:12 -0500 (Mon, 01 + Oct 2007) | 2 lines Show rwlocks in the "core show locks" output. + Before, it only showed mutexes. ........ + +2007-10-01 15:57 +0000 [r84176] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Check to make sure a structure pointer is + non-NULL before touching it... crashing is bad, mmmk? (closes + issue #10831) Reported by: eliel Patches: chan_sip.c.patch + uploaded by eliel (license 64) + +2007-10-01 15:34 +0000 [r84167-84174] Russell Bryant <russell@digium.com> + + * main/say.c: Change simple uses of snprintf to ast_copy_string. + This was provided by mvanbaak as a part of issue #10843, but this + part didn't apply because of a patch I applied right beforehand. + + * channels/chan_misdn.c, main/frame.c, res/res_config_odbc.c, + apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c, + main/say.c, apps/app_minivm.c, pbx/dundi-parser.c, + channels/chan_iax2.c, channels/iax2-parser.c, main/asterisk.c, + main/rtp.c, channels/chan_mgcp.c: Corydon posted this janitor + project to the bug tracker and mvanbaak provided a patch for it. + It replaces a bunch of simple calls to snprintf with + ast_copy_string (closes issue #10843) Reported by: Corydon76 + Patches: 2007092900_10843.diff uploaded by mvanbaak (license 7) + + * main/say.c: Simplify code by using the -= and %= operators. + (closes issue #10848) Reported by: opticron Patches: saymod.diff + uploaded by opticron (license 267) + + * codecs/g722/Makefile, /, res/Makefile, channels/Makefile: The + trunk version of this patch also includes a couple more small + clean fixes from IgorG. Merged revisions 84170 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84170 | russell | 2007-10-01 10:00:56 -0500 (Mon, 01 Oct 2007) | + 3 lines Remove another file in "make clean". (closes issue + #10814, paravoid) ........ + + * main/cli.c: Don't set the full command string until after + verifying that there is not another CLI command with the same + command text registered. This prevents a crash if someone + accidentally calls ast_cli_register() on the same CLI command + data twice. This also fixes a small bug where the helpers list + would get unlocked without being locked if building the full + command failed. (closes issue #10858, reported by jamesgolovich, + patched by me) + + * configs/musiconhold.conf.sample, res/res_musiconhold.c: Add a new + option for files-based music on hold to ensure that the sort + order of the files is alphabetical. (closes issue #10855) + Reported by: jamesgolovich Patches: + asterisk-mohsortalpha.diff.txt uploaded by jamesgolovich (license + 176) + + * apps/app_dial.c, /: Merged revisions 84166 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84166 | russell | 2007-10-01 09:24:49 -0500 (Mon, 01 Oct 2007) | + 2 lines Simplify the CAN_EARLY_BRIDGE macro a bit. ........ + +2007-10-01 14:21 +0000 [r84159-84165] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Add MP4 to part of the SDP code. (closes + issue #10820) Reported by: ruikubo Patches: chan_sip.patch + uploaded by ruikubo (license 250) + + * /: Blocked revisions 84163 via svnmerge ........ r84163 | file | + 2007-10-01 11:10:47 -0300 (Mon, 01 Oct 2007) | 4 lines Remove + chan_usbradio config file from tree, it is not present in here. + (closes issue #10839) Reported by: casper ........ + + * main/dnsmgr.c: Don't register the dnsmgr refresh CLI command + twice. (closes issue #10856) Reported by: jamesgolovich Patches: + asterisk-dnsmgrclireg.diff.txt uploaded by jamesgolovich (license + 176) + + * /, res/res_musiconhold.c: Merged revisions 84160 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r84160 | file | 2007-10-01 10:57:42 -0300 (Mon, 01 Oct + 2007) | 6 lines Fix randomness. save_pos was being set to 0 + initially instead of -1, causing it to jump to position 0 when + moh started. (closes issue #10859) Reported by: jamesgolovich + Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich + (license 176) ........ + + * apps/app_dial.c, /: Merged revisions 84158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84158 | file | 2007-10-01 10:49:36 -0300 (Mon, 01 Oct 2007) | 4 + lines Only attempt early bridging if the options given to Dial() + permit it. (closes issue #10861) Reported by: peekyb ........ + +2007-09-30 20:06 +0000 [r84143-84147] Russell Bryant <russell@digium.com> + + * /, include/asterisk/module.h: Merged revisions 84146 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r84146 | russell | 2007-09-30 16:02:16 -0400 (Sun, 30 + Sep 2007) | 4 lines Fix the AST_MODULE_INFO macro for C++ + modules. The load and reload parameters were in the wrong place. + (closes issue #10846, alebm) ........ + + * funcs/func_lock.c: * The documentation for the LOCK() function + says that it will block for up to 3 seconds while waiting on a + lock when other locks are currently held to avoid deadlocks. + Change the code to reflect this. * Since trying to grab a lock + may block for some time, put the channel in autoservice so that + audio is still read from the channel and that any active + generators on the channel don't pause. + +2007-09-29 23:47 +0000 [r84134-84137] Steve Murphy <murf@digium.com> + + * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 84133 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84133 | murf | 2007-09-29 15:47:53 -0600 (Sat, 29 Sep 2007) | 1 + line This issue sort of closes 10786; All config files support + #include with globbing (you know, *,[chars],?,{list,list},etc), + so I've updated the AEL system to support this also. ........ + + * pbx/ael/ael-test/ael-ntest22/t2 (added), + pbx/ael/ael-test/ael-ntest22/t3 (added), + pbx/ael/ael-test/ael-ntest22/extensions.ael (added), + pbx/ael/ael-test/ael-ntest22 (added), + pbx/ael/ael-test/ael-ntest22/t1/a.ael (added), + pbx/ael/ael-test/ael-ntest22/t1/b.ael (added), + pbx/ael/ael-test/ael-ntest22/t1/c.ael (added), + pbx/ael/ael-test/ael-ntest22/t2/d.ael (added), + pbx/ael/ael-test/ael-ntest22/t2/e.ael (added), + pbx/ael/ael-test/ael-ntest22/t2/f.ael (added), + pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22 + (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added), + pbx/ael/ael-test/ref.ael-test3, + pbx/ael/ael-test/ael-ntest22/t3/h.ael (added), + pbx/ael/ael-test/ref.ael-test4, + pbx/ael/ael-test/ael-ntest22/t3/i.ael (added), + pbx/ael/ael-test/ael-ntest22/t3/j.ael (added), + pbx/ael/ael-test/ael-ntest22/qq.ael (added), + pbx/ael/ael-test/ael-ntest22/t1 (added): the last commit for AEL + affected a small number of tests. Added a regression test for + glob'd includes + +2007-09-29 18:21 +0000 [r84130] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_manager.c: Set enablecdr at the end of re-reading the + config file (Closes issue #10852) + +2007-09-29 00:19 +0000 [r84115] Matthew Fredrickson <creslin@digium.com> + + * main/translate.c: Let's use process time instead of wall clock + time for show translation + +2007-09-28 14:35 +0000 [r84050-84080] Tilghman Lesher <tlesher@digium.com> + + * configure, configure.ac: Autoconf requires version 2.60, not + 2.59, to process (Closes issue #10842) + + * /, main/say.c: Merged revisions 84078 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84078 | tilghman | 2007-09-28 09:13:47 -0500 (Fri, 28 Sep 2007) + | 2 lines Correct pronunciations of numbers for .nl (Closes issue + #10837) ........ + + * main/channel.c, /: Merged revisions 84049 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r84049 | tilghman | 2007-09-28 00:30:22 -0500 (Fri, 28 Sep 2007) + | 3 lines Avoid a deadlock with ALL of the locks in the + masquerade function, not just the pairs of channels. (Closes + issue #10406) ........ + +2007-09-27 23:18 +0000 [r84019] Dwayne M. Hubbard <dhubbard@digium.com> + + * main/manager.c, /, channels/chan_agent.c, + include/asterisk/channel.h: Merged revisions 84018 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r84018 | dhubbard | 2007-09-27 18:12:25 -0500 (Thu, 27 + Sep 2007) | 1 line if an Agent is redirected, the base channel + should actually be redirected. This was causing multiple issues, + especially issue 7706 and BE-160 ........ + +2007-09-27 00:08 +0000 [r83978-83986] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_alsa.c: Merged revisions 83974 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83974 | kpfleming | 2007-09-26 16:53:03 -0700 (Wed, 26 Sep 2007) + | 2 lines avoid the weird usage of assert() in the ALSA header + files that gcc 4.2 wants to complain about ........ + + * res/ael/ael.tab.c, res/ael/ael.y: deal with more gcc 4.2 const + pointer warnings + +2007-09-27 00:02 +0000 [r83911-83977] Russell Bryant <russell@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 83976 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83976 | russell | 2007-09-26 19:01:29 -0500 (Wed, 26 Sep 2007) | + 1 line remove a todo item that has been completed ........ + + * /, channels/chan_sip.c: Merged revisions 83943 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83943 | russell | 2007-09-26 16:35:23 -0500 (Wed, 26 Sep 2007) | + 2 lines I changed my mind ... I think this should be a + LOG_NOTICE. ........ + + * /, channels/chan_sip.c: Merged revisions 83941 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83941 | russell | 2007-09-26 16:15:15 -0500 (Wed, 26 Sep 2007) | + 5 lines Add a log message that was requested by the masses in the + developer tutorial session at Astricon. chan_sip did not output + any message when a call was rejected because the extension was + not found. This adds a verbose message (at verbose level 3) to + note when this happens. ........ + + * /: Merged revisions 83910 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83910 | russell | 2007-09-26 15:50:09 -0500 (Wed, 26 Sep 2007) | + 3 lines Fix building chan_misdn under dev-mode. (please run the + configure script with --enable-dev-mode so this doesn't happen + again ...) ........ + +2007-09-26 18:43 +0000 [r83880] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_zap.c, /: Merged revisions 83879 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83879 | tilghman | 2007-09-26 13:35:56 -0500 (Wed, 26 Sep 2007) + | 2 lines Remove unused 4k of memory on the program stack (closes + issue #10827) ........ + +2007-09-26 06:53 +0000 [r83849-83864] Russell Bryant <russell@digium.com> + + * include/asterisk/event.h: fix a typo in a comment + + * include/asterisk/file.h: Change function documentation to use + doxygen tags. (Really, I just needed to make some minor change in + trunk to test something with automerge ...) + +2007-09-25 23:14 +0000 [r83834] Matthew Fredrickson <creslin@digium.com> + + * doc/ss7.txt: Fix typo in readme + +2007-09-25 21:06 +0000 [r83819] Russell Bryant <russell@digium.com> + + * include/asterisk/devicestate.h: Don't note that functions are + deprecated in favor of themselves. This was found by showing a + very poor example doxygen function in a presentation this + morning. :) + +2007-09-25 16:34 +0000 [r83804] Philippe Sultan <philippe.sultan@gmail.com> + + * res/res_jabber.c: Added a CLI command that shows our buddy list, + as suggested by Daniel McKeehan, thanks! + +2007-09-25 14:18 +0000 [r83774] Tilghman Lesher <tlesher@digium.com> + + * /, main/app.c: Merged revisions 83773 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83773 | tilghman | 2007-09-25 09:13:25 -0500 (Tue, 25 Sep 2007) + | 2 lines jmls pointed out that unsetting the group and setting + the group to the blank string aren't quite the same. ........ + +2007-09-25 13:41 +0000 [r83758] Joshua Colp <jcolp@digium.com> + + * res/ael/pval.c: Fix minor memory leak in pval.c. Overwriting a + value without freeing the previous result is bad, mmmk? + +2007-09-25 09:07 +0000 [r83743] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_jingle.c, include/asterisk/jingle.h: Comply with + latest XEP-0166, XEP-0167, XEP-0176. No real Jingle + implementation being available, testing was made using two + Asterisk servers relaying SIP calls over their Jingle channels: + SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- + [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Thus, it was + possible to test the code in both ways, and make the Jingle + channel comply with the latest specifications. No sound available + yet. Main modifications include : - modified the + 'jingle_candidate' structure and the 'jingle_create_candidates' + function according to XEP-0176 ; - modified the 'jingle_action' + function in order to properly terminate a Jingle session, in + conformance with XEP-0166 ; - modified username format used in + STUN requests ; - actually make the bindaddr configuration field + useable. Todo : - set audio paths up (no native bridging) ; - + make the CLI gtalk functions available to jingle ; - clean up the + storage space used in strings. + +2007-09-25 08:09 +0000 [r83741] Russell Bryant <russell@digium.com> + + * utils/Makefile, utils: Add some files to the utils directory + svn:ignore and Makefile clean target (closes issue #10808, + reported by mvanbaak) + +2007-09-24 22:06 +0000 [r83696-83726] Tilghman Lesher <tlesher@digium.com> + + * Makefile, main/asterisk.c: Permit custom locations for astdb and + the keys directory (though default to the current locations) + (Closes issue #10267) + + * /, build_tools/make_defaults_h: Merged revisions 83695 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83695 | tilghman | 2007-09-24 12:22:08 -0500 (Mon, 24 Sep 2007) + | 4 lines In the source, keys are relative to the datadir, not + varlib (which is the same in most cases, but it's good to be + accurate). Closes issue #10811 ........ + +2007-09-24 17:10 +0000 [r83671] Dwayne M. Hubbard <dhubbard@digium.com> + + * channels/chan_sip.c, configs/sip.conf.sample: merged jcmoore's + patch for configurable SDP origin-field username and session + field, closes issue# 10795 + +2007-09-24 17:00 +0000 [r83656] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: interface_exists_global was never returning 1. + Most likely an error from my merge on Friday. (closes issue + #10817, reported and patched by snar, patch simplified by me) + +2007-09-24 16:42 +0000 [r83654-83655] Tilghman Lesher <tlesher@digium.com> + + * /: Blocked revisions 83653 via svnmerge ........ r83653 | + tilghman | 2007-09-24 11:37:52 -0500 (Mon, 24 Sep 2007) | 2 lines + Oops. Removed the unworkable workaround. This note should never + have been in the release. ........ + + * /, main/app.c: Merged revisions 83637 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83637 | tilghman | 2007-09-24 10:17:06 -0500 (Mon, 24 Sep 2007) + | 3 lines Making change to group splitting, as discussed on the + -dev list. The main effect of this will be to permit + Set(GROUP([cat])=), i.e. unsetting a group. ........ + +2007-09-22 19:54 +0000 [r83575-83590] Steve Murphy <murf@digium.com> + + * res/ael/pval.c, /: Merged revisions 83589 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83589 | murf | 2007-09-22 13:39:16 -0600 (Sat, 22 Sep 2007) | 1 + line This closes issue #10788 -- The exact same fixes are made + here for the first arg in the for(arg1; arg2; arg3) {} statement, + as were done for the 3rd arg. It can now be an assignment that + will embedded in a Set() app, or a macro call, or an app call. + ........ + + * res/ael/pval.c, /, pbx/pbx_ael.c: Merged revisions 83558 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83558 | murf | 2007-09-22 10:41:43 -0600 (Sat, 22 Sep 2007) | 1 + line This closes issue #10788 -- the 3rd arg in the for statement + is now wrapped in Set() only if there's an '=' in that string. + Otherwise, if it begins with '&', then a Macro call is generated; + otherwise it is made into an app call. A bit more accomodating, + keeps the new guys happy, and the guys with ael-1 code should be + happy, too ........ + +2007-09-22 17:37 +0000 [r83574] Matthew Fredrickson <creslin@digium.com> + + * doc/ss7.txt: Fix potential point of confusion + +2007-09-22 14:45 +0000 [r83517-83545] Tilghman Lesher <tlesher@digium.com> + + * utils/Makefile, utils/hashtest2.c, utils/clicompat.c (added): Fix + build of check_expr and hashtest2 when DEBUG_THREADLOCAL is + defined + + * main/manager.c, apps/app_meetme.c: Add the MeetmeList and Reload + manager commands, which supplement the need to have Command + privilege. (closes issue #10736) + + * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h, + main/ast_expr2.y, configure.ac, main/ast_expr2.c: Fixes for + FreeBSD... testing for every conceivable math function now + +2007-09-21 19:55 +0000 [r83500] Russell Bryant <russell@digium.com> + + * channels/chan_zap.c: Fix compilation errors in CLI command + updates to SS7 CLI commands + +2007-09-21 19:54 +0000 [r83499] Matthew Fredrickson <creslin@digium.com> + + * doc/ss7.txt (added): Add an SS7 readme for setup and use of + libss7 and asterisk + +2007-09-21 18:41 +0000 [r83484] Tilghman Lesher <tlesher@digium.com> + + * apps/app_queue.c: Fix some areas where we were still using '|' + for an argument delimiter (closes issue #10793) + +2007-09-21 18:27 +0000 [r83483] Russell Bryant <russell@digium.com> + + * apps/app_queue.c: Update app_queue to use commas as application + argument separators. (closes issue #10793, snar) + +2007-09-21 17:36 +0000 [r83466] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_manager.c: Fix cdr_manager, such that if the config file + is created past load, it'll start logging (and conversely, if the + config file is destroyed or deactivated, the logging is + disabled). Reported by Juggie via IRC, fix by me. + +2007-09-21 14:40 +0000 [r83433] Russell Bryant <russell@digium.com> + + * res/res_config_pgsql.c, main/dnsmgr.c, /, channels/chan_sip.c, + main/db1-ast/hash/hash.c, include/asterisk/channel.h, + channels/chan_iax2.c, main/rtp.c, channels/misdn_config.c, + main/cdr.c, main/channel.c, channels/chan_misdn.c, + main/ast_expr2f.c, main/file.c, include/asterisk/sched.h, + channels/chan_h323.c, utils/ael_main.c, pbx/pbx_dundi.c, + main/sched.c, channels/chan_mgcp.c, main/ast_expr2.fl: Merged + revisions 83432 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | + 4 lines gcc 4.2 has a new set of warnings dealing with cosnt + pointers. This set of changes gets all of Asterisk (minus + chan_alsa for now) to compile with gcc 4.2. (closes issue #10774, + patch from qwell) ........ + +2007-09-21 14:25 +0000 [r83431] Tilghman Lesher <tlesher@digium.com> + + * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h, + main/ast_expr2.y, configure.ac, main/ast_expr2.c: Check for the + presence of trunc and round, and make the ISOC99 detection a + little more sane (closes issue #10776) + +2007-09-21 13:36 +0000 [r83401] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 83400 via svnmerge ........ r83400 | file | + 2007-09-21 10:34:32 -0300 (Fri, 21 Sep 2007) | 4 lines Fix video + under certain circumstances. It would have been possible for the + formats on the channel to not contain the video format. (closes + issue #10782) Reported by: cwhuang ........ + +2007-09-20 23:14 +0000 [r83381] Jason Parker <jparker@digium.com> + + * apps/app_minivm.c, main/astmm.c, apps/app_playback.c: More + NEW_CLI conversions. (issue #10724) Patches: app_playback.c.patch + uploaded by moy (license 222) app_minivm.c.patch uploaded by + eliel (license 64) astmm.c.patch uploaded by eliel (license 64) + +2007-09-20 21:37 +0000 [r83350-83351] Mark Michelson <mmichelson@digium.com> + + * /: Oops. Getting rid of svnmerge-integrated and automerge stuff + + * /, apps/app_queue.c: Merging changes from queue_refcount_trunk + into trunk. Refcounted queues now in place. + +2007-09-20 21:17 +0000 [r83293-83349] Russell Bryant <russell@digium.com> + + * /, main/asterisk.c: Merged revisions 83348 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83348 | russell | 2007-09-20 16:16:48 -0500 (Thu, 20 Sep 2007) | + 4 lines When daemonizing, don't change working directory to "/". + It makes it not be able to do a core dump when not running as + uid=root. (closes issue #10766, xrg) ........ + + * /, contrib/scripts/safe_asterisk: Merged revisions 83316 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83316 | russell | 2007-09-20 16:01:20 -0500 (Thu, 20 Sep 2007) | + 3 lines Change safe_asterisk to explicitly ask for /bin/bash, as + it uses bashisms. (closes issue #10772, reported by culrich) + ........ + + * main/dsp.c: trivial formatting change + + * main/asterisk.c: trivial formatting change + + * main/app.c: minor spelling fixes in a comment + + * main/app.c: minor grammar fix + + * channels/chan_sip.c: fix spelling in a comment + + * main/asterisk.c: trivial formatting change + +2007-09-20 19:05 +0000 [r83251-83278] Jason Parker <jparker@digium.com> + + * doc/modules.txt: Fix a trivial typo, to test our new commit bot + + * /, apps/app_disa.c: Merged revisions 83246 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83246 | qwell | 2007-09-20 12:09:14 -0500 (Thu, 20 Sep 2007) | 8 + lines If # is pressed after dialing an extension in DISA, stop + trying to collect more digits. (closes issue #10754) Reported by: + atis Patches: app_disa.c.branch.patch uploaded by atis (license + 242) app_disa.c.trunk.patch uploaded by atis (license 242) + ........ + +2007-09-20 16:28 +0000 [r83234] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 83232 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83232 | file | 2007-09-20 13:25:30 -0300 (Thu, 20 Sep 2007) | 7 + lines Make sure the minimum T1 timer value is obeyed in all + cases. (closes issue #10768) Reported by: flefoll Patches: + chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll (license + 244) chan_sip.c.br14.83070.retrans-patch uploaded by flefoll + (license 244) ........ + +2007-09-20 16:27 +0000 [r83233] Russell Bryant <russell@digium.com> + + * main/asterisk.c: Don't start the event processing thread until + after forking. (reported by Simon on the -dev list, thanks!) + +2007-09-20 16:19 +0000 [r83229-83231] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 83230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83230 | file | 2007-09-20 13:17:24 -0300 (Thu, 20 Sep 2007) | 7 + lines Fix a minor spelling error. (closes issue #10769) Reported + by: flefoll Patches: chan_sip.c.trunk.83071.inita-patch uploaded + by flefoll (license 244) chan_sip.c.br14.83070.inita-patch + uploaded by flefoll (license 244) ........ + + * pbx/pbx_dundi.c, cdr/cdr_pgsql.c, main/config.c: Fix memory leaks + in pbx_dundi, cdr_pgsql, and the configuration file parser. + +2007-09-19 23:16 +0000 [r83213] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, apps/app_meetme.c, apps/app_queue.c, + apps/app_voicemail.c: More conversions to NEW_CLI (issue #10724) + Patches: chan_zap.c.patch uploaded by moy (license 222) + app_queue.c.patch uploaded by eliel (license 64) + app_voicemail.c.patch uploaded by eliel (license 64) + app_meetme.c.patch uploaded by eliel (license 64) + +2007-09-19 20:06 +0000 [r83182-83183] Joshua Colp <jcolp@digium.com> + + * cdr/cdr_csv.c: Clean up code in cdr_csv. (Are you sensing a theme + for me today?) + + * res/res_adsi.c: Clean up code in res_adsi. + +2007-09-19 19:54 +0000 [r83176-83181] Russell Bryant <russell@digium.com> + + * funcs/func_shell.c: put the channel in autoservice when executing + func_shell + + * /, apps/app_system.c: Merged revisions 83179 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83179 | russell | 2007-09-19 14:50:48 -0500 (Wed, 19 Sep 2007) | + 5 lines The System() and TrySystem() applications can take a + substantial amount of time to execute while not servicing the + channel. So, put the channel in autoservice while the command is + being executed. (closes issue #10726, reported by mnicholson) + ........ + + * funcs/func_curl.c, /: Merged revisions 83177 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83177 | russell | 2007-09-19 14:34:25 -0500 (Wed, 19 Sep 2007) | + 4 lines Using curl can take a substantial amount of time, so the + channel should be autoserviced while waiting for it to complete. + (closes issue #10725, reported by mnicholson) ........ + + * /, channels/chan_iax2.c: Merged revisions 83175 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83175 | russell | 2007-09-19 14:13:29 -0500 (Wed, 19 Sep 2007) | + 8 lines When handling a reload of chan_iax2, don't use an + ao2_callback() to POKE all peers. Instead, use an iterator. By + using an iterator, the peers container is not locked while the + POKE is being done. It can cause a deadlock if the peers + container is locked because poking a peer will try to lock pvt + structs, while there is a lot of other code that will hold a pvt + lock when trying to go lock the peers container. (reported to me + directly by Loic Didelot. Thank you for the debug info!) ........ + +2007-09-19 17:22 +0000 [r83155-83157] Joshua Colp <jcolp@digium.com> + + * apps/app_db.c: Fix indentation in app_db. + + * apps/app_authenticate.c: Clean up code in app_authenticate. + + * apps/app_adsiprog.c: Clean up code in app_adsiprog. + +2007-09-19 15:11 +0000 [r83126] Russell Bryant <russell@digium.com> + + * main/manager.c, /: Merged revisions 83121 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83121 | russell | 2007-09-19 10:10:14 -0500 (Wed, 19 Sep 2007) | + 4 lines Fix up another potential race condition. Do the loop + decrementing use count on events with the eventq protected from + being changed. (reported on IRC by Ivan) ........ + +2007-09-19 15:08 +0000 [r83105-83114] Joshua Colp <jcolp@digium.com> + + * apps/app_disa.c: DISA only needs to know about the end of DTMF, + not the beginning/duration. + + * apps/app_disa.c: Clean up app_disa code a bit. + +2007-09-19 13:55 +0000 [r83076] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_jingle.c: Replace Google namespace occurrences with + Jingle. The former namespace is handled by chan_gtalk. + +2007-09-19 13:49 +0000 [r83073-83075] Joshua Colp <jcolp@digium.com> + + * /, apps/app_queue.c: Merged revisions 83074 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83074 | file | 2007-09-19 10:47:59 -0300 (Wed, 19 Sep 2007) | 6 + lines Protect the CDR record from modification by pbx_exec so + that the application data contains the Queue data. (closes issue + #10761) Reported by: snar Patches: app-queue-mixmonitor.patch + uploaded by snar (license 245) ........ + + * main/manager.c: Extend manager show connected with additional + information. (closes issue #10757) Reported by: outtolunc + Patches: manager.c.sessionstart.diff uploaded by outtolunc + (license 237) + +2007-09-19 13:29 +0000 [r83072] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_jingle.c: Remove namespaces in payload-type tags. + +2007-09-19 13:21 +0000 [r83071] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 83070 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83070 | file | 2007-09-19 10:18:22 -0300 (Wed, 19 Sep 2007) | 6 + lines (closes issue #10760) Reported by: dimas Patches: + chan_sip.patch uploaded by dimas (license 88) Read in + subscribecontext option in general to be the default. ........ + +2007-09-19 12:23 +0000 [r83055] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_jingle.c, include/asterisk/jingle.h: Transmit + proper invitation, thus conforming to XEP-0166 (Jingle general + specifications), XEP-0167 (Jingle Audio via RTP) and XEP-0176 + (Jingle ICE Transport). + +2007-09-19 09:48 +0000 [r83025] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, + channels/misdn_config.c: Merged revisions 83023-83024 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r83023 | crichter | 2007-09-19 11:31:55 +0200 (Mi, 19 Sep 2007) | + 1 line added 'astdtmf' option to allow configuring the asterisk + dtmf detector instead of the mISDN_dsp ones. also added the patch + from irroot #10190, so that dtmf tones detected by the asterisk + detector are passed outofband to asterisk, to make any use of + dtmf tones at all. ........ r83024 | crichter | 2007-09-19 + 11:32:42 +0200 (Mi, 19 Sep 2007) | 1 line removed comment which + violates the coding guidelines. ........ + +2007-09-19 00:21 +0000 [r82993] Russell Bryant <russell@digium.com> + + * /, apps/app_flash.c: Merged revisions 82992 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82992 | russell | 2007-09-18 19:19:49 -0500 (Tue, 18 Sep 2007) | + 4 lines Change the description of app_flash to note how it can be + a useful tool instead of just saying that it is generally a + worthless feature. (Thanks to Jim Van Meggelen for pointing it + out and providing the proposed text) ........ + +2007-09-18 23:42 +0000 [r82962] Joshua Colp <jcolp@digium.com> + + * /, apps/app_queue.c: Merged revisions 82961 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82961 | file | 2007-09-18 20:41:02 -0300 (Tue, 18 Sep 2007) | 2 + lines Initialize a variable to NULL to make the world happy. + ........ + +2007-09-18 22:46 +0000 [r82931] Russell Bryant <russell@digium.com> + + * include/asterisk/agi.h, /, res/res_agi.c: Merged revisions 82929 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82929 | russell | 2007-09-18 17:42:27 -0500 (Tue, 18 Sep 2007) | + 11 lines Add a new patch to handle interrupting the fgets() call + when using FastAGI. This version of the patch maintains the + original behavior of the code when not using FastAGI. (closes + issue #10553) Reported by: juggie Patches: res_agi_fgets-4.patch + uploaded by juggie (license 24) res_agi_fgets_1.4svn.patch + uploaded by juggie (license 24) Slight mods by me Tested by: + juggie, festr ........ + +2007-09-18 22:43 +0000 [r82871-82930] Jason Parker <jparker@digium.com> + + * main/pbx.c, main/frame.c, main/dnsmgr.c, channels/chan_local.c, + channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, + res/res_musiconhold.c, res/res_jabber.c, main/manager.c, + res/res_agi.c, channels/chan_features.c, main/logger.c, + main/http.c, channels/chan_alsa.c, res/res_realtime.c, + res/res_odbc.c: (issue #10724) Reported by: eliel Patches: + res_features.c.patch uploaded by eliel (license 64) + res_agi.c.patch uploaded by seanbright (license 71) + res_musiconhold.c.patch uploaded by seanbright (license 71) + pbx.c.patch uploaded by moy (license 222) logger.c.patch uploaded + by moy (license 222) frame.c.patch uploaded by moy (license 222) + manager.c.patch uploaded by moy (license 222) http.c.patch + uploaded by moy (license 222) dnsmgr.c.patch uploaded by moy + (license 222) res_realtime.c.patch uploaded by eliel (license 64) + res_odbc.c.patch uploaded by seanbright (license 71) + res_jabber.c.patch uploaded by eliel (license 64) + chan_local.c.patch uploaded by eliel (license 64) + chan_agent.c.patch uploaded by eliel (license 64) + chan_alsa.c.patch uploaded by eliel (license 64) + chan_features.c.patch uploaded by eliel (license 64) + chan_sip.c.patch uploaded by eliel (license 64) RollUp.1.patch + (includes all of the above patches) uploaded by seanbright + (license 71) Convert many CLI commands to the NEW_CLI format. + + * configs/voicemail.conf.sample, apps/app_voicemail.c: (closes + issue #10739) Reported by: ruffle Patches: app_voicemail.c.diff + uploaded by ruffle (license 201) 10739-moveheard.diff uploaded by + qwell (license 4) Tested by: callguy, ruffle Add an option to + disable the automatic moving of "heard" messages to the Old + folder. + +2007-09-18 20:59 +0000 [r82868] Russell Bryant <russell@digium.com> + + * main/manager.c, /: Merged revisions 82867 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82867 | russell | 2007-09-18 15:56:43 -0500 (Tue, 18 Sep 2007) | + 10 lines Fix a memory leak that can occur on systems under higher + load. The issue is that when events are appended to the master + event queue, they use the number of active sessions as a use + count so it will know when all active sessions at the time the + event happened have consumed it. However, the handling of the + number of sessions was not properly synchronized, so the use + count was not always correct, causing an event to disappear + early, or get stuck in the event queue for forever. (closes issue + #9238, reported by bweschke, patch from Ivan, modified by me) + ........ + +2007-09-18 20:10 +0000 [r82866] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 82865 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82865 | mmichelson | 2007-09-18 15:09:02 -0500 (Tue, 18 Sep + 2007) | 4 lines Moving the logic for handling an empty membername + to the create_member function so that there is a common place + where this occurs instead of being spread out to several + different places. ........ + +2007-09-18 19:06 +0000 [r82835] Kevin P. Fleming <kpfleming@digium.com> + + * /, apps/app_queue.c: Merged revisions 82834 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82834 | kpfleming | 2007-09-18 13:59:52 -0500 (Tue, 18 Sep 2007) + | 2 lines there is no need for conditional logic to select + ->interface or ->membername, snince ->membername will always be + populated ........ + +2007-09-18 16:34 +0000 [r82803] Russell Bryant <russell@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 82802 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82802 | russell | 2007-09-18 11:31:01 -0500 (Tue, 18 Sep 2007) | + 4 lines When copying the contents from the wildcard peer, do a + deep copy instead of shallow copy so that it doesn't crash when + beging destroyed. (closes issue #10546, patch by me) ........ + +2007-09-18 16:16 +0000 [r82800] Jason Parker <jparker@digium.com> + + * configs/queues.conf.sample, apps/app_queue.c: (closes issue + #10755) Reported by: snar Patches: app-queue-cdr-trunk.patch + uploaded by snar (license 245) queues.conf.patch uploaded by snar + (license 245) Add an updatecdr option to queues.conf, so that if + a "member name" is specified, the cdr record will be updated with + that, rather than the channel. + +2007-09-18 16:14 +0000 [r82776-82793] Russell Bryant <russell@digium.com> + + * include/asterisk/threadstorage.h: Make sure that libpthread + doesn't try to call free() directly when MALLOC_DEBUG is enabled. + If it does, Asterisk will crash as the address isn't the real + beginning of the allocation. + + * channels/chan_zap.c: Don't use ast_channel_lock_both() here, it + only exists in one of my branches. This is theoretically a + potential deadlock, but it's the way it was before so I'm going + to leave it this way for now. + +2007-09-18 15:29 +0000 [r82752] Jason Parker <jparker@digium.com> + + * /, configs/sip.conf.sample: Merged revisions 82751 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes + issue #10753) ........ r82751 | qwell | 2007-09-18 10:28:21 -0500 + (Tue, 18 Sep 2007) | 4 lines Correct the allowexternaldomains + option in SIP sample config. Issue 10753 ........ + +2007-09-17 22:59 +0000 [r82728] Russell Bryant <russell@digium.com> + + * channels/chan_local.c, channels/chan_zap.c, apps/app_zapscan.c, + channels/chan_agent.c, channels/chan_alsa.c, + channels/chan_iax2.c, channels/chan_mgcp.c: convert various + places that access the channel lock directly to use the channel + lock wrappers + +2007-09-17 21:52 +0000 [r82710-82712] Jason Parker <jparker@digium.com> + + * cdr/cdr_sqlite3_custom.c: Don't try to continue loading + cdr_sqlite3_custom on a module load failure (such as the config + not existing) Closes issue #10749, patch by seanbright. + + * configs/http.conf.sample: Fix the sample redirect to point to a + valid file in the Asterisk GUI. Closes issue #10748, patch by + bkruse + +2007-09-17 20:24 +0000 [r82595-82679] Russell Bryant <russell@digium.com> + + * doc/res_config_sqlite.txt, res/res_config_sqlite.c: Add support + for #include, var_metric, and cat_metric in res_config_sqlite + (closes issue #10738, rbraun_proformatique) + + * /, main/stdtime/localtime.c, apps/app_voicemail.c: Merged + revisions 82676 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82676 | russell | 2007-09-17 15:16:25 -0500 (Mon, 17 Sep 2007) | + 4 lines Put a memset in ast_localtime() instead of a couple + places in app_voicemail to prevent the problem everywhere instead + of just a couple of places. (related to issue #10746) ........ + + * /, apps/app_voicemail.c: Merged revisions 82644 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82644 | russell | 2007-09-17 15:00:32 -0500 (Mon, 17 Sep 2007) | + 6 lines Initialize some memory to fix crashes when leaving + voicemail. This problem was fixed by running Asterisk under + valgrind. (closes issue #10746, reported by arcivanov, patched by + me) *** IMPORTANT NOTE: We need to check to see if this same bug + exists elsewhere. ........ + + * apps/app_dial.c, res/ael/pval.c, include/asterisk/utils.h, + apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c, + res/res_features.c, apps/app_queue.c, channels/chan_iax2.c, + pbx/pbx_config.c: Make the MALLOC_DEBUG output for free() useful + again. After changing calls to free to be ast_free, astmm said + all calls to free were coming from utils.h + + * /, res/res_features.c: Merged revisions 82594 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82594 | russell | 2007-09-17 11:46:59 -0500 (Mon, 17 Sep 2007) | + 5 lines Handle the case where there are multiple dynamic features + with the same digit mapping, but won't always match the activated + on/by access controls. In that case, the code needs to keep + trying features for a match. (reported by Atis on the + asterisk-dev list, patched by me) ........ + +2007-09-17 16:44 +0000 [r82593] Kevin P. Fleming <kpfleming@digium.com> + + * /, apps/app_queue.c: Merged revisions 82590,82592 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r82590 | kpfleming | 2007-09-17 11:33:30 -0500 (Mon, 17 + Sep 2007) | 2 lines fix a couple of places where a logical member + name (if specified) was not used, but instead the direct + interface was listed ........ r82592 | kpfleming | 2007-09-17 + 11:40:12 -0500 (Mon, 17 Sep 2007) | 2 lines revert a change that + wasn't supposed to be committed... doh! ........ + +2007-09-17 14:58 +0000 [r82568] Doug Bailey <dbailey@digium.com> + + * main/http.c: Fix memory leak introduced when POST support was + added. + +2007-09-17 02:20 +0000 [r82516-82546] Joshua Colp <jcolp@digium.com> + + * res/res_features.c: (closes issue #10715) Reported by: + the-chopper Don't bother hanging up the new channel if it does + not exist yet. + + * main/pbx.c, /: Merged revisions 82514 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82514 | file | 2007-09-16 23:00:59 -0300 (Sun, 16 Sep 2007) | 4 + lines (closes issue #10734) Reported by: asgaroth Instead of + passing a NULL pointer into snprintf pass "". It makes Solaris + much happier. ........ + +2007-09-16 15:32 +0000 [r82496] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c: Option maxmessage should be maxsecs + per-folder, too (closes issue #10729) + +2007-09-14 21:30 +0000 [r82457] Steve Murphy <murf@digium.com> + + * main/cdr.c, /: Merged revisions 82444 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82444 | murf | 2007-09-14 15:19:27 -0600 (Fri, 14 Sep 2007) | 1 + line closes issue #10668; thanks to arkadia for his patch; had to + leave out the bit about ending the previous cdr in the fork; it + would destroy current implementations. ........ + +2007-09-14 21:21 +0000 [r82454] Russell Bryant <russell@digium.com> + + * /, configs/zapata.conf.sample: Merged revisions 82435 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82435 | russell | 2007-09-14 16:17:08 -0500 (Fri, 14 Sep 2007) | + 3 lines Add a note to help clarify the value set with the + echocancel option. (inspired by Malcolm's blog post on + blogs.digium.com about HPEC) ........ + +2007-09-14 19:49 +0000 [r82401] Jason Parker <jparker@digium.com> + + * channels/chan_skinny.c, configs/skinny.conf.sample: Add support + in chan_skinny for sending RTP directly to the endpoints. Closes + issue #9154, patch by DEA + +2007-09-14 18:37 +0000 [r82397-82400] Mark Michelson <mmichelson@digium.com> + + * /: Blocking revision 82398 + + * /, apps/app_queue.c: Merged revisions 82396 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82396 | mmichelson | 2007-09-14 13:28:36 -0500 (Fri, 14 Sep + 2007) | 5 lines Adding member name field to manager events where + they were missing before (closes issue #10721, reported by snar) + ........ + +2007-09-14 17:51 +0000 [r82395] Jason Parker <jparker@digium.com> + + * channels/chan_zap.c, /: Merged revisions 82394 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82394 | qwell | 2007-09-14 12:48:05 -0500 (Fri, 14 Sep 2007) | 5 + lines If a channel does not have an owner, do not try to set a + channel variable. This will end up making the channel variable + global, which is not right. Closes issue #10720, patch by + flefoll. ........ + +2007-09-14 17:29 +0000 [r82393] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/res_odbc.h, res/res_odbc.c: Add a direct execute + method to res_odbc (closes issue #10722) + +2007-09-14 16:02 +0000 [r82386-82391] Russell Bryant <russell@digium.com> + + * channels/xpmr/xpmr.h, channels/xpmr/LICENSE (removed), + channels/xpmr/sinetabx.h, channels/xpmr/xpmr.c, + channels/xpmr/xpmr_coef.h: use the standard license header for + the xpmr files + + * channels/chan_usbradio.c (added), channels/xpmr (added): Add + chan_usbradio to trunk + + * /, build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: + Merged revisions 82385 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82385 | russell | 2007-09-14 10:50:49 -0500 (Fri, 14 Sep 2007) | + 3 lines Add checking for libusb here, so nobody has to deal with + conflicts in the chan_usbradio-1.4 branch every time the + configure script gets changed ........ + +2007-09-14 14:44 +0000 [r82377] Mark Michelson <mmichelson@digium.com> + + * doc/CODING-GUIDELINES, /: Merged revisions 82376 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r82376 | mmichelson | 2007-09-14 09:42:29 -0500 (Fri, 14 + Sep 2007) | 5 lines Fixing a typo in the coding guidelines + (closes issue #10717, reported and patched by leedm777) ........ + +2007-09-14 13:02 +0000 [r82373] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_jingle.c: Fix DTMF following what has been done in + issue #9401. Thanks irroot. + +2007-09-13 23:12 +0000 [r82359] Jason Parker <jparker@digium.com> + + * pbx/pbx_spool.c, /: Merged revisions 82358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82358 | qwell | 2007-09-13 18:11:27 -0500 (Thu, 13 Sep 2007) | 4 + lines Fix a small typo. retrytime > waittime ........ + +2007-09-13 21:53 +0000 [r82347-82352] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Changed "in" to "queue" in "queue + {pause|unpause} member" command to be more clear. Also added + check to be sure that sixth argument is the word "reason" if full + command is given + + * CHANGES, apps/app_queue.c: Added the ability to pause and unpause + members via the CLI + + * /, apps/app_queue.c: Merged revisions 82346 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82346 | mmichelson | 2007-09-13 15:16:37 -0500 (Thu, 13 Sep + 2007) | 4 lines Preemptively fixing a possible segfault. It is + possible that queuename is NULL (meaning pause ALL queues), so + use q->name instead. ........ + +2007-09-13 20:13 +0000 [r82345] Jason Parker <jparker@digium.com> + + * /, cdr/cdr_csv.c: Merged revisions 82344 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82344 | qwell | 2007-09-13 15:11:40 -0500 (Thu, 13 Sep 2007) | 9 + lines Fix a crash that could occur in cdr_csv when mutliple + threads tried to close the same file. Do we actually need the + locking here? What happens if you open the same file twice, and + two threads try to write to it at the same time? Is fputs() going + to write out the entire line at once? I suspect that it could be + possible for the second fopen to run during the first fputs, so + the position could be in the middle of the previously written + line... Issue 10347, initial patch by explidous (but I removed + all of the paranoia stuff..) ........ + +2007-09-13 19:16 +0000 [r82338-82341] Russell Bryant <russell@digium.com> + + * /, main/astobj2.c: Merged revisions 82339 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82339 | russell | 2007-09-13 13:57:08 -0500 (Thu, 13 Sep 2007) | + 1 line resolve a warning when not building under dev mode + ........ + + * include/asterisk.h, /, main/astobj2.c, main/asterisk.c: Merged + revisions 82337 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82337 | russell | 2007-09-13 13:45:59 -0500 (Thu, 13 Sep 2007) | + 4 lines Only compile in tracking astobj2 statistics if dev-mode + is enabled. Also, when dev mode is enabled, register the CLI + command that can be used to run the astobj2 test and print out + statistics. ........ + +2007-09-13 18:13 +0000 [r82336] Kevin P. Fleming <kpfleming@digium.com> + + * /, LICENSE: Merged revisions 82335 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r82335 | kpfleming | 2007-09-13 11:12:00 -0700 + (Thu, 13 Sep 2007) | 10 lines Merged revisions 82334 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 + Sep 2007) | 2 lines clarify the OpenSSL and OpenH323 license + exceptions ........ ................ + +2007-09-13 16:58 +0000 [r82329] Joshua Colp <jcolp@digium.com> + + * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add + setvar support to chan_zap. Just like you can in chan_sip and + chan_iax2 you can now use it with zaptel channels. (done while in + Montreal at the Asterisk bootcamp!) + +2007-09-13 16:27 +0000 [r82327] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 82326 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82326 | mmichelson | 2007-09-13 11:25:59 -0500 (Thu, 13 Sep + 2007) | 7 lines Added logic to handle the unlikely case that + someone has two queues with the same name. Asterisk will log a + warning message letting the user know that one was already + defined with that name and is it skipping all further instances. + This also will work for realtime queues but in order for that to + happen, the user would have to trigger a perfectly timed reload + as a realtime queue is being looked up, which is highly unlikely + (but taken care of nonetheless). ........ + +2007-09-13 15:26 +0000 [r82321] Russell Bryant <russell@digium.com> + + * include/asterisk/doxyref.h, doc/res_config_sqlite.txt, + res/res_config_sqlite.c, configs/res_config_sqlite.conf: Various + code and documentation cleanups for res_config_sqlite (closes + issue #10711, rbraun_proformatique) + +2007-09-13 15:25 +0000 [r82312-82320] Philippe Sultan <philippe.sultan@gmail.com> + + * channels/chan_jingle.c: Modify rule filters to match with the + Jingle namespace constant + + * include/asterisk/jingle.h: Assign namespace properly + + * channels/chan_jingle.c, include/asterisk/jingle.h: Changed Jingle + and Jingle DTMF namespaces. As both specifications are in the + Experimental status, the namespaces specified therein shall be of + the form "http://www.xmpp.org/extensions/xep-XXXX.html#ns". See + the Namespace issuance section in XEP-0053 : + http://www.xmpp.org/extensions/xep-0053.html#namespaces + + * channels/chan_jingle.c: Reflect Jingle DTMF specification changes + +2007-09-13 13:34 +0000 [r82311] Russell Bryant <russell@digium.com> + + * apps/app_queue.c: Fix a missing unref of a member struct. This + was pointed out by Marta. Thanks! This function in 1.4 didn't + have the problem. + +2007-09-13 11:54 +0000 [r82310] Philippe Sultan <philippe.sultan@gmail.com> + + * /, channels/chan_gtalk.c: Merged revisions 82309 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r82309 | phsultan | 2007-09-13 13:47:14 +0200 (Thu, 13 + Sep 2007) | 4 lines Closes issue #9401, reported and patched by + irrot, with slight modifications by me. Handle DTMF sent by + Asterisk properly. ........ + +2007-09-12 21:57 +0000 [r82297] Russell Bryant <russell@digium.com> + + * /, res/res_agi.c: Merged revisions 82296 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82296 | russell | 2007-09-12 16:56:32 -0500 (Wed, 12 Sep 2007) | + 3 lines Fix a check of the wrong pointer, as pointed out by an + XXX comment left in the code. The problem was harmless, however. + ........ + +2007-09-12 21:55 +0000 [r82294] Jason Parker <jparker@digium.com> + + * channels/chan_iax2.c: After some discussions, we decided that the + return values here were a bit messy. This also fixes a bug on + reload, where peers may not have reregistered properly. + +2007-09-12 21:32 +0000 [r82290-82292] Tilghman Lesher <tlesher@digium.com> + + * /, main/stdtime/tzfile.h: Merged revisions 82291 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r82291 | tilghman | 2007-09-12 16:28:33 -0500 (Wed, 12 + Sep 2007) | 2 lines Oops, wrong location for FreeBSD zone files + ........ + + * main/stdtime/private.h, /, main/stdtime/tzfile.h, + funcs/func_strings.c, apps/app_sms.c, + include/asterisk/localtime.h, main/stdtime/localtime.c: Merged + revisions 82285 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82285 | tilghman | 2007-09-12 15:12:06 -0500 (Wed, 12 Sep 2007) + | 4 lines Working on issue #10531 exposed a rather nasty 64-bit + issue on ast_mktime, so we updated the localtime.c file from + source. Next we'll have to write ast_strptime to match. ........ + +2007-09-12 21:17 +0000 [r82289] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Removed an unneeded ao2_ref. This was a problem + because unless get_member_status returned QUEUE_NORMAL, a NULL + member would be unreferenced. While this didn't cause any crashes + or anything terrible, it still is incorrect + +2007-09-12 20:50 +0000 [r82288] Steve Murphy <murf@digium.com> + + * main/config.c: This fix closes issue #10642 -- it's not perfect, + but should retain most blank lines in config files, via + read/write cycles. + +2007-09-12 20:47 +0000 [r82287] Dwayne M. Hubbard <dhubbard@digium.com> + + * /, apps/app_meetme.c: Merged revisions 82286 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82286 | dhubbard | 2007-09-12 15:24:24 -0500 (Wed, 12 Sep 2007) + | 1 line remove a race condition for the creation of + recordthread's, and fix a small memory leak. This closes issue# + 10636 ........ + +2007-09-12 16:24 +0000 [r82283] Mark Michelson <mmichelson@digium.com> + + * main/pbx.c, main/app.c, main/asterisk.c: Fixes Solaris build + warnings (closes issue #10698, reported and patched by snuffy) + +2007-09-12 15:53 +0000 [r82279-82282] Russell Bryant <russell@digium.com> + + * utils/hashtest2.c: Change the traversal to use ao2_callback() + instead of an ao2_iterator. Using ao2_callback() is a much more + efficient way of performing an operation on every item in the + container. This change makes hashtest2 run in about 25% of the + time it ran before on my system. In general, I would say that it + makes the most sense to use an ao2_iterator if the operation + being performed is going to take a long time and you don't want + to keep the container locked while you work with each object. + Otherwise, the use of ao2_callback is preferred. + + * /, main/asterisk.c: Merged revisions 82280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82280 | russell | 2007-09-12 10:16:49 -0500 (Wed, 12 Sep 2007) | + 4 lines Clean up the output of "asterisk -h". This tweaks the + wording and wraps lines at 80 characters. (closes issue #10699, + seanbright) ........ + + * /, res/res_agi.c: Merged revisions 82278 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82278 | russell | 2007-09-12 10:11:11 -0500 (Wed, 12 Sep 2007) | + 3 lines revert patch from issue #10553, as someone not using + fastagi reported that this broke their system. ........ + +2007-09-12 14:31 +0000 [r82275-82277] Mark Michelson <mmichelson@digium.com> + + * /: Blocking changes from revision 82276 + + * /, apps/app_queue.c: Merged revisions 82274 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82274 | mmichelson | 2007-09-12 09:24:53 -0500 (Wed, 12 Sep + 2007) | 6 lines We should only initialize a realtime queue when + it is allocated, not every time we access it. This prevents the + members ao2_container from being reallocated every time the queue + is accessed. I also removed a debug message I had accidentally + left in on a previous commit. ........ + +2007-09-11 23:07 +0000 [r82273] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Fix to make sure we don't hangup a call when + getting a RLC without sending REL. Found making sure we are Q.784 + (the SS7 test specification) compliant + +2007-09-11 22:38 +0000 [r82269-82270] Russell Bryant <russell@digium.com> + + * main/config.c: remove unused functions that made this file not + build under dev mode + + * /, apps/app_queue.c: Merged revisions 82267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82267 | russell | 2007-09-11 17:37:17 -0500 (Tue, 11 Sep 2007) | + 3 lines Fix incorrect uses of ao2_find(). Every one of these + calls was reading bogus memory ... ........ + +2007-09-11 22:37 +0000 [r82268] Steve Murphy <murf@digium.com> + + * utils/Makefile, main/config.c: This solves an unreported solaris + compile problem (missing -lnsl -lsocket). + +2007-09-11 21:43 +0000 [r82266] Joshua Colp <jcolp@digium.com> + + * /, codecs/gsm/src/long_term.c, codecs/gsm/src/lpc.c: Merged + revisions 82265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82265 | file | 2007-09-11 18:41:49 -0300 (Tue, 11 Sep 2007) | 4 + lines (closes issue #10679) Reported by: andrew Build under dev + mode when K6OPTS is enabled. ........ + +2007-09-11 20:50 +0000 [r82264] Russell Bryant <russell@digium.com> + + * /, apps/app_queue.c: Merged revisions 82263 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82263 | russell | 2007-09-11 15:49:34 -0500 (Tue, 11 Sep 2007) | + 5 lines Fix another missing unref of member objects. This one was + pointed out by Marta. When building the outgoing list in + try_calling(), a member reference is stored in each outgoing + entry. However, when this list got destroyed, the reference was + not released. ........ + +2007-09-11 20:49 +0000 [r82262] Steve Murphy <murf@digium.com> + + * main/cdr.c, /: Merged revisions 82261 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82261 | murf | 2007-09-11 14:36:15 -0600 (Tue, 11 Sep 2007) | 1 + line this change should fix issue # 10659 -- what I worry about + is how many other bug reports it may generate. Hopefully, we can + please the/a majority. Hopefully. We shall see. Calls not marked + ANSWERED and with only one channel name will not be posted. This + should eliminate the double CDR's. ........ + +2007-09-11 18:37 +0000 [r82257-82258] Joshua Colp <jcolp@digium.com> + + * configs/sip.conf.sample: Lil' bit more documentation to keep + folks happy. + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: (closes + issue #9433) Reported by: junky Patches: register_trying.diff.txt + uploaded by jcmoore Disable sending 100 Trying on REGISTER + attempts and make it an option. This has been signed off by oej. + +2007-09-11 17:16 +0000 [r82256] Steve Murphy <murf@digium.com> + + * utils/Makefile: fixing up the pthread stuff for hashtest2 + +2007-09-11 16:15 +0000 [r82254] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, channels/misdn/isdn_lib.c: Merged + revisions 82249 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82249 | crichter | 2007-09-11 18:01:27 +0200 (Di, 11 Sep 2007) | + 1 line fixed a hold/retrieve issue. ........ + +2007-09-11 16:12 +0000 [r82253] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 82252 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82252 | mmichelson | 2007-09-11 11:05:56 -0500 (Tue, 11 Sep + 2007) | 6 lines All instances of ao2_iterators which were just + named 'i' have been renamed to 'mem_iter' so that when refcounted + queues are merged into trunk, there will be little confusion + regarding iterator names, especially when a queue and member + iterator are used in the same function. ........ + +2007-09-11 16:05 +0000 [r82251] Russell Bryant <russell@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 82250 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82250 | russell | 2007-09-11 11:03:42 -0500 (Tue, 11 Sep 2007) | + 4 lines The sample dundi.conf claims support for a wildcard peer + entry - [*], but the code did not support it. This patch makes it + work. (closes issue #10546, patch by dds, with some changes by + me) ........ + +2007-09-11 15:34 +0000 [r82248] Joshua Colp <jcolp@digium.com> + + * main/cdr.c: (closes issue #10666) Reported by: arkadia Patches: + cdr_lockorder.patch uploaded by arkadia (license 233) Optimize + CDR stuff a bit. + +2007-09-11 15:31 +0000 [r82246-82247] Russell Bryant <russell@digium.com> + + * res/res_agi.c: Remove an unused variable. I have no idea why this + was marked with the unused attribute instead of just removing it. + :) + + * /, res/res_agi.c: Merged revisions 82245 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82245 | russell | 2007-09-11 10:26:51 -0500 (Tue, 11 Sep 2007) | + 9 lines (closes issue #10553) Reported by: juggie Patches: + res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by: + juggie When using fastagi, fgets() can return before a full line + is read. Add explicit handling for the case where it gets + interrupted. ........ + +2007-09-11 14:58 +0000 [r82242-82244] Joshua Colp <jcolp@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 82243 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82243 | file | 2007-09-11 11:56:39 -0300 (Tue, 11 Sep 2007) | 6 + lines (closes issue #10577) Reported by: jamesgolovich Patches: + asterisk-dundifree.diff.txt uploaded by jamesgolovich (license + 176) Don't leak memory when unloading DUNDi. ........ + + * apps/app_meetme.c: (closes issue #10560) Reported by: ruffle + Patches: rb uploaded by ruffle (license 201) Show whether the + conference is locked or not on the CLI. + +2007-09-11 14:35 +0000 [r82237-82241] Russell Bryant <russell@digium.com> + + * /, apps/app_queue.c: Merged revisions 82240 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82240 | russell | 2007-09-11 09:34:12 -0500 (Tue, 11 Sep 2007) | + 2 lines Add a couple more missing unrefs of queue member objects + ........ + + * /, apps/app_queue.c: Merged revisions 82238 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82238 | russell | 2007-09-11 09:21:17 -0500 (Tue, 11 Sep 2007) | + 2 lines Add a missing unref of a queue member in an error + handling block ........ + + * /, apps/app_queue.c: Merged revisions 82236 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82236 | russell | 2007-09-11 09:09:43 -0500 (Tue, 11 Sep 2007) | + 2 lines Document why membercount can not simply be replaced by + ao2_container_count() ........ + +2007-09-11 13:46 +0000 [r82231-82235] Joshua Colp <jcolp@digium.com> + + * utils/Makefile: Include string compatibility file in hashtest2. + + * utils/hashtest2.c: Include compat.h to hopefully make it + compatible with FreeBSD. + + * utils/hashtest2.c: Fix building under FreeBSD. Make sure alloca.h + exists before including it. + + * main/manager.c: (closes issue #10695) Reported by: junky Patches: + count_showconn.diff uploaded by junky (license 177) Provide a + count of connected users to manager. + + * main/minimime/minimime.c, main/minimime/tests/create.c, + main/minimime/mm_mem.c, main/minimime/tests/parse.c: (closes + issue #10692) Reported by: snuffy Patches: minivm.diff uploaded + by snuffy (license 35) Instead of using err (which is not + available under Solaris) use fdprintf with stderr. + +2007-09-10 20:03 +0000 [r82200] Tilghman Lesher <tlesher@digium.com> + + * UPGRADE.txt, channels/chan_iax2.c: Change the IAXPeers command to + have manager-style output, instead of CLI-style output (closes + issue #8254) + +2007-09-10 19:56 +0000 [r82199] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 82198 via svnmerge ........ r82198 | russell + | 2007-09-10 14:53:17 -0500 (Mon, 10 Sep 2007) | 3 lines backport + astobj2 race condition fix. This function is the exact same as + trunk so it applies here as well. ........ + +2007-09-10 19:10 +0000 [r82185] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Fixing a problem where NULL channels would + cause a crash when calling indisposed queue members (i.e. paused, + wrapup time not completed, etc.) + +2007-09-10 18:32 +0000 [r82178] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_queue.c: Merged revisions 82155 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r82155 | tilghman | 2007-09-10 13:02:02 -0500 (Mon, 10 Sep 2007) + | 2 lines Convert struct member to use refcounts (closes issue + #10199) ........ + +2007-09-10 17:39 +0000 [r82154] Jason Parker <jparker@digium.com> + + * main/db.c: Add a counter to the 'database deltree' CLI command. + Note: this is slightly different than the initial patch, because + I felt that using res <= 0 would be a change in behavior. Closes + issue #10687, patch by junky + +2007-09-10 16:59 +0000 [r82140] Steve Murphy <murf@digium.com> + + * utils/Makefile, utils/hashtest2.c (added): Committing my test for + astobj2, hashtest2.c, along with makefile changes in utils. + +2007-09-10 16:24 +0000 [r82125] Jason Parker <jparker@digium.com> + + * main/db.c: Add counter to 'database show' CLI command. (also a + minor whitespace change that I found along the way) Closes issue + #10683, patch by junky + +2007-09-10 16:19 +0000 [r82124] Steve Murphy <murf@digium.com> + + * main/astobj2.c: Changes applied from marta's team/marta/astobj2 + branch to solve a race condition + +2007-09-10 15:05 +0000 [r82092] Mark Michelson <mmichelson@digium.com> + + * /, configs/misdn.conf.sample: Merged revisions 82091 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r82091 | mmichelson | 2007-09-10 10:02:12 -0500 (Mon, 10 + Sep 2007) | 5 lines Removing non-existent options from misdn + configuration sample. (closes issue #10678, reported and patched + by IgorG) ........ + +2007-09-10 14:26 +0000 [r82062-82077] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: (closes issue #10688) Reported by: casper + Patches: chan_sip.c.82076.diff uploaded by casper (license 55) + Remove double check for zombie flag and optimize things a bit. + + * res/res_agi.c: (closes issue #10684) Reported by: junky Patches: + debug.diff uploaded by junky (license 177) Fix issue with debug + always showing up. + + * apps/app_meetme.c: (closes issue #10686) Reported by: junky + Patches: meet.diff uploaded by junky (license 177) Change NOTICE + message to DEBUG. + +2007-09-09 02:45 +0000 [r82029] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 82028 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r82028 | tilghman | 2007-09-08 21:35:18 -0500 (Sat, 08 + Sep 2007) | 2 lines Fix inline compiles on really old compilers + (who uses gcc 2.7 anymore, really?) (closes issue #10675) + ........ + +2007-09-08 19:01 +0000 [r81998-81999] Russell Bryant <russell@digium.com> + + * include/asterisk/slinfactory.h: Add doxygen documentation for + slinfactory_destroy(), mainly just noting that it doesn't free + the slinfactory itself. (This isn't related to a bug, i'm just + looking over random code) + + * /, main/asterisk.c: Merged revisions 81997 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81997 | russell | 2007-09-08 13:41:32 -0500 (Sat, 08 Sep 2007) | + 2 lines Fix a small memory leak. ast_unregister_atexit() did not + free the entry it removed. ........ + +2007-09-08 16:37 +0000 [r81984] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: Make Callerid more consistent in IMAP mail + headers (closes issue #10056, reported and patched by jaroth, + with small modification by me) + +2007-09-08 13:45 +0000 [r81953] Russell Bryant <russell@digium.com> + + * /, .cleancount: Merged revisions 81952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81952 | russell | 2007-09-08 08:42:26 -0500 (Sat, 08 Sep 2007) | + 11 lines (closes issue #10672) Bump the cleancount so that a + "make clean" will be forced. This is needed because my fix in + revision 81599 made a change to a data structure in file.h, and + since file dependency tracking is only on with dev-mode enabled, + file format modules that don't get rebuilt may crash, as is the + case with this issue. This makes me wonder - how much faster does + the code build without the file dependency tracking enabled? If + it doesn't make much of a difference, then it may be worth just + keeping it on all of the time, or perhaps just not in release + tarballs, so that this type of issue is avoided. ........ + +2007-09-07 19:53 +0000 [r81910-81924] Jason Parker <jparker@digium.com> + + * /, apps/app_queue.c: Merged revisions 81923 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #10671) ........ r81923 | qwell | 2007-09-07 14:48:00 -0500 + (Fri, 07 Sep 2007) | 5 lines Allow the MEMBERINTERFACE variable + to be used as the mixmonitor filename. This moves the setting of + the MEMBERINTERFACE variable to before mixmonitor. Issue 10671, + patch by sim. ........ + + * apps/app_queue.c: Add an optional reason parameter to + PauseQueueMember/UnpauseQueueMember applications and manager + events. Issue 8738, patch by rgollent + +2007-09-07 15:29 +0000 [r81891] Mark Michelson <mmichelson@digium.com> + + * /, configs/queues.conf.sample: Merged revisions 81886 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81886 | mmichelson | 2007-09-07 10:25:19 -0500 (Fri, 07 Sep + 2007) | 3 lines Moving the explanation for joinempty to a more + appropriate place ........ + +2007-09-07 12:32 +0000 [r81858-81873] Joshua Colp <jcolp@digium.com> + + * configure, configure.ac: Don't check for epoll support when cross + compiling. + + * main/channel.c, main/audiohook.c: Fix memory issue that crept up + with Russell's testing. It is *not* proper to free the frame we + get in ast_write. + +2007-09-06 22:32 +0000 [r81839-81849] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: fix the build ... oops + + * /, channels/chan_sip.c: Merged revisions 81832 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81832 | russell | 2007-09-06 17:28:57 -0500 (Thu, 06 Sep 2007) | + 16 lines (closes issue #9724, closes issue #10374) Reported by: + kenw Patches: 9724.txt uploaded by russell (license 2) Tested by: + kenw, russell Resolve a deadlock that occurs when doing a SIP + transfer to parking. I come across this type of deadlock fairly + often it seems. It is very important to mind the boundary between + the channel driver and the core in respect to the channel lock + and the channel-pvt lock. Channel drivers lock to lock the pvt + and then the channel once it calls into the core, while the core + will do it in the opposite order. The way this is avoided is by + having channel drivers either release their pvt lock while + calling into the core, or such as in this case, unlocking the pvt + just long enough to acquire the channel lock. ........ + +2007-09-06 22:06 +0000 [r81827] Jason Parker <jparker@digium.com> + + * Makefile, /: Merged revisions 81826 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81826 | qwell | 2007-09-06 17:05:02 -0500 (Thu, 06 Sep 2007) | 1 + line We added COPTS for ASTCFLAGS additions, but not LDOPTS for + ASTLDFLAGS. This adds LDOPTS ........ + +2007-09-06 21:01 +0000 [r81814] Joshua Colp <jcolp@digium.com> + + * channels/iax2-parser.c: Initialize iax_frames variable to NULL, + keeps valgrind happy. + +2007-09-06 20:54 +0000 [r81783-81813] Russell Bryant <russell@digium.com> + + * CHANGES, funcs/func_extstate.c (added): Add EXTENSION_STATE() + function that can retrieve the state of an extension that has a + hint. (closes issue #10635, adamgundy) + + * CHANGES: s/DEVSTATE/DEVICE_STATE/ + + * funcs/func_devstate.c: Rename the DEVSTATE() function to + DEVICE_STATE() to better conform to how other functions are + named. (inspired by issue #10635) + + * CHANGES, funcs/func_devstate.c: Merge HINT() dialplan function + from my sandbox branch into trunk. This function will let you + retrieve the list of devices or name associated with a hint. + (inspired by issue #10635) + +2007-09-06 20:16 +0000 [r81782] Joshua Colp <jcolp@digium.com> + + * channels/chan_skinny.c, CHANGES: (closes issue #10377) Reported + by: mvanbaak Patches: chan_skinny_info.diff uploaded by mvanbaak + (license 7) Add skinny show device, skinny show line, and skinny + show settings CLI commands. + +2007-09-06 20:05 +0000 [r81781] Russell Bryant <russell@digium.com> + + * configs/extensions.conf.sample: Fix the syntax of declaring a + hint with a name to be compatible with trunk + +2007-09-06 20:00 +0000 [r81779] Jason Parker <jparker@digium.com> + + * /, include/asterisk/astobj2.h: Merged revisions 81778 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81778 | qwell | 2007-09-06 14:59:07 -0500 (Thu, 06 Sep 2007) | 2 + lines This should fix a build issue that people building against + uClibc were seeing with the addition of astobj2 ........ + +2007-09-06 19:43 +0000 [r81777] Joshua Colp <jcolp@digium.com> + + * /, apps/app_meetme.c: Merged revisions 81776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81776 | file | 2007-09-06 16:40:37 -0300 (Thu, 06 Sep 2007) | 7 + lines (closes issue #10122) Reported by: stevefeinstein Patches: + meetme-unmute-manager.diff uploaded by qwell (license 4) Tested + by: stevefeinstein After looking over the code I agree with + Qwell. Setting the file descriptor to conference each time just + causes a fight back and forth. ........ + +2007-09-06 17:00 +0000 [r81745] Philippe Sultan <philippe.sultan@gmail.com> + + * /, include/asterisk/jabber.h, channels/chan_gtalk.c: Merged + revisions 81743 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81743 | phsultan | 2007-09-06 18:56:29 +0200 (Thu, 06 Sep 2007) + | 1 line Various string length fixes. Removed an unused variable + in aji_client structure (context) ........ + +2007-09-06 16:57 +0000 [r81744] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/safe_asterisk: Incorporate the ability to log + output of safe_asterisk to syslog (closes issue #9882) + +2007-09-06 16:38 +0000 [r81742] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Patch on 10575. Add support for unequipped + CIC (UCIC) message as well as improve some of our CIC flags in + chan_zap + +2007-09-06 16:31 +0000 [r81730] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 81713 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81713 | mmichelson | 2007-09-06 11:25:40 -0500 (Thu, 06 Sep + 2007) | 6 lines Fixes an issue where valid DTMF had to be pressed + twice to exit a queue if a member's phone was ringing. (closes + issue #10655, reported by strider2k, patched by me) ........ + +2007-09-06 15:43 +0000 [r81712] Luigi Rizzo <rizzo@icir.org> + + * include/asterisk/astobj2.h, main/astobj2.c: various changes to + the documentation, and redefinition of ao2_hash_fn and + ao2_callback_fn typedefs, in preparation to more cleanup of the + _search_flags Please do not merge this change to 1.4 yet - there + are no functional changes anyways. + +2007-09-06 15:21 +0000 [r81683] Mark Michelson <mmichelson@digium.com> + + * /, res/res_features.c: Merged revisions 81682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81682 | mmichelson | 2007-09-06 10:20:36 -0500 (Thu, 06 Sep + 2007) | 5 lines Fixes a memory leak (closes issue #10658, + reported and patched by Ivan) ........ + +2007-09-06 14:24 +0000 [r81651] Philippe Sultan <philippe.sultan@gmail.com> + + * /, res/res_jabber.c: Merged revisions 81650 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81650 | phsultan | 2007-09-06 16:20:54 +0200 (Thu, 06 Sep 2007) + | 3 lines According to both RFC 3920 - section 9.1.2 - and + Google's XMPP server complaint, if set, the 'from' attribute must + be set to the user's full JID. ........ + +2007-09-05 21:59 +0000 [r81632] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Not having this epoll specific code in + wait_for_answer was causing app_queue to infinitely loop. This + makes it so it doesn't. Thanks to file for pointing out where the + problem was and showing a similar function in app_dial as an + example of how to fix it. + +2007-09-05 21:45 +0000 [r81631] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 81569 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r81569 | tilghman | 2007-09-05 12:18:24 -0500 (Wed, 05 + Sep 2007) | 2 lines Solaris x86 compatibility fix ........ + +2007-09-05 20:58 +0000 [r81601] Dwayne M. Hubbard <dhubbard@digium.com> + + * apps/app_zapateller.c: added ZAPATELLERSTATUS to app_zapateller + +2007-09-05 20:58 +0000 [r81600] Russell Bryant <russell@digium.com> + + * include/asterisk/file.h, /, main/say.c, res/res_features.c, + main/file.c, include/asterisk/channel.h: Merged revisions 81599 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) | + 11 lines Fix an issue that can occur when you do an attended + transfer to parking. If you complete the transfer before the + announcement of the parking spot finishes, then the channel being + parked will hear the remainder of the announcement. These changes + make it so that will not happen anymore. Basically, res_features + sets a flag on the channel is playing the announcement to so that + the file streaming core knows that it needs to watch out for a + channel masquerade, and if it occurs, to abort the announcement. + (closes BE-182) ........ + +2007-09-05 16:48 +0000 [r81568] Tilghman Lesher <tlesher@digium.com> + + * utils: Add two more generated files (requested by mvanbaak via + irc) + +2007-09-05 16:31 +0000 [r81560] Jason Parker <jparker@digium.com> + + * include/asterisk/devicestate.h, res/res_config_odbc.c, + channels/chan_sip.c, include/asterisk/audiohook.h, main/sha1.c, + res/res_features.c, include/asterisk/astobj2.h, res/res_crypto.c, + include/asterisk/strings.h, main/audiohook.c, res/res_jabber.c, + res/res_config_sqlite.c, include/asterisk/sha1.h, + include/asterisk/stringfields.h, include/asterisk/features.h: + Doxygen cleanups/fixes. Closes issue #10654, patch by snuffy + +2007-09-05 15:32 +0000 [r81526-81535] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Weird. When I merged my changes from 1.4, they + merged into the wrong function. This should fix the build for + trunk. + + * /, apps/app_queue.c: Merged revisions 81525 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81525 | mmichelson | 2007-09-05 10:19:47 -0500 (Wed, 05 Sep + 2007) | 4 lines Fixing the build... ........ + +2007-09-05 15:16 +0000 [r81524] Jason Parker <jparker@digium.com> + + * channels/chan_phone.c, /: Merged revisions 81523 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #10651) ........ r81523 | qwell | 2007-09-05 10:14:30 -0500 + (Wed, 05 Sep 2007) | 5 lines Do not try to unregister a NULL + channel tech. Also changed load_module function to use defines + rather than numbers for return values. Issue 10651, patch by + rbraun_proformatique, with additions by me. ........ + +2007-09-05 15:04 +0000 [r81522] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 81520 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81520 | mmichelson | 2007-09-05 10:03:22 -0500 (Wed, 05 Sep + 2007) | 6 lines Reverting behavior of QUEUE_MEMBER_COUNT to only + count members who are logged in and available. (related to issue + #10652, reported by wuwu) ........ + +2007-09-05 14:47 +0000 [r81519] Steve Murphy <murf@digium.com> + + * include/asterisk/config.h, main/config.c: this set of changes + fixes issue # 10643 by keeping track of the last object defined + in a file, and attaching any accumulated comments to that object + (category header or variable declaration). The file_save routine + also had to be upgraded to output these trailing comments. + Config.h was modified to include the trailing comment list on + categories and variables. + +2007-09-05 13:13 +0000 [r81459-81493] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 81492 via svnmerge ........ r81492 | file | + 2007-09-05 10:11:48 -0300 (Wed, 05 Sep 2007) | 4 lines (closes + issue #10650) Reported by: tacvbo Only print out that the spy was + removed while holding the spy lock. ........ + + * main/editline/sys.h: Finish up commit from revision 81452 by + removing last remnants of strlcat/strlcpy checks. + +2007-09-04 20:59 +0000 [r81454-81456] Jason Parker <jparker@digium.com> + + * /, apps/app_followme.c: Merged revisions 81455 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #10634) ........ r81455 | qwell | 2007-09-04 15:54:51 -0500 + (Tue, 04 Sep 2007) | 4 lines Rather than attempt to play a file, + we can just check whether it exists. Issue 10634, patch by me, + testing by pabelanger, sanity checked by bweschke ........ + + * /, configs/followme.conf.sample: Merged revisions 81453 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #10644) ........ r81453 | qwell | 2007-09-04 14:56:06 -0500 + (Tue, 04 Sep 2007) | 4 lines Change default followme config file + to point to the correct files. Issue 10644, patch by pabelanger + ........ + +2007-09-04 19:51 +0000 [r81445-81452] Russell Bryant <russell@digium.com> + + * main/editline/configure, main/editline/configure.in: Don't check + for and include strlcpy and strlcat in editline. We also include + them directly in Asterisk. For platforms that need them (like my + mac), you will get a linker error due to the functions being + included twice. + + * /, include/asterisk/astobj2.h, channels/chan_iax2.c, + main/astobj2.c: Merged revisions 81448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81448 | russell | 2007-09-04 13:37:44 -0500 (Tue, 04 Sep 2007) | + 4 lines Remove the typedefs on ao2_container and ao2_iterator. + This is simply because we don't typedef objects anywhere else in + Asterisk, so we might as well make this follow the same + convention. ........ + + * include/asterisk/logger.h: logger.h depends on options.h, so go + ahead and include it + +2007-09-04 16:41 +0000 [r81443] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c: Merged revisions 81442 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81442 | kpfleming | 2007-09-04 11:40:39 -0500 (Tue, 04 Sep 2007) + | 2 lines there is no point in sending 401 Unauthorized to a UAS + that sent us a properly-formatted Authentication header with the + expected username and nonce but an incorrect response (which + indicates the shared secret does not match)... instead, let's + send 403 Forbidden so that the UAS doesn't retry with the same + authentication credentials repeatedly ........ + +2007-09-04 14:28 +0000 [r81436-81441] Joshua Colp <jcolp@digium.com> + + * configs/extensions.ael.sample: (closes issue #10633) Reported by: + pabelanger Patches: extensions.ael.sample.patch uploaded by + pabelanger (license 224) Update extensions.ael.sample with + voicemail and | changes. + + * /, channels/chan_iax2.c: Merged revisions 81439 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81439 | file | 2007-09-04 11:23:18 -0300 (Tue, 04 Sep 2007) | 6 + lines (closes issue #10632) Reported by: jamesgolovich Patches: + asterisk-iaxfirmwareleak.diff.txt uploaded by jamesgolovich + (license 176) Fix memory leak when unloading chan_iax2. The + firmware files were not being freed. ........ + + * main/channel.c, /: Merged revisions 81437 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81437 | file | 2007-09-04 10:46:23 -0300 (Tue, 04 Sep 2007) | 4 + lines (closes issue #10476) Reported by: mdu113 Only look for the + end of a digit when waiting for a digit. This in turn disables + emulation in the core. ........ + + * /, main/dns.c: Merged revisions 81435 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81435 | file | 2007-09-04 10:10:56 -0300 (Tue, 04 Sep 2007) | 7 + lines (closes issue #10610) Reported by: john Patches: + dns.c.patch uploaded by john (license 218) Tested by: mvanbaak + Don't return a match if no SRV record actually exists. ........ + +2007-09-03 18:59 +0000 [r81434] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 81433 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81433 | russell | 2007-09-03 13:57:53 -0500 (Mon, 03 Sep 2007) | + 5 lines Remove a couple of calls to ast_string_field_free_pools() + on peers in error handling blocks in the code for building peers. + The peer object destructor does this and doing it twice will + cause a crash. (closes issue #10625, reported by and patched by + pnlarsson) ........ + +2007-09-03 18:01 +0000 [r81430-81432] Tilghman Lesher <tlesher@digium.com> + + * main/config.c: Once we get past the file checks, we're loading, + so clear the FILEUNCHANGED flag (fixes #include) (closes issue + #10629) + + * /, funcs/func_logic.c: Merged revisions 81415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81415 | tilghman | 2007-08-31 14:16:52 -0500 (Fri, 31 Aug 2007) + | 2 lines The IF() function was not allowing true values that had + embedded colons (closes issue #10613) ........ + + * main/config.c: We shouldn't use a filename blindly without + checking to make sure it's unused first + +2007-09-01 06:03 +0000 [r81427] Mark Michelson <mmichelson@digium.com> + + * /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions + 81426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81426 | mmichelson | 2007-09-01 01:02:06 -0500 (Sat, 01 Sep + 2007) | 4 lines Making match_by_addr into ao2_match_by_addr and + making it available everywhere since it could be a handy callback + to have ........ + +2007-08-31 21:29 +0000 [r81419] Russell Bryant <russell@digium.com> + + * /, include/asterisk/astobj2.h: Merged revisions 81418 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81418 | russell | 2007-08-31 16:27:49 -0500 (Fri, 31 Aug 2007) | + 2 lines Remove references to a debugging parameter that does not + exist ........ + +2007-08-31 19:50 +0000 [r81417] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 81416 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81416 | mmichelson | 2007-08-31 14:48:55 -0500 (Fri, 31 Aug + 2007) | 6 lines Fixed broken behavior of a reload on realtime + queues. Prior to this patch, if a reload was issued and a + realtime queue had callers waiting in it, then the queue would be + removed from the queue list, but it would not actually be freed + (in fact, a debug message warning about a memory leak would come + up). With this patch, reloads do not touch realtime queues at + all. ........ + +2007-08-31 18:46 +0000 [r81413] Jason Parker <jparker@digium.com> + + * apps/app_dial.c, /: Merged revisions 81412 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #10621) ........ r81412 | qwell | 2007-08-31 13:44:44 -0500 + (Fri, 31 Aug 2007) | 4 lines Re-order dial options to be in line + with the existing alpha order. Issue 10621, initial patch by + junky ........ + +2007-08-31 17:43 +0000 [r81411] Philippe Sultan <philippe.sultan@gmail.com> + + * /, channels/chan_gtalk.c: Merged revisions 81410 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r81410 | phsultan | 2007-08-31 19:38:26 +0200 (Fri, 31 + Aug 2007) | 3 lines Make the 'gtalk show channels' CLI command + available. Closes issue 10548, reported by keepitcool. ........ + +2007-08-31 15:58 +0000 [r81408] Kevin P. Fleming <kpfleming@digium.com> + + * /, codecs/codec_zap.c: Merged revisions 81405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81405 | kpfleming | 2007-08-31 10:51:45 -0500 (Fri, 31 Aug 2007) + | 2 lines add missing "transcoder show" (and deprecated "show + transcoder") CLI commands that were in 1.2 but never added to 1.4 + ........ + +2007-08-31 15:54 +0000 [r81402-81407] Joshua Colp <jcolp@digium.com> + + * /, res/res_speech.c: Merged revisions 81406 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81406 | file | 2007-08-31 12:53:16 -0300 (Fri, 31 Aug 2007) | 2 + lines Make it the engine's responsible to check for the presence + of results. ........ + + * /, res/res_features.c: Merged revisions 81403 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81403 | file | 2007-08-31 11:38:59 -0300 (Fri, 31 Aug 2007) | 4 + lines (closes issue #10618) Reported by: dimas Don't pass through + the stopped sounds frame.... just drop it. ........ + + * /, res/res_features.c: Merged revisions 81401 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81401 | file | 2007-08-30 20:53:41 -0300 (Thu, 30 Aug 2007) | 4 + lines (closes issue #10009) Reported by: dimas Don't output a + bridge failed warning message if it failed because one of the + channels was part of the masquerade process. That is perfectly + normal. ........ + +2007-08-30 23:52 +0000 [r81400] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_zap.c: Add new queryable fields from zaptel to 'zap + show status' + +2007-08-30 22:08 +0000 [r81398] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 81397 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81397 | mmichelson | 2007-08-30 17:05:56 -0500 (Thu, 30 Aug + 2007) | 7 lines Removing an extraneous (and possibly misleading) + log message. Firstly, if the announce file isn't found, the + streaming functions will report it. Secondly, not all non-zero + returns from play_file mean that the announce file wasn't found. + Positive return values simply mean that a digit was pressed (most + likely to skip through the announcement). (closes issue #10612, + reported and patched by dimas) ........ + +2007-08-30 21:25 +0000 [r81394-81396] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 81395 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81395 | file | 2007-08-30 18:23:50 -0300 (Thu, 30 Aug 2007) | 6 + lines (closes issue #10514) Reported by: casper Patches: + chan_sip.c.80129.diff uploaded by casper (license 55) Remove + needless check for AUTH_UNKNOWN_DOMAIN. It was impossible for it + to ever be that value. ........ + + * channels/chan_sip.c: (closes issue #10565) Reported by: tootai + Make sure the external IP address has the standard SIP port set + for when the user does not specify the port in the externip + setting. + +2007-08-30 21:16 +0000 [r81393] Steve Murphy <murf@digium.com> + + * main/cdr.c, /: Merged revisions 81392 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81392 | murf | 2007-08-30 15:11:48 -0600 (Thu, 30 Aug 2007) | 1 + line via issue 10599, where 'CDR already initialized' messages + are being generated. Since all channels will have an init'd CDR + attached at creation time, this message is now particularly + useless. Removed. ........ + +2007-08-30 20:55 +0000 [r81391] Joshua Colp <jcolp@digium.com> + + * apps/app_minivm.c: (closes issue #10336) Reported by: junky + Patches: minivm_output2.diff uploaded by junky (license 177) + Change console output of minivm show stats to be more simple for + external parsing. + +2007-08-30 20:31 +0000 [r81389-81390] Tilghman Lesher <tlesher@digium.com> + + * main/sched.c: A schedule id of 0 is not possible and is used to + flag that we want to add a new item + + * apps/app_readexten.c: Change wording as requested by Kevin + +2007-08-30 18:52 +0000 [r81388] Mark Michelson <mmichelson@digium.com> + + * configs/queues.conf.sample: Added note to sample queues.conf file + to line up with most recent change regarding setinterfacevar. + MEMBERREALTIME indicates whether a member is realtime. + +2007-08-30 17:51 +0000 [r81387] Tilghman Lesher <tlesher@digium.com> + + * main/logger.c: Always force reread of the config when we're + rotating the log file (closes issue #10598) + +2007-08-30 15:40 +0000 [r81384] Russell Bryant <russell@digium.com> + + * /, channels/h323/ast_h323.cxx: Merged revisions 81383 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81383 | russell | 2007-08-30 10:38:29 -0500 (Thu, 30 Aug 2007) | + 3 lines Add missing checks for the PTRACING define. (closes issue + #10559, paravoid) ........ + +2007-08-30 15:36 +0000 [r81382] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 81381 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81381 | mmichelson | 2007-08-30 10:35:51 -0500 (Thu, 30 Aug + 2007) | 3 lines Changed some manager event messages to reflect + whether a queue member is a realtime member or not ........ + +2007-08-30 15:34 +0000 [r81380] Russell Bryant <russell@digium.com> + + * configs/modem.conf.sample (removed), /, configs/enum.conf.sample, + configs/extensions.ael.sample: Merged revisions 81379 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81379 | russell | 2007-08-30 10:33:48 -0500 (Thu, 30 Aug 2007) | + 3 lines Fix a typo, update a reload command, and remove an unused + configuration file. (closes issue #10606, casper) ........ + +2007-08-30 15:24 +0000 [r81378] Tilghman Lesher <tlesher@digium.com> + + * apps/app_readexten.c (added): Add ReadExten app and VALID_EXTEN + function (closes issue #10082) + +2007-08-30 14:55 +0000 [r81377] Joshua Colp <jcolp@digium.com> + + * /: Blocked revisions 81375 via svnmerge ........ r81375 | file | + 2007-08-30 11:53:43 -0300 (Thu, 30 Aug 2007) | 6 lines (closes + issue #10603) Reported by: jmls Patches: pbx.diff uploaded by + jmls (license 141) Backport changes from 81372. Add REASON + dialplan variable for when an originated call fails and the + failed extension is executed. ........ + +2007-08-30 14:54 +0000 [r81376] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /: Merged revisions 81373 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r81373 | crichter | 2007-08-30 16:43:33 +0200 (Do, 30 + Aug 2007) | 1 line Fixed some warnings. ........ + +2007-08-30 14:42 +0000 [r81370-81372] Joshua Colp <jcolp@digium.com> + + * main/pbx.c, CHANGES: (closes issue #10603) Reported by: jmls + Patches: pbx.diff uploaded by jmls (license 141) Add REASON + dialplan variable for when an originated call fails and the + failed extension is executed. + + * /, res/res_features.c: Merged revisions 81369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81369 | file | 2007-08-30 11:23:40 -0300 (Thu, 30 Aug 2007) | 4 + lines (issue #10599) Reported by: dimas Handle the -1 control + subclass during feature dialing (it indicates to stop sounds). + ........ + +2007-08-30 08:50 +0000 [r81368] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged + revisions 81367 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81367 | crichter | 2007-08-30 10:31:59 +0200 (Do, 30 Aug 2007) | + 11 lines Fixed a severe issue where a misdn_read would lock the + channel, but read would not return because it blocks. later + chan_misdn would try to queue a frame like a AST_CONTROL_ANSWER + which could result in a deadlock situation. misdn_read will now + not block forever anymore, and we don't queue the ANSWER frame at + all when we already was called with misdn_answer -> answer would + be called twice. Also we don't explicitly send a RELEASE_COMPLETE + on receiption of a RELEASE anymore, because mISDN does that for + us, this resulted in a problem on some switches, which would + block our port after some calls for a short while. ........ + +2007-08-29 22:05 +0000 [r81365] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Added the MEMBERREALTIME variable when using + setinterfacevar in queues.conf + +2007-08-29 21:55 +0000 [r81364] Joshua Colp <jcolp@digium.com> + + * include/asterisk/event.h: Make the event header file work under + C++. + +2007-08-29 21:30 +0000 [r81363] Steve Murphy <murf@digium.com> + + * main/config.c: init newer so compile won't complain. + +2007-08-29 21:25 +0000 [r81362] Russell Bryant <russell@digium.com> + + * main/config.c: make trunk build again. murf will have to review + this to see if it was the right fix, as it is related to his last + change. + +2007-08-29 20:55 +0000 [r81361] Steve Murphy <murf@digium.com> + + * res/res_config_pgsql.c, channels/chan_sip.c, + include/asterisk/config.h, channels/chan_iax2.c, + channels/iax2-parser.c, res/res_config_sqlite.c, main/config.c, + main/channel.c, res/res_config_odbc.c, pbx/pbx_spool.c, + main/manager.c, channels/chan_skinny.c, apps/app_minivm.c, + main/http.c, utils/extconf.c, apps/app_directory.c, + apps/app_parkandannounce.c, apps/app_voicemail.c: This code was + in team/murf/bug8684-trunk; it should fix bug 8684 in trunk. I + didn't add it to 1.4 yet, because it's not entirely clear to me + if this is a bug fix or an enhancement. A lot of files were + affected by small changes like ast_variable_new getting an added + arg, for the file name the var was defined in; ast_category_new + gets added args of filename and lineno; ast_category and + ast_variable structures now record file and lineno for each + entry; a list of all #include and #execs in a config file (or any + of its inclusions are now kept in the ast_config struct; at save + time, each entry is put back into its proper file of origin, in + order. #include and #exec directives are folded in properly. + Headers indicating that the file was generated, are generated + also for each included file. Some changes to main/manager.c to + take care of file renaming, via the UpdateConfig command. + Multiple inclusions of the same file are handled by exploding + these into multiple include files, uniquely named. There's + probably more, but I can't remember it right now. + +2007-08-29 19:41 +0000 [r81353-81356] Russell Bryant <russell@digium.com> + + * main/event.c: Try to clarify the rules on changing ast_event and + ast_event_ie + + * main/event.c: Fix parenthesis from my last commit + + * main/event.c: Change pointer aritmetic on void * to char * + + * main/event.c: there is not actually code that sends these over + the network in trunk yet + +2007-08-29 16:39 +0000 [r81350] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 81349 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81349 | mmichelson | 2007-08-29 11:35:29 -0500 (Wed, 29 Aug + 2007) | 12 lines This patch, in essence, will correctly pause a + realtime queue member and reflect those changes in the realtime + engine. (issue #10424, reported by irroot, patch by me) This + patch creates a new function called update_realtime_member_field, + which is a generic function which will allow any one field of a + realtime queue member to be updated. This patch only uses this + function to update the paused status of a queue member, but it + lays the foundation for persisting the state of a realtime member + the same way that static members' state is maintained when using + the persistentmembers setting ........ + +2007-08-29 16:25 +0000 [r81348] Joshua Colp <jcolp@digium.com> + + * main/event.c: Return ast_event_get_ie_raw to using an iterator + and fix logic in ast_event_iterator_next. + +2007-08-29 16:09 +0000 [r81347] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 81346 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81346 | mmichelson | 2007-08-29 11:08:09 -0500 (Wed, 29 Aug + 2007) | 3 lines Changed some tabs to spaces ........ + +2007-08-29 16:07 +0000 [r81344-81345] Joshua Colp <jcolp@digium.com> + + * main/event.c: This concludes bringing trunk back to a working + state. + + * include/asterisk/event.h, main/event.c: To keep others happy... + revert part of my additions so trunk works. + +2007-08-29 15:59 +0000 [r81343] Russell Bryant <russell@digium.com> + + * /, main/Makefile: Merged revisions 81342 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81342 | russell | 2007-08-29 10:57:29 -0500 (Wed, 29 Aug 2007) | + 3 lines If chan_h323 is not being built, don't use g++ to do the + final link of Asterisk. (in response to a question on the + asterisk-dev list) ........ + +2007-08-29 15:57 +0000 [r81341] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 81340 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81340 | mmichelson | 2007-08-29 10:52:42 -0500 (Wed, 29 Aug + 2007) | 8 lines This fix creates a more accurate way of detecting + whether realtime members were deleted. (closes issue 10541, + reported by Alric, patched by me) The REALLY nice things about + this patch is that queue members now have a "realtime" field + which will be true if the member is a realtime member. This means + we can check this value prior to certain processing if it should + ONLY be done for realtime members. ........ + +2007-08-29 15:21 +0000 [r81335] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_iax2.c: Changed one too many variable settings in + issue #9315 (closes issue #10592) + +2007-08-29 15:19 +0000 [r81334] Joshua Colp <jcolp@digium.com> + + * include/asterisk/event.h, include/asterisk/event_defs.h, + main/event.c: Add API calls for iterating through an event. This + should allow events to have multiple information elements (while + there was nothing preventing it before you could not actually + access any except the first one). + +2007-08-29 14:19 +0000 [r81333] Mark Michelson <mmichelson@digium.com> + + * apps/app_meetme.c: Changing a NOTICE to a DEBUG. (closes issue + #10591, reported and patched by junky, with small modification by + me) + +2007-08-29 14:16 +0000 [r81326-81332] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 81331 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81331 | file | 2007-08-29 11:13:55 -0300 (Wed, 29 Aug 2007) | 4 + lines (closes issue #9690) Reported by: mattv Make rtp timeouts + work even if two RTP streams are directly bridged in the RTP + stack. ........ + + * include/asterisk/utils.h: Add inline function for signed linear + subtraction. + +2007-08-28 21:39 +0000 [r81292] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 81291 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81291 | russell | 2007-08-28 16:38:26 -0500 (Tue, 28 Aug 2007) | + 3 lines Change the message about receiving a mini-frame before + the first full voice frame to a DEBUG message. ........ + +2007-08-28 21:35 +0000 [r81290] Joshua Colp <jcolp@digium.com> + + * main/logger.c: Add some read/write locking magic to make logger + reload operate again. + +2007-08-28 20:03 +0000 [r81277] Tilghman Lesher <tlesher@digium.com> + + * main/logger.c, UPGRADE.txt, configs/logger.conf.sample: Support + better rotation of log files to be more like system logging + (closes issue #10398) + +2007-08-28 19:12 +0000 [r81227-81264] Russell Bryant <russell@digium.com> + + * include/asterisk/audiohook.h: Change the audiohook lock and + unlock wrappers to macros instead of inline functions. As inline + functions, the lock debug information will show that these are + always locked in audiohooks.h instead of the file where the lock + was actually acquired. + + * funcs/func_enum.c, pbx/pbx_dundi.c: Add proper channel locking + around the uses of datastore_add and _find. There are still more + places in the tree that I have not yet changed if someone wants + to go through and find the places they are used without the + channel locked. + + * main/channel.c, funcs/func_volume.c, include/asterisk/channel.h: + * Constify the uid field of channel datastores * Convert some + spaces to tabs in func_volume * Add a note in channel.h making it + clear that none of the datastore API calls lock the channel they + are given, so the channel should be locked before calling the + functions that take a channel argument. + + * include/asterisk/app.h, main/app.c, CHANGES, main/asterisk.c, + doc/tex/asterisk-conf.tex: (closes issue #7852) Reported by: + nic_bellamy Patches: + 2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by + nic_bellamy (license 213) Add support for configurable file + locking methods. The default is "lockfile", which is the old + behavior. There is an additional option, "flock", which is + intended for use in situations where the lockfile method will not + work, such as with SMB/CIFS mounts. + + * /, configs/indications.conf.sample: Merged revisions 81226 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81226 | russell | 2007-08-28 10:41:15 -0500 (Tue, 28 Aug 2007) | + 2 lines Add Russian tones. (closes issue #7953, hanabana) + ........ + +2007-08-28 14:37 +0000 [r81210] Joshua Colp <jcolp@digium.com> + + * res/res_features.c: (closes issue #10579) Reported by: ornati + Make sure the called channel during the attended transfer process + becomes associated with the calling channel so that the + ast_waitfor_* call works properly under epoll. + +2007-08-28 14:12 +0000 [r81121-81190] Mark Michelson <mmichelson@digium.com> + + * /, contrib/scripts/vmail.cgi: Merged revisions 81189 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r81189 | mmichelson | 2007-08-28 09:12:14 -0500 (Tue, 28 + Aug 2007) | 5 lines Fixes a forwarding problem when using + res_config_mysql (closes issue #10573, reported by chrisvaughan, + patch suggested by chrisvaughan as well) ........ + + * /, apps/app_queue.c: Merged revisions 81158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81158 | mmichelson | 2007-08-27 17:40:19 -0500 (Mon, 27 Aug + 2007) | 5 lines Resolve a potential deadlock. In this case, a + single queue is locked, then the queue list. In changethread(), + the queue list is locked, and then each individual queue is + locked. Under the right circumstances, this could deadlock. As + such, I have unlocked the individual queue before locking the + queue list, and then locked the queue back after the queue list + is unlocked. ........ + + * /, channels/chan_agent.c: Merged revisions 81120 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r81120 | mmichelson | 2007-08-27 16:08:48 -0500 (Mon, 27 + Aug 2007) | 7 lines DTMF begin frames should be ignored so that + when an agent acks a call with the '#' key, he doesn't cause a + queue's announce file to be interrupted. Also went ahead and did + the same for the '*' key and for ending a call. (closes issue + #10528, reported by deskhack, patched by me) ........ + +2007-08-27 20:55 +0000 [r81118] Tilghman Lesher <tlesher@digium.com> + + * apps/app_directed_pickup.c: Enhance Pickup to do native + pickupgroup pickup when no arguments are specified (closes issue + #10404) + +2007-08-27 17:44 +0000 [r81043-81098] Russell Bryant <russell@digium.com> + + * /, pbx/pbx_dundi.c: This should have been trunk only, I guess. oh + well ... it's harmless. Merged revisions 81065 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81065 | russell | 2007-08-27 11:38:33 -0500 (Mon, 27 Aug 2007) | + 1 line explicity define a variable as a boolean ........ + + * /, pbx/pbx_dundi.c: Merged revisions 81074 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81074 | russell | 2007-08-27 12:27:48 -0500 (Mon, 27 Aug 2007) | + 3 lines Add a \todo to note that this module leaks most of the + memory it allocates on unload and should be fixed (when I'm not + in the middle of something else ...). ........ + + * /, res/res_musiconhold.c: Merged revisions 81042 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r81042 | russell | 2007-08-27 11:16:25 -0500 (Mon, 27 + Aug 2007) | 11 lines (closes issue #10419) Reported by: + mustardman Patches: asterisk-mohposition.diff.txt uploaded by + jamesgolovich (license 176) This patch fixes a few problems with + music on hold. * Fix issues with starting at the beginning of a + file when it shouldn't. * Fix the inuse counter to be decremented + even if the class had not been set to be deleted when not in use + anymore * Don't arbitrarily limit the number of MOH files to 255 + ........ + +2007-08-27 15:03 +0000 [r81013] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 81012 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81012 | file | 2007-08-27 12:01:59 -0300 (Mon, 27 Aug 2007) | 6 + lines (closes issue #10561) Reported by: jesselang Patches: + chan_sip-ChannelReload-20080825.patch uploaded by jesselang + (license 202) Remove an extra \r\n to make the ChannelReload + event conform with every other event. ........ + +2007-08-27 14:56 +0000 [r81011] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 81010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r81010 | mmichelson | 2007-08-27 09:55:44 -0500 (Mon, 27 Aug + 2007) | 3 lines Found a case where the queue's membercount is + off. It does not take into account dynamic members on a reload. + ........ + +2007-08-27 13:35 +0000 [r80962-80991] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Remove places that say if no language is + specified it will default to english... since on some setups this + is untrue. + + * /, main/rtp.c: Merged revisions 80974 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80974 | file | 2007-08-27 10:20:31 -0300 (Mon, 27 Aug 2007) | 4 + lines (closes issue #10562) Reported by: idkpmiller Correct + jitter value output in the CLI to be as expected. ........ + + * configs/sip.conf.sample: (closes issue #10569) Reported by: IgorG + Patches: sip_conf-80933-1.patch uploaded by IgorG (license 20) + Fix up sip.conf sample configuration. + +2007-08-26 18:12 +0000 [r80933] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 80932 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80932 | russell | 2007-08-26 13:11:26 -0500 (Sun, 26 Aug 2007) | + 3 lines Remove an extra signal_condition() for the scheduler + thread. (closes issue #10564, patch from casper) ........ + +2007-08-25 17:55 +0000 [r80821-80898] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 80895 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80895 | russell | 2007-08-25 12:37:39 -0500 (Sat, 25 Aug 2007) | + 7 lines Fix some issues with the handling of the scheduler in + chan_iax2. Most of the places that scheduled items to be executed + by the scheduler thread did not signal the scheduler thread to + wake up so that it could recalculate the time until the next + action. These changes will make the scheduler thread more + responsive and ensure that actions get executed as close to when + intended as possible instead of it being possible for very long + delays. ........ + + * pbx/pbx_dundi.c: localize a variable and remove a duplicate error + message + + * apps/app_queue.c: use ast_strlen_zero + + * /, channels/chan_iax2.c: Merged revisions 80849 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80849 | russell | 2007-08-24 16:22:50 -0500 (Fri, 24 Aug 2007) | + 5 lines If dnsmgr is in use, and no DNS servers are available + when Asterisk first starts, then don't give up on poking peers. + Allow the poke to get rescheduled so that it will work once the + dnsmgr is able to resolve the host. (closes issue #10521, patch + by jamesgolovich) ........ + + * /, main/dsp.c: Merged revisions 80820 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80820 | russell | 2007-08-24 15:24:05 -0500 (Fri, 24 Aug 2007) | + 7 lines Improve the debouncing logic in the DTMF detector to fix + some reliability issues. Previously, this code used a shift + register of hits and non-hits. However, if the start of the digit + isn't clean, it is possible for the leading edge detector to miss + the digit. These changes replace the flawed shift register logic + and also does the debouncing on the trailing edge as well. + (closes issue #10535, many thanks to softins for the patch) + ........ + +2007-08-24 20:21 +0000 [r80819] BJ Weschke <bweschke@btwtech.com> + + * apps/app_queue.c: Merged revisions 80818 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80818 | bweschke | 2007-08-24 15:52:06 -0400 (Fri, 24 Aug 2007) + | 3 lines A minor correction to the available logic of autofill. + If a queue member is paused, they're not really "available" so + don't count them as such. Somewhat related to issue #10155 + ........ + +2007-08-24 19:50 +0000 [r80817] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c: Fix documentation for Set (closes issue #10549) + +2007-08-24 19:03 +0000 [r80790] Steve Murphy <murf@digium.com> + + * main/cdr.c, /: Merged revisions 80789 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80789 | murf | 2007-08-24 12:52:15 -0600 (Fri, 24 Aug 2007) | 1 + line From a complaint by jmls, I realize that the message in + cdr_disposition is unnecessary. To get failure disposition, just + return -1; no use having more than one case do that. ........ + +2007-08-24 18:05 +0000 [r80778] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_zap.c: Add VMWI chan_zap support #9909 + +2007-08-24 15:53 +0000 [r80751] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 80750 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80750 | mmichelson | 2007-08-24 10:51:03 -0500 (Fri, 24 Aug + 2007) | 3 lines Fix a possible crash in IMAP voicemail. ........ + +2007-08-24 15:49 +0000 [r80749] Tilghman Lesher <tlesher@digium.com> + + * /: Blocked revisions 80747 via svnmerge ........ r80747 | + tilghman | 2007-08-24 10:41:43 -0500 (Fri, 24 Aug 2007) | 2 lines + Make the deprecation warning inline with the code, instead of + only in documentation (closes issue #10549) ........ + +2007-08-24 15:42 +0000 [r80748] Steve Murphy <murf@digium.com> + + * utils/conf2ael.c: fix up the MODULEINFO in conf2ael.c as well + +2007-08-24 15:29 +0000 [r80725] Russell Bryant <russell@digium.com> + + * /, utils/ael_main.c: Merged revisions 80722 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80722 | russell | 2007-08-24 10:28:05 -0500 (Fri, 24 Aug 2007) | + 3 lines Tweak the formatting of this MODULEINFO block. I think + this would have caused a "*" to get in the menuselect-tree file. + ........ + +2007-08-24 14:55 +0000 [r80690-80718] Steve Murphy <murf@digium.com> + + * /, utils/ael_main.c, utils/conf2ael.c: Merged revisions 80717 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80717 | murf | 2007-08-24 08:48:49 -0600 (Fri, 24 Aug 2007) | 1 + line This change addresses JerJer's complaint that aelparse + builds and installs even if pbx_ael is unchecked in the + menuselect stuff. ........ + + * /: Blocked 80689, the fix to ael.y; already in trunk. + +2007-08-24 11:49 +0000 [r80662] Philippe Sultan <philippe.sultan@gmail.com> + + * /, channels/chan_gtalk.c: Merged revisions 80661 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r80661 | phsultan | 2007-08-24 13:42:46 +0200 (Fri, 24 + Aug 2007) | 9 lines Closes issue #10509 Googletalk calls are + answered too early, which results in CDRs wrongly stating that a + call was ANSWERED when the calling party cancelled a call before + before being established. We must not answer the call upon + reception of a 'transport-accept' iq packet, but this packet + still needs to be acknowledged, otherwise the remote peer would + close the call (like in #8970). ........ + +2007-08-23 23:37 +0000 [r80649] Steve Murphy <murf@digium.com> + + * pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c, + res/ael/ael.y, res/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test6, + pbx/ael/ael-test/ref.ael-test7: an unreported crash I debugged, + looked like it was backing up way too far after hitting the + syntax error. An inspection of the code revealed that error + tokens in lists were not rearranged when the rules were + rearranged as part of a code neatening-up process. By moving the + error tokens to where they should be, I also reduced the number + of shift/reduce conflicts to 3 instead of 8. This introduces + subtle differences in error messages, so the regressions had to + be updated. + +2007-08-23 21:34 +0000 [r80510-80616] Russell Bryant <russell@digium.com> + + * apps/app_while.c: Use the comma separator in app_while. reported + by blitzrage on irc, patched by me + + * /, res/res_features.c, include/asterisk/features.h: Merged + revisions 80573 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80573 | russell | 2007-08-23 15:16:41 -0500 (Thu, 23 Aug 2007) | + 5 lines When executing a dynamic feature, don't look it up a + second time by digit pattern after we already looked it up by + name. This causes broken behavior if there is more than one + feature defined with the same digit pattern. (closes issue + #10539, reported by bungalow, patch by me) ........ + + * /, funcs/func_timeout.c: Merged revisions 80547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80547 | russell | 2007-08-23 14:29:44 -0500 (Thu, 23 Aug 2007) | + 3 lines Revert very broken fix for issue #10540 ... none of these + values take ms so I don't know what I was thinking ........ + + * /, funcs/func_timeout.c: Merged revisions 80539 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80539 | russell | 2007-08-23 14:21:53 -0500 (Thu, 23 Aug 2007) | + 4 lines Fix func_timeout to take values in floating point so 1.5 + actually means 1.5 seconds instead of being rounded. (closes + issue #10540, reported by spendergrass, patch by me) ........ + + * doc/asterisk-mib.txt, res/snmp/agent.c: Fix a typo in the + Asterisk MIB and fix astNumChanBridged so it acts as a counter + again (closes issue #10118, patch by jeffg) + +2007-08-23 17:18 +0000 [r80508] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_zap.c, /: Merged revisions 80501 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80501 | kpfleming | 2007-08-23 12:08:25 -0500 (Thu, 23 Aug 2007) + | 2 lines report the actual channel number that was unregistered, + instead of assuming that the interface list consists of channels + 1 through <x> with no gaps in the sequence ........ + +2007-08-23 17:04 +0000 [r80470-80500] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 80499 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80499 | russell | 2007-08-23 12:02:50 -0500 (Thu, 23 Aug 2007) | + 3 lines Fix some code where it was possible for a reference to a + peer to not get released when it should. Thank you to Marta + Carbone for pointing this out! ........ + + * /: Blocked revisions 80497 via svnmerge ........ r80497 | russell + | 2007-08-23 11:53:52 -0500 (Thu, 23 Aug 2007) | 5 lines This is + a hack to maintain old behavior of chan_iax2. This ensures that + if the peers and users are being stored in a linked list, that + they go in the list in the same order that the older code used. + This is necessary to maintain the behavior of which peers and + users get matched when traversing the container. ........ + + * /, res/res_agi.c: Merged revisions 80469 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80469 | russell | 2007-08-23 10:49:28 -0500 (Thu, 23 Aug 2007) | + 2 lines Revert res_agi fix that didn't quite work until we get it + right ... ........ + +2007-08-23 15:48 +0000 [r80453-80468] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: If no default language has been specified + print out that it will default to english when using sip show + peer or sip show user. + + * main/minimime/mm.h: Return trunk to a working state by including + compat.h in minimime. + +2007-08-22 23:26 +0000 [r80428-80429] Jason Parker <jparker@digium.com> + + * main/minimime/mm_util.c, main/minimime/mm_codecs.c, + main/minimime/mm_mem.h, main/minimime/mm_base64.c, + main/minimime/mm.h: Convert minimime to use the proper uint*_t + types, rather than u_int*_t + + * apps/app_minivm.c: Cast calls to getpid. This was done in 1.4 + already, this one was just new + +2007-08-22 22:54 +0000 [r80361-80427] Russell Bryant <russell@digium.com> + + * /, include/asterisk/astobj2.h: Merged revisions 80426 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80426 | russell | 2007-08-22 17:54:03 -0500 (Wed, 22 Aug 2007) | + 6 lines Add some more documentation on iterating ao2 containers. + The documentation implies that is possible to miss an object or + see an object twice while iterating. After looking through the + code and talking with mmichelson, I have documented the exact + conditions under which this can happen (which are rare and + harmless in most cases). ........ + + * /, main/astobj2.c: Merged revisions 80424 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80424 | russell | 2007-08-22 17:40:27 -0500 (Wed, 22 Aug 2007) | + 10 lines When converting this code to use the list macros, I + changed it so objects are added to the head of a bucket instead + of the tail. However, while looking over code with mmichelson, we + noticed that the algorithm used in ao2_iterator_next requires + that items are added to the tail. This wouldn't have caused any + huge problem, but it wasn't correct. It meant that if an object + was added to a container while you were iterating it, and it was + added to the same bucket that the current element is in, then the + new object would be returned by ao2_iterator_next, and any other + objects in the bucket would be bypassed in the traversal. + ........ + + * channels/chan_iax2.c: allow peers and users to go into a hash + table + + * /, channels/chan_sip.c: Merged revisions 80390 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80390 | russell | 2007-08-22 16:00:44 -0500 (Wed, 22 Aug 2007) | + 3 lines Don't crash when using realtime in chan_sip without an + insecure setting in the database. (closes issue #10348, reported + by link55, fixed by me) ........ + + * channels/chan_iax2.c: Unsubscribe from MWI events in the peer + destructor + + * /, main/Makefile, include/asterisk/astobj2.h (added), + include/asterisk/strings.h, channels/chan_iax2.c, main/astobj2.c + (added): Merged revisions 80362 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80362 | russell | 2007-08-22 15:21:36 -0500 (Wed, 22 Aug 2007) | + 34 lines Merge changes from team/russell/iax_refcount. This set + of changes fixes problems with the handling of iax2_user and + iax2_peer objects. It was very possible for a thread to still + hold a reference to one of these objects while a reload operation + tries to delete them. The fix here is to ensure that all + references to these objects are tracked so that they can't go + away while still in use. To accomplish this, I used the astobj2 + reference counted object model. This code has been in one of + Luigi Rizzo's branches for a long time and was primarily + developed by one of his students, Marta Carbone. I wanted to go + ahead and bring this in to 1.4 because there are other problems + similar to the ones fixed by these changes, so we might as well + go ahead and use the new astobj if we're going to go through all + of the work necessary to fix the problems. As a nice side benefit + of these changes, peer and user handling got more efficient. + Using astobj2 lets us not hold the container lock for peers or + users nearly as long while iterating. Also, by changing a define + at the top of chan_iax2.c, the objects will be distributed in a + hash table, drastically increasing lookup speed in these + containers, which will have a very big impact on systems that + have a large number of users or peers. The use of the hash table + will be made the default in trunk. It is not the default in 1.4 + because it changes the behavior slightly. Previously, since peers + and users were stored in memory in the same order they were + specified in the configuration file, you could influence peer and + user matching order based on the order they are specified in the + configuration. The hash table does not guarantee any order in the + container, so this behavior will be going away. It just means + that you have to be a little more careful ensuring that peers and + users are matched explicitly and not forcing chan_iax2 to have to + guess which user is the right one based on secret, host, and + access list settings, instead of simply using the username. If + you have any questions, feel free to ask on the asterisk-dev + list. ........ + + * /, res/res_agi.c: Merged revisions 80360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80360 | russell | 2007-08-22 14:53:30 -0500 (Wed, 22 Aug 2007) | + 5 lines Juggie in #asterisk-dev was reporting problems where + fgets would return without reading the whole line when using + fastagi. When this happens, errno was set to EINTR or EAGAIN. + This patch accounts for the possibility and lets fgets continue + in that case. ........ + +2007-08-22 18:54 +0000 [r80303-80331] Jason Parker <jparker@digium.com> + + * Makefile, build_tools/mkpkgconfig, /, build_tools/make_build_h, + build_tools/strip_nonapi, build_tools/prep_moduledeps, + build_tools/make_buildopts_h: Merged revisions 80330 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r80330 | qwell | 2007-08-22 13:53:18 -0500 (Wed, 22 Aug + 2007) | 7 lines Fix a few build issues in Solaris (and likely + others). Use GREP and ID variables from autoconf. Reported to me + in #asterisk-dev I forgot who reported this - sorry. :( ........ + + * Makefile, /: Merged revisions 80304 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80304 | qwell | 2007-08-22 13:25:34 -0500 (Wed, 22 Aug 2007) | 2 + lines Change a syntax that the GNU make in Solaris dislikes. + ........ + + * /, build_tools/make_version: Merged revisions 80302 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r80302 | qwell | 2007-08-22 13:06:00 -0500 (Wed, 22 Aug + 2007) | 3 lines Fix a bashism (we explicitly request /bin/sh). + Remove some oddly placed quotes I found in passing. ........ + +2007-08-22 16:27 +0000 [r80258-80262] Russell Bryant <russell@digium.com> + + * utils/check_expr.c: Ensure that the object code for + ast_atomic_fetchadd_int() gets included in the check_expr binary + when building with LOW_MEMORY defined. (reported by Brian Capouch + on the asterisk-dev list, patch by me) + + * Makefile, /: Merged revisions 80257 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80257 | russell | 2007-08-22 11:21:58 -0500 (Wed, 22 Aug 2007) | + 4 lines Honor the contents of the COPTS variable as custom target + CFLAGS. Apparently this is what openwrt does. (reported by Brian + Capouch on the asterisk-dev list, patch by me) ........ + +2007-08-22 16:16 +0000 [r80256] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 80255 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80255 | file | 2007-08-22 13:14:38 -0300 (Wed, 22 Aug 2007) | 4 + lines (closes issue #10526) Reported by: sinistermidget Revert + commit from issue #10355 and return timestamp skew to 640. + ........ + +2007-08-22 14:17 +0000 [r80241-80242] Steve Murphy <murf@digium.com> + + * /: blocking 80167 + + * /, main/alaw.c: Merged revisions 80166 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80166 | murf | 2007-08-21 10:36:34 -0600 (Tue, 21 Aug 2007) | 1 + line This patch solves problem 1 in 8126; it should not slow down + the alaw codec, but should prevent signal degradation via + multiple trips thru the codec. Fossil estimates the twice thru + this codec will prevent fax from working. 4-6 times thru would + result hearable, noticeable, voice degradation. ........ + +2007-08-21 21:58 +0000 [r80226] Russell Bryant <russell@digium.com> + + * funcs/func_odbc.c: use ast_atomic_fetchadd_int for incrementing + resultcount + +2007-08-21 20:55 +0000 [r80217] Steve Murphy <murf@digium.com> + + * res/ael/pval.c: As per 10472, mvanbaak thought the generated code + would look better this way. + +2007-08-21 18:49 +0000 [r80184] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 80183 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80183 | russell | 2007-08-21 13:42:15 -0500 (Tue, 21 Aug 2007) | + 7 lines Don't record SIP dialog history if it's not turned on. + Also, put an upper limit on how many history entires will be + stored for each SIP dialog. It is currently set to 50, but can be + increased if deemed necessary. (closes issue #10421, closes issue + #10418, patches suggested by jmoldenhauer, patches updated by me) + (Security implications documented in AST-2007-020) ........ + +2007-08-21 15:51 +0000 [r80157] Joshua Colp <jcolp@digium.com> + + * main/audiohook.c: Minor tweak. Don't manipulate volume of the + audio in the buffer if no audio is actually there. + +2007-08-21 15:23 +0000 [r80133] Russell Bryant <russell@digium.com> + + * /, channels/chan_mgcp.c: Merged revisions 80132 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80132 | russell | 2007-08-21 10:22:22 -0500 (Tue, 21 Aug 2007) | + 3 lines Don't try to dereference the owner channel when it may + not exist (issue #10507, maxper) ........ + +2007-08-21 15:04 +0000 [r80131] Jason Parker <jparker@digium.com> + + * /, configs/cdr.conf.sample: Merged revisions 80130 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r80130 | qwell | 2007-08-21 10:03:45 -0500 (Tue, 21 Aug + 2007) | 7 lines (closes issue #10510) Reported by: casper + Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few + errors in sample cdr config file. ........ + +2007-08-20 22:53 +0000 [r80113] Steve Murphy <murf@digium.com> + + * build_tools/cflags.xml, main/ulaw.c, codecs/slin_ulaw_ex.h, + codecs/ulaw_slin_ex.h, include/asterisk/alaw.h, main/translate.c, + include/asterisk/ulaw.h, main/alaw.c: This change set fixes bug + 8126 in trunk. It is implemented via compile time options, + activated via the menuselect stuff, which defaults to the old + way. non-zero sample data added. Translate tables expressed in + microseconds instead of milliseconds, with 5-digit data now + instead of 3, giving 2 more digits of precision. + +2007-08-20 22:00 +0000 [r80089] Russell Bryant <russell@digium.com> + + * /: Blocked revisions 80088 via svnmerge ........ r80088 | russell + | 2007-08-20 16:57:08 -0500 (Mon, 20 Aug 2007) | 2 lines Fix the + build of app_queue ........ + +2007-08-20 21:42 +0000 [r80087] Mark Michelson <mmichelson@digium.com> + + * /: Blocked revisions 80086 via svnmerge ........ r80086 | + mmichelson | 2007-08-20 16:39:17 -0500 (Mon, 20 Aug 2007) | 5 + lines After a discussion on #asterisk-dev, it was decided that + this should be in 1.4 as well. (issue #10424, reported and + patched by irroot) ........ + +2007-08-20 17:37 +0000 [r80075] Steve Murphy <murf@digium.com> + + * include/asterisk/lock.h, utils/extconf.c: Stephn Davies reports + that this will help make things work on 64-bit machines + +2007-08-20 16:18 +0000 [r80050] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 80049 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80049 | mmichelson | 2007-08-20 11:17:43 -0500 (Mon, 20 Aug + 2007) | 4 lines Found a pointless ternary if. member->dynamic was + set to 1 and has no opportunity to change between then and this + line, so "dynamic" will ALWAYS be output. ........ + +2007-08-20 16:12 +0000 [r80048] Jason Parker <jparker@digium.com> + + * /, configs/extensions.conf.sample: Merged revisions 80047 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80047 | qwell | 2007-08-20 11:08:49 -0500 (Mon, 20 Aug 2007) | 7 + lines (closes issue #10499) Reported by: casper Patches: + extensions.conf.sample.diff uploaded by casper (license 55) + Update CLI examples in extensions.conf.sample to reflect command + changes. ........ + +2007-08-20 15:53 +0000 [r80046] Joshua Colp <jcolp@digium.com> + + * apps/app_voicemail.c: Remove remnants of last commit so trunk + builds again. + +2007-08-20 15:37 +0000 [r80045] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 80044 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r80044 | mmichelson | 2007-08-20 10:34:43 -0500 (Mon, 20 Aug + 2007) | 5 lines Ukrainian language voicemail support. (closes + issue #10458, reported and patched by Oleh) ........ + +2007-08-20 15:27 +0000 [r80037] Steve Murphy <murf@digium.com> + + * utils/pval.c (removed): pval.c should not be in svn, in the utils + dir + +2007-08-20 15:10 +0000 [r80023-80033] Joshua Colp <jcolp@digium.com> + + * utils/pval.c: Bring pval.c in utils up to date with pval.c in + res/ael. + + * channels/chan_zap.c: Fix random segfault issue when loading + chan_zap. Trying to access a configuration structure that has + already been destroyed is bad, mmmk? + +2007-08-20 02:46 +0000 [r79999] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 79998 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79998 | tilghman | 2007-08-19 21:42:49 -0500 (Sun, 19 Aug 2007) + | 2 lines Missing curly braces. Oops. (Reported by snuffy via + IRC) ........ + +2007-08-20 00:54 +0000 [r79988-79990] Joshua Colp <jcolp@digium.com> + + * channels/chan_iax2.c: (closes issue #10495) Reported by: + stevedavies Make sure context pointer is valid or else chan_iax2 + will go kaboom. + + * utils/Makefile: (closes issue #10496) Reported by: caio1982 Fix + building on OSX. + + * channels/chan_h323.c: Fix building of trunk. I'm doing work on a + Sunday night just to avoid watching Snakes on a Plane which my + roommate is watching. + +2007-08-19 14:17 +0000 [r79980] Tilghman Lesher <tlesher@digium.com> + + * utils/Makefile: Add strcompat dependency for check_expr (needed + for platforms that don't have strndup) + +2007-08-18 23:58 +0000 [r79972] Joshua Colp <jcolp@digium.com> + + * configure, configure.ac: Actually check the return value of + epoll_create to make sure it works. + +2007-08-18 14:34 +0000 [r79940-79949] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 79947 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79947 | tilghman | 2007-08-18 09:30:44 -0500 (Sat, 18 Aug 2007) + | 3 lines Don't allocate vmu for messagecount when we could just + use the stack instead (closes issue #10490) Also, remove a + useless (and leaky) SQLAllocHandle (closes issue #10480) ........ + + * channels/chan_zap.c, channels/chan_sip.c, channels/chan_h323.c, + channels/chan_iax2.c: We weren't properly encapsulating the mtime + ignores of config files (closes issue #10488) + +2007-08-17 21:19 +0000 [r79915] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: I broke the build. Now I'm fixing it. + +2007-08-17 21:04 +0000 [r79913] Russell Bryant <russell@digium.com> + + * channels/chan_zap.c, /: Merged revisions 79912 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79912 | russell | 2007-08-17 16:01:43 -0500 (Fri, 17 Aug 2007) | + 4 lines Avoid a crash in the handling of DTMF based Caller ID. It + is valid for ast_read to return NULL in the case that the channel + has been hung up. (crash reported by anonymouz666 on IRC in + #asterisk-dev) ........ + +2007-08-17 19:16 +0000 [r79907] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 79906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79906 | mmichelson | 2007-08-17 14:14:05 -0500 (Fri, 17 Aug + 2007) | 6 lines Patch allows for more seamless transition from + file storage voicemail to ODBC storage voicemail. If a retrieval + of a greeting from the database fails, but the file is found on + the file system, then we go ahead an insert the greeting into the + database. The result of this is that people who switch from file + storage to ODBC storage do not need to rerecord their voicemail + greetings. ........ + +2007-08-17 19:13 +0000 [r79903-79905] Jason Parker <jparker@digium.com> + + * /, channels/chan_sip.c, main/utils.c, include/asterisk/strings.h: + Merged revisions 79904 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #10430) ........ r79904 | qwell | 2007-08-17 14:12:19 -0500 + (Fri, 17 Aug 2007) | 11 lines Don't send a semicolon over the + wire in sip notify messages. Caused by fix for issue 9938. I + basically took the code that existed before 9938 was fixed, and + copied it into a new function - ast_unescape_semicolon There + should be very few places this will be needed (pbx_config does + NOT need this (see issue 9938 for details)) Issue 10430, patch by + me, with help/ideas from murf (thanks murf). ........ + + * channels/chan_local.c, /: Merged revisions 79902 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes + issue #10485) ........ r79902 | qwell | 2007-08-17 12:44:22 -0500 + (Fri, 17 Aug 2007) | 4 lines Re-add the setting of callerid name + and number. Issue 10485, reported by and fix explained by + paradise. ........ + +2007-08-17 16:39 +0000 [r79901] Tilghman Lesher <tlesher@digium.com> + + * configs/logger.conf.sample: Documentation for %q in logger.conf, + as suggested by jtodd (closes issue #10475) + +2007-08-17 16:04 +0000 [r79888-79894] Jason Parker <jparker@digium.com> + + * res/res_features.c: Fix Dial arguments in res_features. Closes + issue #10484, patch by lunn. + + * pbx/pbx_dundi.c: Correct the argument separator for a Dial + statement in pbx_dundi. Closes issue #10483, patch by lunn + +2007-08-17 14:41 +0000 [r79885] Tilghman Lesher <tlesher@digium.com> + + * main/config.c: Change this flag... might not otherwise unlock in + an OOM situation + +2007-08-17 14:14 +0000 [r79861-79862] Russell Bryant <russell@digium.com> + + * channels/chan_iax2.c: Make use of ast_sched_replace() in some + places in chan_iax2 + + * channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: This + commit adds a scheduler API call, ast_sched_replace that can be + used in place of a very common construct. I also used it in a + number of places in chan_sip. if (id > -1) ast_sched_del(sched, + id); id = ast_sched_add(sched, ...); changes to: + ast_sched_replace(id, sched, ...); + +2007-08-17 13:45 +0000 [r79859-79860] Tilghman Lesher <tlesher@digium.com> + + * res/res_config_odbc.c, res/res_config_sqlite.c: store and destroy + implementations for sqlite (closes issue #10446) and odbc (closes + issue #10447) + + * res/res_config_pgsql.c, funcs/func_lock.c: store and destroy + implementations for realtime pgsql (closes issue #10372) + +2007-08-17 13:39 +0000 [r79858] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 79857 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79857 | russell | 2007-08-17 08:37:08 -0500 (Fri, 17 Aug 2007) | + 5 lines Fix some crashes in chan_sip. This patch changes various + places that add items to the scheduler to ensure that they don't + overwrite the ID of a previously scheduled item. If there is one, + it should be removed. (closes issue #10391, closes issue #10256, + probably others, patch by me) ........ + +2007-08-17 08:29 +0000 [r79841] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /: Merged revisions 79833 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r79833 | crichter | 2007-08-17 10:22:36 +0200 (Fr, 17 + Aug 2007) | 1 line sometimes we don't need to signal dtmf tones + to asterisk, we just want them to go through as inband. Otherwise + they might be generated by the other channel partner and then + there is a double tone. ........ + +2007-08-17 01:19 +0000 [r79824] Joshua Colp <jcolp@digium.com> + + * channels/chan_zap.c: Fix building of chan_zap under development + mode without libpri and libss7 installed. + +2007-08-16 23:31 +0000 [r79813] Tilghman Lesher <tlesher@digium.com> + + * funcs/func_lock.c: Revise dialplan locks to permit multiple locks + per channel, but with deadlock avoidance + +2007-08-16 22:33 +0000 [r79764-79794] Russell Bryant <russell@digium.com> + + * /: Merged revisions 79792 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79792 | russell | 2007-08-16 17:32:33 -0500 (Thu, 16 Aug 2007) | + 4 lines Fix a little race condition that could cause a crash if + two channels had MOH stopped at the same time that were using a + class that had been marked for deletion when its use count hits + zero. ........ + + * /, res/res_musiconhold.c: Merged revisions 79778 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r79778 | russell | 2007-08-16 17:24:25 -0500 (Thu, 16 + Aug 2007) | 14 lines This patch fixes a bug where reloading the + module with "module reload" did not delete classes from memory + that were no longer in the config. This patch fixes that problem + as well as another one. Previously, if you reloaded MOH using the + "moh reload" CLI command, which behaved differently than "module + reload ...", MOH had to be stopped on every channel and started + again immediately. However, there was no way to tell what class + was being used, so they would all fall back to the default class. + (closes issue #10139) Reported by: blitzrage Patches: + asterisk-10139-advanced.diff.txt uploaded by jamesgolovich + (license 176) Tested by: jamesgolovich ........ + + * /, channels/chan_iax2.c: Merged revisions 79756 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79756 | russell | 2007-08-16 16:29:24 -0500 (Thu, 16 Aug 2007) | + 11 lines Fix more deadlocks in chan_iax2 that were introduced by + making frame handling and scheduling multi-threaded. + Unfortunately, we have to do some expensive deadlock avoidance + when queueing frames on to the ast_channel owner of the IAX2 pvt + struct. This was already handled for regular frames, but + ast_queue_hangup and ast_queue_control were still used directly. + Making these changes introduced even more places where the IAX2 + pvt struct can disappear in the context of a function holding its + lock due to calling a function that has to unlock/lock it to + avoid deadlocks. I went through and fixed all of these places to + account for this possibility. (issue #10362, patch by me) + ........ + +2007-08-16 21:28 +0000 [r79755] Joshua Colp <jcolp@digium.com> + + * /: Fix properties on trunk again. + +2007-08-16 21:21 +0000 [r79749] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 79748 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r79748 | mmichelson | 2007-08-16 16:16:40 -0500 (Thu, 16 + Aug 2007) | 8 lines Fixes a problem where agents would get stuck + busy due to their wrapuptime being longer than the queue's + wrapuptime and ringinuse=no for the queue. (closes issue #10215, + reported by Doug, repaired by me) Special thanks to fkasumovic + for pointing out the source of the problem and to bweschke for + helping to come up with a solution! ........ + +2007-08-16 21:09 +0000 [r79747] Tilghman Lesher <tlesher@digium.com> + + * main/udptl.c, cdr/cdr_sqlite3_custom.c, /, res/res_features.c, + codecs/codec_adpcm.c, apps/app_alarmreceiver.c, + cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c, main/config.c, + main/loader.c, res/res_smdi.c, channels/chan_skinny.c, + main/http.c, apps/app_amd.c, channels/chan_alsa.c, + cdr/cdr_odbc.c, cdr/cdr_manager.c, codecs/codec_g722.c, + apps/app_privacy.c, codecs/codec_speex.c, channels/chan_agent.c, + codecs/codec_g726.c, channels/iax2-provision.c, + apps/app_playback.c, channels/iax2-provision.h, + channels/chan_misdn.c, res/res_indications.c, pbx/pbx_config.c, + main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c, + channels/chan_vpb.cc, res/res_snmp.c, apps/app_meetme.c, + codecs/codec_gsm.c, res/res_musiconhold.c, channels/chan_gtalk.c, + cdr/cdr_pgsql.c, apps/app_followme.c, res/res_jabber.c, + cdr/cdr_radius.c, codecs/codec_zap.c, res/res_config_sqlite.c, + main/enum.c, channels/misdn_config.c, cdr/cdr_csv.c, main/cdr.c, + channels/chan_phone.c, res/res_config_odbc.c, main/manager.c, + apps/app_osplookup.c, funcs/func_odbc.c, apps/app_minivm.c, + main/logger.c, apps/app_directory.c, apps/app_rpt.c, + cdr/cdr_custom.c, channels/chan_mgcp.c, codecs/codec_lpc10.c, + res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, + channels/chan_sip.c, apps/app_festival.c, codecs/codec_alaw.c, + res/res_adsi.c, include/asterisk/config.h, apps/app_queue.c, + channels/chan_oss.c, main/rtp.c, cdr/cdr_tds.c, + channels/chan_jingle.c, channels/misdn/chan_misdn_config.h, + channels/chan_h323.c, pbx/pbx_dundi.c, codecs/codec_ulaw.c: Don't + reload a configuration file if nothing has changed. + +2007-08-16 19:40 +0000 [r79736] Steve Murphy <murf@digium.com> + + * utils/pval.c, utils/conf2ael.c: Many thanks to mvanbaak for his + update to translate hints; I added the -d option for local + testing purposes. This is from bug 10472 + +2007-08-16 18:23 +0000 [r79724-79725] Dwayne M. Hubbard <dhubbard@digium.com> + + * channels/chan_iax2.c: added counter for iax2 show registry CLI + output, closes issue 10461, thanks junky + + * apps/app_voicemail.c: added counter for voicemail show users, + issue 10462, thanks junky + +2007-08-16 17:34 +0000 [r79714-79719] Steve Murphy <murf@digium.com> + + * utils/conf2ael.c: mvanbaak asks: why did you include that twice? + Answer: dunno. removed redundant include + + * utils/extconf.c, utils/conf2ael.c: svn did me dirty for some + reason. Left 5 files out of the commit; Tilghman copied them in + from the branch, but I had made changes to these. Here they are. + +2007-08-16 15:59 +0000 [r79691] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 79690 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79690 | mmichelson | 2007-08-16 10:58:34 -0500 (Thu, 16 Aug + 2007) | 5 lines base_encode is not trying to open a log file, so + we should not call it a log file in the warning. (related to + issue #10452, reported by bcnit) ........ + +2007-08-16 15:29 +0000 [r79687-79688] Joshua Colp <jcolp@digium.com> + + * pbx/pbx_dundi.c: (closes issue #10467) Reported by: lunn Patches: + pbx_dundi.diff uploaded by lunn (license 179) Don't print a + warning saying an ethernet interface was found when it indeed + was. + + * utils/conf2ael.c: Make conf2ael build on 64-bit systems. + +2007-08-16 09:45 +0000 [r79666] Philippe Sultan <philippe.sultan@gmail.com> + + * /, res/res_jabber.c: Merged revisions 79665 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79665 | phsultan | 2007-08-16 11:37:10 +0200 (Thu, 16 Aug 2007) + | 21 lines A fix for two critical problems detected while working + with Daniel McKeehan in issue #10184. Upon priority change, the + resource list is not NULL terminated when moving an item to the + end of the list. This makes Asterisk endlessy loop whenever it + needs to read the list. Jids with different resource and priority + values, like in Gmail's and GoogleTalk's jabber clients put that + problem in evidence. Upon reception of a 'from' attribute with an + empty resource string, Asterisk crashes when trying to access the + found->cap pointer if the resource list for the given buddy is + not empty. This situation is perfectly valid and must be handled. + The Gizmoproject's jabber client put that problem in evidence. + Also added a few comments in the code as well as a handle for the + capabilities from Gmail's jabber client, which are stored in a + caps:c tag rather than the usual c tag. Closes issue #10184. + ........ + +2007-08-16 09:22 +0000 [r79660] Christian Richter <christian.richter@beronet.com> + + * /, channels/misdn/ie.c: Merged revisions 79642 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79642 | crichter | 2007-08-16 10:21:21 +0200 (Do, 16 Aug 2007) | + 1 line 0x80 + protocol is wrong for USERUSER when we want to send + IA5 Chars. ........ + +2007-08-16 06:52 +0000 [r79638] Olle Johansson <oej@edvina.net> + + * CHANGES: Doc change + +2007-08-15 22:53 +0000 [r79634] Jason Parker <jparker@digium.com> + + * res/res_musiconhold.c: Modify the names of functions/variables in + res_musiconhold to be useful. Closes issue #10464, patch by + caio1982 + +2007-08-15 21:25 +0000 [r79623] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/pval.h (added), utils/pval.c (added), + include/asterisk/extconf.h (added), utils/extconf.c (added), + utils/conf2ael.c (added): Missing from murf's last trunk commit, + which was why trunk won't compile + +2007-08-15 19:34 +0000 [r79611] Joshua Colp <jcolp@digium.com> + + * /: Remove properties that appeared from Steve's last branch + merge. Automerge has already run so everyone's branches based off + of trunk are probably toast by now. + +2007-08-15 19:21 +0000 [r79595] Steve Murphy <murf@digium.com> + + * /, pbx/ael/ael.y (removed), pbx/ael/ael-test/ref.ael-test11, + res/Makefile, pbx/ael/ael-test/ref.ael-test14, + pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-test16, + pbx/ael/ael-test/ref.ael-test19, include/asterisk/ast_expr.h, + pbx/ael/ael_lex.c (removed), pbx/pbx_ael.c, pbx/ael/ael.flex + (removed), res/ael (added), main/pbx.c, UPGRADE.txt, + res/res_ael_share.c (added), pbx/Makefile, CHANGES, + utils/Makefile, pbx/ael/ael-test/ref.ael-ntest10, + pbx/ael/ael.tab.c (removed), pbx/ael/ael-test/ref.ael-test1, + pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, + pbx/ael/ael-test/ref.ael-test4, include/asterisk/ael_structs.h, + pbx/ael/ael.tab.h (removed), pbx/ael/ael-test/ref.ael-test5, + utils/ael_main.c, include/asterisk/pbx.h, + pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-test7, + utils/check_expr.c: This commit closes bug 7605, and half-closes + 7638. The AEL code has been redistributed/repartitioned to allow + code re-use both inside and outside of Asterisk. This commit + introduces the utils/conf2ael program, and an external + config-file reader, for both normal config files, and for + extensions.conf (context, exten, prio); It provides an API for + programs outside of asterisk to use to play with the dialplan and + config files. + +2007-08-15 14:42 +0000 [r79558] Joshua Colp <jcolp@digium.com> + + * /, main/rtp.c: Merged revisions 79553 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79553 | file | 2007-08-15 11:40:23 -0300 (Wed, 15 Aug 2007) | 6 + lines (closes issue #10440) Reported by: irroot (closes issue + #10454) Reported by: flo_turc Increase maximum timestamp skew to + 120. 20 was apparently far too low. ........ + +2007-08-15 14:27 +0000 [r79529] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_voicemail.c: Merged revisions 79527 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79527 | mmichelson | 2007-08-15 09:26:40 -0500 (Wed, 15 Aug + 2007) | 5 lines Fixed an error in the Russian language voicemail + intro. (issue #10458, reported and patched by Oleh) ........ + +2007-08-15 14:20 +0000 [r79524] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 79523 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79523 | file | 2007-08-15 11:18:44 -0300 (Wed, 15 Aug 2007) | 6 + lines (closes issue #10456) Reported by: irroot Patches: + sip_timeout.patch uploaded by irroot (license 52) Change + hardcoded timer value to defined value. I'm doing this in 1.4 as + well so if it needs to be changed in the future this place would + not have been forgotten. ........ + +2007-08-15 11:27 +0000 [r79507] Christian Richter <christian.richter@beronet.com> + + * channels/chan_misdn.c, /, channels/misdn/ie.c, + channels/misdn/isdn_msg_parser.c: Merged revisions 78936 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r78936 | crichter | 2007-08-10 15:24:03 +0200 (Fr, 10 Aug 2007) | + 1 line fixed a bug with the useruser information element. We send + them now also in the disconnect message. ........ + +2007-08-14 18:50 +0000 [r79437-79471] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 79470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79470 | russell | 2007-08-14 13:49:10 -0500 (Tue, 14 Aug 2007) | + 2 lines Fix another spot where an iax2_peer would be leaked if + realtime was in use. ........ + + * /, channels/chan_iax2.c: Merged revisions 79436 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79436 | russell | 2007-08-14 12:31:39 -0500 (Tue, 14 Aug 2007) | + 3 lines Fix some memory leaks throughout chan_iax2 related to the + use of realtime. I found these while working on iax2_peer object + reference tracking. ........ + +2007-08-14 15:30 +0000 [r79403] Joshua Colp <jcolp@digium.com> + + * /, res/res_features.c: Merged revisions 79397 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r79397 | file | 2007-08-14 12:27:13 -0300 (Tue, 14 Aug 2007) | 4 + lines (closes issue #10415) Reported by: atis Revert fix for + #10327 as it causes more issues then it solves. ........ + +2007-08-14 14:32 +0000 [r79392] Steve Murphy <murf@digium.com> + + * pbx/ael/ael-test/ref.ael-vtest17, /, + pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, + pbx/ael/ael-test/ael-test5/extensions.ael, + pbx/ael/ael-test/ael-test6/extensions.ael, + pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ael-vtest21/extensions.ael, |