From e745edd4e84590f50fe8335d231a63e7cfd4c46b Mon Sep 17 00:00:00 2001 From: russell Date: Tue, 10 Jul 2007 16:08:16 +0000 Subject: importing files for 1.4.7.1 release git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.7.1@74333 f38db490-d61c-443f-a65b-d21fe96a405b --- .lastclean | 1 + .version | 1 + ChangeLog | 9354 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 9356 insertions(+) create mode 100644 .lastclean create mode 100644 .version create mode 100644 ChangeLog diff --git a/.lastclean b/.lastclean new file mode 100644 index 000000000..9902f1784 --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +28 diff --git a/.version b/.version new file mode 100644 index 000000000..89c9c2249 --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.4.7.1 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..1240a80b2 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,9354 @@ +2007-07-10 Russell Bryant + + * Asterisk 1.4.7.1 released. + +2007-07-10 16:00 +0000 [r74323] Russell Bryant + + * res/res_musiconhold.c: fix an uninitialized variable + +2007-07-10 15:38 +0000 [r74317] Jason Parker + + * apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4 + lines Fix a small typo in description in of Voicemail() + application. Issue 10170, patch by casper. ........ + +2007-07-10 15:31 +0000 [r74314] Russell Bryant + + * res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10 + Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue + #10075, this part reported by jmls on IRC, patch by me) ........ + +2007-07-10 14:50 +0000 [r74262-74265] Joshua Colp + + * /, main/app.c: Merged revisions 74264 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2 + lines Ensure the group information category exists before trying + to do a string comparison with it. (issue #10171 reported by + mlegas) ........ + + * channels/chan_sip.c: Only spit out an inringing warning message + when it is applicable. Since call limits are already toast in + realtime let's not scare the user if they are using it. (issue + #10166 reported by bcnit) + +2007-07-09 Russell Bryant + + * Asterisk 1.4.7 released. + +2007-07-09 21:31 +0000 [r74162-74211] Russell Bryant + + * configure, configure.ac: Update the configure script to check for + a required function that is not present in the 1.2 version of + libpri. This will prevent the configure script from thinking that + it has compatible libpri support for Asterisk 1.4, when it + actually does not because the installed version is from 1.2. + + * /: Blocked revisions 74165 via svnmerge ........ r74165 | russell + | 2007-07-09 16:00:17 -0500 (Mon, 09 Jul 2007) | 4 lines When the + specified class isn't found, properly fall back to the channel's + music class or the default. (issue #10123, reported by blitzrage, + patches from juggie, qwell, and me) ........ + + * res/res_musiconhold.c: (closes issue #10123) Reported by: + blitzrage Patches submitted by: juggie, qwell, me Tested by: + blitzrage When trying to find a music on hold class to use, try + all of the options, instead of only the first one that is set. + Also, change the MusicOnHold applications to not hang up on the + channel when a class can not be found. + +2007-07-09 20:19 +0000 [r74159] Jason Parker + + * channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8 + lines Several chan_zap options were not working on reload because + they were arbitrarily disallowed when reloading some/most PRI + options (such as signalling) was disallowed. Options such as + polarityonanswerdelay and answeronpolarityswitch can safely be + changed on a reload. This corrects that behavior. Issue 9186, + patch by tzafrir. ........ + +2007-07-09 18:38 +0000 [r74120-74122] Mark Michelson + + * apps/app_queue.c: Forgot to get rid of an extraneous debug + message. + + * apps/app_queue.c: The n option for Queue should make the queue + exit immediately after failure to reach any members and should + not be dependent on the timeout value passed to Queue (closes + issue #10127, reported by bcnit, repaired by me) + +2007-07-09 15:32 +0000 [r74082] Joshua Colp + + * channels/chan_skinny.c: Only destroy the scheduler context if it + was allocated. (issue #10124 reported by gzero) + +2007-07-09 14:57 +0000 [r74047] Mark Michelson + + * apps/app_voicemail.c: Fixed a logic error in leave_voicemail. + Pass the mailbox instead of the context to inbox_count when the + context is "default." (closes issue #10135, reported by yannj, + repaired by me) + +2007-07-09 14:49 +0000 [r74043-74045] Joshua Colp + + * channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread + synchronization tweaks. (issue #10124 reported by gzero) + + * configure, acinclude.m4: Use AC_CHECK_HEADER to check for + ptlib/openh323 to allow for cross compiling. (issue #9675 + reported by zandbelt) + +2007-07-09 04:03 +0000 [r73985] Tilghman Lesher + + * main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while + 'make progdocs'. (Closes issue #10104) + +2007-07-09 03:13 +0000 [r73930-73980] Joshua Colp + + * main/cdr.c: Give Agent channel names priority when doing CDR + merging. (issue #10011 reported by krtorio) + + * pbx/pbx_config.c: Add a few sanity checks when writing out the + dialplan. (issue #10157 reported by dome) + +2007-07-08 09:47 +0000 [r73849] Olle Johansson + + * channels/chan_sip.c: While tracking down a bug, I need some more + history. Dumphistory is very useful, indeed. + +2007-07-06 23:02 +0000 [r73769] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | + 4 lines If a sip_pvt struct has already registered an extension + state callback, remove the old one before adding a new one. If + this isn't done, Asterisk will crash. (issue #10120) ........ + +2007-07-06 16:36 +0000 [r73727] Mark Michelson + + * apps/app_voicemail.c: Fixing a rare case which causes voicemail + to crash when compiled with IMAP storage. inboxcount has the + possibility of finding an "interactive" vm_state when no + persistent "non-interactive" vm_state exists for that mailbox. If + this should happen when someone attempts to leave a message, it + results in a crash. This patch, along with my commit in revision + 72670 fix issue 10053, reported by jaroth. closes issue #10053 + +2007-07-06 16:12 +0000 [r73679-73696] Russell Bryant + + * res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06 + Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras + Patches submitted by: Corydon76 Tested by: apsaras Fix a problem + with MSSQL 2005 by explicitly stating that '\' is being used as + an escape character. ........ + + * /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | + 7 lines (closes issue #10125) Reported by: makoto Patches + submitted by: makoto This fixes a crash in chan_sip that happens + when the bindaddr setting is not valid on Asterisk startup, gets + fixed, and then a reload gets issued. ........ + +2007-07-06 15:27 +0000 [r73675] Mark Michelson + + * /, channels/chan_agent.c: Merged revisions 73674 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06 + Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy. + (issue 9618, reported by jiddings, patched by moi) closes issue + #9618 ........ + +2007-07-06 03:34 +0000 [r73551-73629] Russell Bryant + + * BUGS: fix a little spelling error + + * channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop + the monitor thread if it was never started. (closes issue #10124, + reported by gzero, fixed by me) + + * channels/chan_iax2.c: copy from the correct buffer when deferring + a full frame (related to issue #9937) + + * channels/chan_iax2.c: * Store the call number that a thread is + processing without the full frame bit set to ease debugging * + When deferring a full frame for processing, stick it into the + queue for the thread that is processing frames for that call, not + the one that read the current frame and is about to go back into + the idle list (related to issue #9937) + +2007-07-05 22:20 +0000 [r73548] Kevin P. Fleming + + * /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) + | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just + like we don't support it for G.729 ........ + +2007-07-05 20:50 +0000 [r73512] Russell Bryant + + * res/res_features.c: Pass HOLD and UNHOLD frames to the other + channel when they are returned from a native bridge function. + This fixes a problem where when two zap channels are natively + bridged and one does a flash hook, the other channel did not + receive music on hold. (Reported to me directly by Doug Bailey at + Digium) + +2007-07-05 19:18 +0000 [r73467] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 + lines Copy language information to the dialog structure when + calling a peer for situations where a PBX may be started on the + dialed channel. (issue #10121 reported by clegall_proformatique) + ........ + +2007-07-05 15:59 +0000 [r73400] Mark Michelson + + * apps/app_queue.c: Correcting a minor CLI bug I found. When + issuing the queue show command, if you type queue show and then + press tab, you can continue pressing tab and it will keep + auto-completing queue names even though only 1 queue can be used + as an argument. + +2007-07-05 15:28 +0000 [r73398] Russell Bryant + + * channels/chan_vpb.cc, channels/Makefile: Make this module build + for me in dev-mode + +2007-07-05 14:21 +0000 [r73316-73355] Joshua Colp + + * apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2 + lines Tweak spy locking. (issue #9951 reported by welles) + ........ + + * channels/chan_local.c, /: Merged revisions 73318 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul + 2007) | 2 lines Actually check to make sure a PBX was started on + one of the Local channels instead of blindly assuming it was. + (issue #10112 reported by makoto) ........ + + * /, apps/app_queue.c: Merged revisions 73315 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2 + lines Reset ServicelevelPerf variable back to 0 if we are unable + to calculate it each time... otherwise we will get previous + values. (issue #10117 reported by noriyuki) ........ + +2007-07-04 14:53 +0000 [r73208-73253] Christian Richter + + * channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04 + Jul 2007) | 1 line bchannel configurations like echocancel and + volume control, need to be setuped on inbound calls too. ........ + + * channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04 + Jul 2007) | 1 line bad bug in overlapdial case, we called + start_pbx multiple times, because the state wasn't changed.. + ........ + +2007-07-03 20:17 +0000 [r73143] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2.c, main/Makefile, + main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing + expr floating patch from 1.4; too much of a behavior change. If + you want this fix, try trunk instead. bug 9508. + +2007-07-03 15:42 +0000 [r73104-73106] Jason Parker + + * /: What the heck. This should not have happened. + + * /: use autotagged externals + +2007-07-03 12:38 +0000 [r73053] Tilghman Lesher + + * apps/app_dial.c, /: Merged revisions 73052 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007) + | 2 lines RetryDial should accept a 0 argument, but it does not, + because atoi does not distinguish between 0 and error (closes + issue #10106) ........ + +2007-07-03 08:17 +0000 [r73005] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03 + Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only + be called from mISDN Source channels.. #9449 ........ + +2007-07-02 20:16 +0000 [r72933] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput, + main/Makefile, main/ast_expr2.h, main/ast_expr2.y, + main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support + for floating point numbers added to ast_expr2 $\[...\] exprs. + Fixes bug 9508, where the expr code fails with fp numbers. The + MATH function returns fp numbers by default, so this fix is + considered necessary. + +2007-07-02 18:18 +0000 [r72926] Russell Bryant + + * main/manager.c: Remove a bogus comment and add proper locking to + the handler function for the CLI command to show information on + manager actions. + +2007-07-02 17:59 +0000 [r72925] Jason Parker + + * /: Blocked revisions 72924 via svnmerge ........ r72924 | qwell | + 2007-07-02 12:58:25 -0500 (Mon, 02 Jul 2007) | 4 lines Fix an + issue with playing "oclock" multiple times in French with 24 hour + time format. Issue 10101 ........ + +2007-07-02 14:32 +0000 [r72888] Joshua Colp + + * main/channel.c: Added additional DTMF debug messages for when + emulation occurs. + +2007-07-02 08:41 +0000 [r72850-72852] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 72585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) | + 1 line check if the bchannel stack id is already used, if so + don't use it a second time. Also added a release_chan lock, so + that the same chan_list object cannot be freed twice. chan_misdn + does not crash anymore on heavy load with these changes. ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 72099 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) | + 1 line simplified generation for dummy bchannels, also we mark + them as dummies, so they are not used later as real-bchannels, + optimized the RESTART mechanisms, we block a channel now on + cause:44, and send out a RESTART automatically, then on reception + of RESTART_ACKNOWLEDGE we unblock the channel again. ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged + revisions 72087 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) | + 1 line simplified channel finding and locking a lot. removed + unnecessary #ifdefed areas. ........ + +2007-07-01 23:52 +0000 [r72806] Russell Bryant + + * pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) | + 5 lines When appending lines to call files to keep track of + retries, write a leading newline just in case the original call + file did not have a newline at the end. This fix is in response + to a problem I saw reported on the asterisk-users mailing list. + ........ + +2007-06-30 16:50 +0000 [r72705-72766] Russell Bryant + + * configure, configure.ac: Tweak the configure script so that error + output isn't spewed to the console when searching for GTK2 libs, + and they aren't found. + + * formats/format_pcm.c: give format_pcm a more concise destription + +2007-06-29 19:07 +0000 [r72665] Luigi Rizzo + + * main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for + absence of the function. This was already done in trunk. + +2007-06-29 Russell Bryant + + * Asterisk 1.4.6 released. + +2007-06-29 16:31 +0000 [r72630] Russell Bryant + + * /: Blocked revisions 72629 via svnmerge ........ r72629 | russell + | 2007-06-29 11:30:56 -0500 (Fri, 29 Jun 2007) | 4 lines Backport + changes that make chan_iax2 not start the PBX on an incoming + channel until the three-way call setup is completed. These + changes are already in 1.4 and trunk. ........ + +2007-06-29 14:26 +0000 [r72597-72599] Joshua Colp + + * main/cdr.c: Minor change for older GCC versions. + + * Makefile, configure, configure.ac, makeopts.in: Backport fix for + GCC versions without support for declaration-after-statement. + +2007-06-29 04:47 +0000 [r72554-72556] Tilghman Lesher + + * main/manager.c: Issue 10055 - Change memory allocation to use the + heap for a command, since the output has the potential to + overflow the stack (as it did here) + + * res/res_jabber.c: Fix 1.4 breakage + +2007-06-28 19:44 +0000 [r72493] Russell Bryant + + * configure, include/asterisk/autoconfig.h.in: regenerate the + configure script for rizzo + +2007-06-28 19:29 +0000 [r72453-72489] Luigi Rizzo + + * configure.ac: add a check for gethostbyname_r so we can simplify + the handling e.g. in utils.c Also add comments on a couple of + features which are not working on FreeBSD. All the above has been + already done in trunk so the merge must be blocked. Can someone + please regenerate ./configure ? + + * Makefile, channels/chan_zap.c, main/say.c: Add + -Wdeclaration-after-statement to AST_DEVMODE flags to catch + variable declarations in the middle of a block. Fix the few + instances of the above spotted out by the compiler. All of this + has been already done or is not applicable in trunk, so the merge + of this change will be blocked. + + * apps/app_meetme.c: cast a time_t so that it does not conflict + with the print format. This change was already done on trunk so + this change needs to be blocked from merging. + +2007-06-27 23:29 +0000 [r72383] Brett Bryant + + * main/asterisk.c, /: Merged revisions 72373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) | + 3 lines Reinstating patch. This actually fixes the problem, + however I was running a development branch without it and + mistakenly thought it wasn't fixed. Fixes issue #10010, and + #9654: 100% CPU usage caused by an asterisk console losing it's + controlling terminal. ........ + +2007-06-27 23:25 +0000 [r72381] Joshua Colp + + * apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun + 2007) | 2 lines Update documentation to clarify variable usage + with MixMonitor. (issue #9494 reported by netoguy) ........ + +2007-06-27 23:03 +0000 [r72335] Brett Bryant + + * main/asterisk.c, /: Merged revisions 72333 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) | + 2 lines Reverted changes for earlier revisions 72259 to 72261. + Issue #9654, #10010 ........ + +2007-06-27 22:58 +0000 [r72328-72331] Joshua Colp + + * channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to + our list. Since these are dynamic payloads the other side + shouldn't care. (issue #9426 reported by irroot) + + * /, apps/app_queue.c: Merged revisions 72327 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2 + lines Fix issue where queue log events might be missing. (issue + #7765 reported by mtryfoss) ........ + +2007-06-27 21:08 +0000 [r72272] Russell Bryant + + * /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) | + 5 lines Fix a minor issue with parsing the priority number. You + could have as much whitespace as you want around a numeric + priority, but you couldn't have any whitespace around a special + priority like "n" or "hint". (issue #10039, reported by mitheloc, + fixed by me) ........ + +2007-06-27 20:46 +0000 [r72260] Brett Bryant + + * main/asterisk.c, /: Merged revisions 72259 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) | + 4 lines Fixes 100% load when controlling terminal disappears. + Issue #9654, #10010 ........ + +2007-06-27 20:25 +0000 [r72257] Joshua Colp + + * main/channel.c, /: Merged revisions 72256 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 + lines I may possibly get shot for doing this... but... defer CDR + processing until after the channel has been dealt with. This + should eliminate all of the issues with channels going funky + (SIP/PRI) when you are posting CDRs to a database that is either + slow or unavailable and do not want to enable batching. ........ + +2007-06-27 19:13 +0000 [r72205] Kevin P. Fleming + + * channels/chan_zap.c: use the proper type for storing group number + bits so that if someone specifies 'group=42' it will actually + work instead of being silently ignored + +2007-06-27 18:40 +0000 [r72182-72185] Jason Parker + + * /: Blocked revisions 72184 via svnmerge ........ r72184 | qwell | + 2007-06-27 13:40:15 -0500 (Wed, 27 Jun 2007) | 4 lines Fix + another problem in voicemail with missing symbols. Issue 10074, + patch by kryptolus, extended to include #if 0'd blocks (just in + case) ........ + + * apps/app_voicemail.c: Fix another problem in voicemail with + missing symbols. Issue 10074, patch by kryptolus, extended to + include #if 0'd blocks (just in case) + +2007-06-27 17:31 +0000 [r72148] Joshua Colp + + * main/channel.c: Make the ast_read_noaudio API call behave better + under circumstances where DTMF emulation was happening and a + generator was setup. (issue #10065 reported by stevefeinstein) + +2007-06-27 17:10 +0000 [r72125] Jason Parker + + * channels/chan_gtalk.c: Don't modify a variable that we don't want + modified. Make a copy of it instead. Issue 10029, patch by + phsultan with slight modifications by me (to remove needless + casts). + +2007-06-27 16:34 +0000 [r72112] Russell Bryant + + * main/rtp.c: Only output debug information related to RTCP + timestamps when RTCP debug is turned on (issue #10066, patch by + me) + +2007-06-27 07:58 +0000 [r72042] Christian Richter + + * channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) | + 1 line for inbound TE calls, we setup the bchannel when we get + the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready. + removed some #if 0 areas which weren't used anymore. ........ + r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) | + 1 line isdn_lib.c didn't compile ........ + +2007-06-27 00:58 +0000 [r72006] Joshua Colp + + * pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work. + +2007-06-26 23:02 +0000 [r71953] Mark Michelson + + * apps/app_voicemail.c: Removing a pointless line. This variable + was already set earlier and between then and this line, there is + no way that the values on the right side of the assignment could + have changed. + +2007-06-26 20:36 +0000 [r71915] Jason Parker + + * main/rtp.c: Don't dereference a pointer that may be NULL here. + Issue 10017. + +2007-06-26 19:00 +0000 [r71877] Mark Michelson + + * apps/app_voicemail.c: A few changes, the ultimate goal of which + is to keep better track of the number of messages that a mailbox + currently has. A description of the changes: 1. Changed the + "updated" field of the vm_state struct to act more as a binary + semaphore than a counting semaphore, since its current + implementation made the inboxcount function not work properly. + This change falls in line with a change made by UPenn with their + IMAP setup and helps to sync our changes with theirs. 2. + Eliminated some redundant calls to get_vm_state_by_mailbox inside + leave_voicemail 3. Use the play_folder variable to keep track of + the number of old and new messages in a mailbox as the messages + are deleted 4. Added an increment to the number of new messages + that was not there previously in the leave_voicemail function + +2007-06-26 17:49 +0000 [r71848] Jason Parker + + * /: Blocked revisions 71847 via svnmerge ........ r71847 | qwell | + 2007-06-26 12:49:14 -0500 (Tue, 26 Jun 2007) | 4 lines Don't try + to install an init script that doesn't exist. Reported to me on + #asterisk on Freenode IRC. ........ + +2007-06-26 15:47 +0000 [r71796] Mark Michelson + + * apps/app_voicemail.c: Fixing bug where the authuser was + mistakenly pulled from the mailbox string instead of the IMAP + user. (closes issue 10054, reported and patched by jaroth) + +2007-06-26 12:27 +0000 [r71657-71751] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007) + | 2 lines Issue 10062 - Trying to move a message without + selecting one first results in memory corruption ........ + + * /, res/res_agi.c: Merged revisions 71656 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007) + | 2 lines Issue 10035 - handle_exec returns a result inconsistent + with all of the other AGI commands ........ + +2007-06-25 14:13 +0000 [r71522-71576] Joshua Colp + + * channels/chan_h323.c: Build a peer as well when hash323 is + enabled in users.conf (issue #9599 reported by asagage) + + * channels/chan_agent.c: Minor tweak for queueing up the unhold + frame... this will teach me to do bugs while half asleep. (issue + #10046 reported by dimas) + +2007-06-25 12:40 +0000 [r71519] Russell Bryant + + * doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue + #10048, Matti) + +2007-06-25 01:10 +0000 [r71412-71430] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 + lines Ignore other URIs after the first in a 300 Multiple Choice + response. (issue #10041 reported by homesick) ........ + + * main/cdr.c: Fix it so 1.4 actually compiles on my box. + + * channels/chan_agent.c: Check to make sure the channel pointer is + present before queueing up an unhold frame on it. (issue #10046 + reported by dimas) + +2007-06-24 20:16 +0000 [r71362-71371] Russell Bryant + + * build_tools/prep_tarball: Include the menuselect-tree file in + tarballs to make builds from tarballs a little bit faster + + * main/asterisk.c, /: Merged revisions 71358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) | + 2 lines Revert the patch from issue 9654 due to an unexpected + side effect ........ + +2007-06-24 17:50 +0000 [r71289-71291] Tilghman Lesher + + * res/res_features.c: Issue 10044 - chan->cdr is NULL here, so + peer->cdr is what we really wanted to use + + * main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24 + Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to + be able to set variables to the empty string. ........ + +2007-06-23 03:29 +0000 [r71230] Steve Murphy + + * main/cdr.c, res/res_features.c: This patch is meant to fix 8433; + where clid and src are lost via bridging. + +2007-06-22 22:44 +0000 [r71214] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20 + Jun 2007) | 1 line fixed a bug that was introduced by copy and + paste in the last commit ..bchannels weren't cleaned properly. + ........ + +2007-06-22 16:05 +0000 [r71128] Joshua Colp + + * /: Blocked revisions 71124 via svnmerge ........ r71124 | file | + 2007-06-22 12:02:40 -0400 (Fri, 22 Jun 2007) | 2 lines Send an + unhold indication when going off hold. (issue #10036 reported by + speedy) ........ + +2007-06-22 15:38 +0000 [r71096-71123] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 70672 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) | + 1 line we activate the bchannels in TE mode on incoming calls + only when we want to connect the call. ........ + + * channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20 + Jun 2007) | 1 line forgot one place .. ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20 + Jun 2007) | 1 line on receiption of cause:44 we mark the channel + as in use and inform the user about the situation, we need to + test the RESTART stuff then. Also shuffled the + empty_chan_in_stack function after the bchannel cleaning + functions, to avoid race conditions. ........ + + * channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19 + Jun 2007) | 1 line when we send out a SETUP, but get no response, + we should cleanup everything after reception of a hangup. + ........ + + * /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) | + 1 line restart indicator 0x80 is correct, at least that's what + libpri does. ........ + + * channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12 + Jun 2007) | 1 line if the bridged partner is mISDN too we should + not send dtmf tones, they are transmitted inband always ........ + + * channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12 + Jun 2007) | 1 line if we have already some digits, we just stop + the tones. ........ + +2007-06-22 15:00 +0000 [r71068] Jason Parker + + * apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged + revisions 71065 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 + lines Fix a few silly usages of ast_playstream() - it only ever + returns 0... Issue 10035 ........ + +2007-06-22 14:53 +0000 [r71066] Brett Bryant + + * main/asterisk.c, /: Merged revisions 71064 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) | + 10 lines Fixed infinite loop when controlling terminal was lost + and return value of input function wasn't checked for errors. + This would cause 100% cpu to be taken up. (closes issue #9654, + issue #10010) Reported by: mnicholson, and eserra Idea for the + patch from mnicholson, patched by me ........ + +2007-06-22 14:10 +0000 [r71063] Steve Murphy + + * main/cdr.c: My conditions for merging amaflags info was naive; + DOCUMENTATION is the default, although null is possible; theft of + user-settable fields is not good. Just copy them, leave them + alone. + +2007-06-22 03:14 +0000 [r71003] Russell Bryant + + * channels/chan_iax2.c: Fix a small typo which ... well ... + completely broke chan_iax2. oops! (issue #9937, patch by me) + +2007-06-21 22:34 +0000 [r70949] Steve Murphy + + * main/cdr.c, /: Merged revisions 70948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1 + line This little fix is in response to bug 10016, but may not + cure it. The code is wrong, clearly. In a situation where you set + the CDR's amaflags, and then ForkCDR, and then set the new CDR's + amaflags to some other value, you will see that all CDRs have had + their amaflags changed. This is not good. So I fixed it. ........ + +2007-06-21 21:40 +0000 [r70899] Joshua Colp + + * apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2 + lines Don't explode if the gain option is specified without a + value. (issue #9274 reported by mfarver) ........ + +2007-06-21 21:14 +0000 [r70866-70883] Russell Bryant + + * channels/chan_iax2.c: Put the thread reading from the socket back + in the idle list if it deferred the processing of a full frame to + another thread + + * channels/chan_iax2.c: If a full frame is received while one of + the iax2 threads is in the middle of handling a full frame for + the same call, queue it up for processing by that same thread + later instead of dropping it. (issue #9937, patch by me) + +2007-06-21 20:19 +0000 [r70841] Steve Murphy + + * cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1 + line it was pointed out that the cdr_custom config load could get + a lock, and under certain circumstances, would never release it. + I also noted that the situation where more than one mapping spec + was warned about, but did not ignore further mappings as it had + promised. I think I have fixed both situations. ........ + +2007-06-21 19:49 +0000 [r70808] Mark Michelson + + * apps/app_voicemail.c: When volgain is used don't leave a + temporary file behind. (Closes Issue 8514, Reported and patched + by ulogic, code reviewed by Jason Parker) + +2007-06-21 15:22 +0000 [r70727] Joshua Colp + + * main/rtp.c: Do not Packet2Packet bridge if packetization settings + do not allow it. (issue #9117 reported by phsultan) + +2007-06-21 15:21 +0000 [r70726] Russell Bryant + + * apps/app_meetme.c: Remove a couple of duplicate unlocks + +2007-06-21 13:58 +0000 [r70677] Joshua Colp + + * apps/app_voicemail.c: Fix building with ODBC storage enabled. + (issue #10025 reported by denisgalvao) + +2007-06-21 13:00 +0000 [r70656] Steve Murphy + + * main/cdr.c: Via complaints aired in asterisk-users, I submit + these changes, which allow cdr updates to see macro + context/exten, whether hung up or not + +2007-06-20 23:32 +0000 [r70554-70612] Jason Parker + + * cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql. + Issue 10020, patch by my, with credit to prashant_jois for + pointing out the problem. + + * cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race + condition fix + + * cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur + when reloading the module. Issue 10022, patch by me, with credit + to prashant_jois for finding the bug. + +2007-06-20 22:22 +0000 [r70552] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 + lines Don't overwrite the configured username setting upon a + REGISTER. (issue #8565 reported by jsmith) ........ + +2007-06-20 20:53 +0000 [r70494] Jason Parker + + * channels/chan_skinny.c: Make sure we clear the previously dialed + number if it did not exist. Issue 9958. + +2007-06-20 19:29 +0000 [r70445] Tilghman Lesher + + * apps/app_dial.c, /: Merged revisions 70444 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007) + | 2 lines Issue 9997 - Timelimit times out the wrong channel + ........ + +2007-06-20 18:46 +0000 [r70397] Russell Bryant + + * channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) | + 5 lines Fix a problem where an established call would not be + properly disconnected when a PRI disconnect is received depending + on which cause code was received. (issue #9588, original patch by + softins, updated patch from jtexter3, and some additional + feedback from mhardeman) ........ + +2007-06-20 17:52 +0000 [r70198-70360] Joshua Colp + + * main/rtp.c, main/frame.c: Put the speex packetization values back + in but disable it when setting up the smoother. + + * main/frame.c: Don't do packetization/smoother stuff with speex, + it doesn't work. + +2007-06-20 00:03 +0000 [r70084-70164] Russell Bryant + + * contrib/scripts/ast_grab_core: don't delete the backtrace in + ast_grab_core + + * channels/chan_gtalk.c: Only attempt to queue a hangup on the + owner channel if it actually exists. (issue #9795, patch from + zandbelt) + +2007-06-19 18:23 +0000 [r70062] Steve Murphy + + * main/channel.c, /: Merged revisions 70053 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 + line This fixes 9246, where channel variables are not available + in the 'h' exten, on a 'ZOMBIE' channel. The fix is to + consolidate the channel variables during a masquerade, and then + copy the merged variables back onto the clone, so the zombie has + the same vars that the 'original' has. ........ + +2007-06-19 17:07 +0000 [r70003] Joshua Colp + + * main/rtp.c, /: Merged revisions 69992 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 + lines Handle the CC field in the RTP header. (issue #9384 + reported by DoodleHu) ........ + +2007-06-19 16:46 +0000 [r69991] Russell Bryant + + * /: Blocked revisions 69990 via svnmerge ........ r69990 | russell + | 2007-06-19 11:45:37 -0500 (Tue, 19 Jun 2007) | 12 lines + Backport fix for crashes related to subscriptions from 1.4 ... + Fix a crash that could occur when handing device state changes. + When the state of a device changes, the device state thread tells + the extension state handling code that it changed. Then, the + extension state code calls the callback in chan_sip so that it + can update subscriptions to that extension. A pointer to a + sip_pvt structure is passed to this function as the call which + needs a NOTIFY sent. However, there was no locking done to ensure + that the pvt struct didn't disappear during this process. (issue + #9946, reported by tdonahue, patch by me, patch updated to trunk + to use the sip_pvt lock wrappers by eliel) ........ + +2007-06-19 16:24 +0000 [r69987] Joshua Colp + + * main/channel.c, /: Merged revisions 69986 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2 + lines Update BRIDGEPEER variable if set to the new channel name + when a masquerade happens. (issue #9699 reported by dimas) + ........ + +2007-06-19 15:22 +0000 [r69944] Russell Bryant + + * channels/chan_sip.c: Fix a crash that could occur when handing + device state changes. When the state of a device changes, the + device state thread tells the extension state handling code that + it changed. Then, the extension state code calls the callback in + chan_sip so that it can update subscriptions to that extension. A + pointer to a sip_pvt structure is passed to this function as the + call which needs a NOTIFY sent. However, there was no locking + done to ensure that the pvt struct didn't disappear during this + process. (issue #9946, reported by tdonahue, patch by me, patch + updated to trunk to use the sip_pvt lock wrappers by eliel) + +2007-06-19 13:55 +0000 [r69805-69895] Joshua Colp + + * /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2 + lines Perform an extra hangup check just in case. (issue #9589 + reported by bcnit) ........ + + * /, res/res_features.c: Merged revisions 69846 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2 + lines Add parked call extension AFTER the parking slot has been + announced, otherwise two threads will try to handle the same + channel and it will go kaboom. (issue #9191 reported by japple) + ........ + + * main/callerid.c: Fix for building on PowerPC under Linux. + +2007-06-18 19:48 +0000 [r69796] Tilghman Lesher + + * channels/chan_sip.c: Issue 10005 - Segfault with missing + arguments, plus fix a missing define for SIP INFO channels + +2007-06-18 19:00 +0000 [r69775-69794] Joshua Colp + + * channels/chan_sip.c: Don't count RTP timeout when involved in a + T38 fax session. (issue #9222 reported by ivoc) + + * /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 + lines Set the peer name on the dialog to the one configured in + sip.conf and NOT the username to be used for authentication + attempts. (issue #9967 reported by achauvin) ........ + +2007-06-18 17:46 +0000 [r69744] Tilghman Lesher + + * contrib/scripts/safe_asterisk, /: Merged revisions 69743 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007) + | 2 lines Issue 9998 - Remove SIG prefix, since it's not + supported by ksh ........ + +2007-06-18 16:51 +0000 [r69708] Joshua Colp + + * main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called + for the first time so that it does not needlessly spit out + changed messages when the host really didn't change. + +2007-06-18 16:35 +0000 [r69689-69702] Russell Bryant + + * res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c, + build_tools/menuselect-deps.in, configure, funcs/func_odbc.c, + include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c: + To prevent 92138749238754 more reports of "I have unixodbc + installed, but still can't build *_odbc.so!", check for ltdl + directly, instead of just listing it as another library to + include in the unixodbc check in the configure script. This also + makes ltdl show up as a dependency in menuselect so people know + what to go install. (related to issue #9989, patch by me) + + * build_tools/prep_moduledeps: Change the use of "echo -e" to + "printf". On systems where /bin/sh is not bash, most of the lines + in menuselect-tree were getting a "-e" at the beginning of every + line. I'm surprised nobody noticed this, but I think the XML + parser was being very nice and ignoring them. + +2007-06-18 16:04 +0000 [r69661-69668] Joshua Colp + + * channels/chan_sip.c: Don't defer the BYE till later on a transfer + when the transfer itself goes kaboom and has no hope of working. + + * channels/chan_sip.c: Few minor transfer tweaks. We can't unlock + something we never locked, and better handle a specific scenario + with doing an attended transfer between two non-bridged calls. + +2007-06-18 15:46 +0000 [r69660] Russell Bryant + + * Makefile: Tweak paths for BSD systems (issue #10001, stuarth) + +2007-06-18 13:55 +0000 [r69625] Joshua Colp + + * channels/chan_sip.c: Fix issue where it would be possible for the + negotiated codecs to get set back to nothing. (issue #9992 + reported by yehavi) + +2007-06-15 Russell Bryant + + * Asterisk 1.4.5 released. + +2007-06-15 20:18 +0000 [r69579] Russell Bryant + + * res/res_features.c: Fix a silly deadlock in res_features that I + found while debugging on one of blitzrage's test machines. It was + one of the situations where he was seeing hung channels, and may + be the cause of some of the reports from other people. (related + to issue #9235) + +2007-06-15 19:23 +0000 [r69558] Joshua Colp + + * apps/app_speech_utils.c: Add support for setting the maximum + length of acceptable DTMF in SpeechBackground. + +2007-06-15 15:27 +0000 [r69518] Russell Bryant + + * apps/app_meetme.c: The SLATRUNK_STATUS variable indicated + "SUCCESS" for both an answer of the incoming call on the trunk, + or if the trunk reached its ring timeout. This patch changes the + variable to say "RINGTIMEOUT" in that case. (issue #9973, + reported by n00dle, patch by me) + +2007-06-14 23:22 +0000 [r69434-69470] Jason Parker + + * main/config.c, /: Merged revisions 69469 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4 + lines Fix an issue where the line number in an unterminated + comment block error message would show the wrong line number. + "Reported" to me on #asterisk (somebody posted an error message, + and I happened to catch it) ........ + + * sounds/Makefile: Update to latest versions of sound files. + +2007-06-14 21:50 +0000 [r69392] Kevin P. Fleming + + * cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c, + cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c, + main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c, + apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c, + main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c, + channels/chan_iax2.c: use ast_localtime() in every place + localtime_r() was being used + +2007-06-14 21:08 +0000 [r69358] Russell Bryant + + * main/say.c: Fix some problems with saying dates and times for the + "tw" langauge (issue #9964, ljmid) + +2007-06-14 15:21 +0000 [r69259] Jason Parker + + * funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun + 2007) | 4 lines Change a quite broken while loop to a for loop, + so "continue;" works as expected instead of eating 99% CPU... + Issue 9966, patch by me. ........ + +2007-06-13 21:19 +0000 [r69184-69222] Joshua Colp + + * channels/chan_iax2.c: Whoops... + + * channels/chan_iax2.c: Let's make chan_iax2 media only native + transfers actually work. (issue #9376 reported by simone + cittadini) + + * channels/iax2-parser.c: Add TXMEDIA to list so that it is + properly displayed during iax2 packet output. + +2007-06-13 19:57 +0000 [r69183] Russell Bryant + + * channels/chan_sip.c: Move the logic for destroying a call when no + response is received to a BYE outside of the block that checks + for FLAG_FATAL to be set. This flag is only set when the packet + is transmitted with the reliability set to XMIT_CRITICAL when the + original packet is transmitted. A BYE is always sent with it set + to XMIT_RELIABLE, meaning this code could never be encountered. + This resulted in seeing some SIP channels that would never go + away with the last packet sent being a BYE. (part of issue #9235, + patch from jcmoore) + +2007-06-13 19:41 +0000 [r69181] Mark Michelson + + * apps/app_voicemail.c: Contains a patch for fixing an encoding + problem when using Outlook to view voicemail emails and + attachments. This fix has also been tested on Thunderbird, + Evolution, Pine, and Mutt. (Issue 9336, reported by marwick, + patched by mutterc) + +2007-06-13 19:08 +0000 [r69128-69144] Joshua Colp + + * apps/app_meetme.c: Really ignore NULL frames and check whether + the channel hungup or not. (issue #9912 reported by junky) + + * /, main/app.c: Merged revisions 69127 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2 + lines Return group counting to previous behavior where you could + only have one group per category. (issue #9711 reported by + irroot) ........ + +2007-06-13 16:56 +0000 [r69016-69071] Russell Bryant + + * channels/chan_sip.c: Clarify a bit of logic. This doesn't change + behavior in any way, but it is helpful when following the logic + to debug problems like 9235. + + * channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct + was accessed without the lock held. This issue was reported to me + via email by Dmitry Mishchenko. Thanks! + + * cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois + in #asterisk-bugs. PQclear() was not called on the result + structure after doing a PQexec(). Also, fix up some formatting in + passing. + +2007-06-12 19:36 +0000 [r69012-69014] Joshua Colp + + * channels/chan_iax2.c: Change the full frame dropping log message + to debug to avoid future bug reports. + + * channels/chan_iax2.c: Schedule the sending of a PING packet a + second later than previously so that it does not collide with the + LAGRQ. + +2007-06-12 19:13 +0000 [r69010] Russell Bryant + + * main/channel.c: In ast_channel_make_compatible(), just return if + the channels' read and write formats already match up. There are + code paths that call this function on a pair of channels multiple + times. This made calls fail that were using g729 in some cases. + The reason is that codec_g729a will unregister itself from the + list of available translators will all licenses are in use. So, + the first time the function got called, the right translation + path was allocated. However, the second time it got called, the + code would not find a translation path to/from g729 and make the + call fail, even if the channel actually already had a g729 + translation path allocated. (SPD-32) + +2007-06-12 14:23 +0000 [r68922] Joshua Colp + + * main/rtp.c, /: Merged revisions 68921 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 + lines Bring RTP back to Asterisk at the end of a native bridge no + matter what. ........ + +2007-06-11 21:20 +0000 [r68814] Jason Parker + + * include/asterisk/time.h: Solaris 10 sometimes (?) needs this + include in order to have NULL defined. + +2007-06-11 20:45 +0000 [r68781] Tilghman Lesher + + * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly + used + +2007-06-11 16:57 +0000 [r68733] Christian Richter + + * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 68732 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) | + 1 line added check for NULL Pointer when calling misdn_new. + Asterisk does not allow us to create channels anymore when stop + gracefully is used :). also modified the restart_indicator to 0 + ........ + +2007-06-11 14:33 +0000 [r68683] Joshua Colp + + * main/channel.c, /: Merged revisions 68682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 + lines Improve deadlock handling of the channel list. (issue #8376 + reported by one47) ........ + +2007-06-11 10:29 +0000 [r68644] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /, channels/misdn/ie.c, + channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) | + 1 line fixed problem that the dummybc chanels had no lock, + checking for the lock now. Also fixed the channel restart stuff, + we can now specify and restart particular channels too. ........ + +2007-06-11 04:21 +0000 [r68595] Tilghman Lesher + + * pbx/pbx_config.c: "dialplan save" produced garbage in the config + file + +2007-06-08 22:23 +0000 [r68527] Russell Bryant + + * /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) | + 4 lines Don't automatically hang up after running Dictate so that + callers can exit cleanly using '#' (closes issue #9577, patch + from Thomas Andrews) ........ + +2007-06-08 15:52 +0000 [r68450] Kevin P. Fleming + + * channels/chan_iax2.c: actually remember the type/subclass of full + frames that are in process + +2007-06-08 00:17 +0000 [r68370-68401] Joshua Colp + + * /, main/say.c: Merged revisions 68397 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2 + lines Don't call ast_waitstream_full when the control file + descriptor and audio file descriptor are not set, simply call + ast_waitstream! (issue #8530 reported by rickead2000) ........ + + * main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2 + lines Do a DNS lookup immediately upon calling the dnsmgr + function, don't wait until a refresh happens. (issue #9097 + reported by plack) ........ + +2007-06-07 23:14 +0000 [r68354] Russell Bryant + + * /, main/say.c: Merged revisions 68351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) | + 3 lines Fix a problem where saying a character wouldn't properly + break out when the caller pressed '#' (issue #8113, reported by + patbaker82, patch from jamesgolovich (hey, long time no see!) and + patbaker82) ........ + +2007-06-07 23:00 +0000 [r68326] Jason Parker + + * apps/app_voicemail.c: Fix incorrect French syntax of "old + messages". Request for feedback was sent to asterisk-dev mailing + list, with little response. Issue 9118, patch by junky. + +2007-06-07 22:14 +0000 [r68313] Kevin P. Fleming + + * channels/chan_iax2.c: some improvements to the IAX2 full frame + dropping logic recently added: - use inaddrcmp(), since we have + it - output the type of frame and subclass being dropped, and the + type/subclass that is already being processed (which caused the + drop) + +2007-06-07 21:16 +0000 [r68280] Russell Bryant + + * channels/chan_agent.c, apps/app_queue.c: Fix loading persistent + queue members when using realtime configuration for queues. Also, + remove an unneeded leading slash for the astdb family. (issue + #9911, patch by atis) + +2007-06-07 20:25 +0000 [r68211-68249] Jason Parker + + * channels/chan_skinny.c: Fix an issue with newer phones which + require packets be padded out to the correct length. Issue 9887, + patch by DEA. + + * apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4 + lines Don't try to save voicemail greetings unless the user + presses '1' to accept/save. Issue 9904, patch by me. ........ + +2007-06-07 19:47 +0000 [r68198] Mark Michelson + + * apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a + check to make sure that greetings get stored properly. (Issue + 8016, reported by edhorton, patched by alamantia with + modification by me. Thanks to Jason Parker for the advice on + this). + +2007-06-07 19:46 +0000 [r68196] Olle Johansson + + * channels/chan_features.c: Disable chan_features by default in + menuselect + +2007-06-07 19:30 +0000 [r68192] Russell Bryant + + * main/strcompat.c: Include stdarg.h for build issues on Solaris + (issue #9381) + +2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp + + * main/channel.c: Fix logic when doing a name based channel search + for a structure when you want to start from a specific point in + the channel list. (issue #9324 reported by slavon) + + * apps/app_dial.c, /: Merged revisions 68070 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 + lines Allow the 'g' option to work if used with the 'S' option. + (issue #9888 reported by gasparz) ........ + +2007-06-07 10:00 +0000 [r67993-68030] Olle Johansson + + * res/res_jabber.c: Adding a few Todo's to res_jabber so we don't + forget. + + * res/res_jabber.c: Ok, we found out that this is not about if you + have any *active* clients using TLS, but if you have initialized + TLS at all during the lifetime of the module. So if you reload to + disable TLS, it won't help. + + * res/res_jabber.c: If you have a jabber client that uses TLS, + refuse unload. Bad fix, but will prevent crashes while we are + trying to find a workaround. Iksemel development seems to have + stalled and we might have to stop using the TCP/TLS connections + in that library and use our own, which would scale better from a + poll/select perspective I guess. It would also make it easier to + migrate to OpenSSL and stop Asterisk from depending on both + OpenSSL and GnuTLS. + + * include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make + sure we can unload res_jabber. Patch by phsultan - thanks! Due to + a bug in the iksemel library, this will not work if you are using + GTLS in the connection. That's being investigated. If you figure + out a way to handle that without us having to patch iksemel, let + us know in the bug report. Thanks. + +2007-06-07 00:10 +0000 [r67924-67941] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 + lines Only notify the devicestate system of a peer state change + when the peer is built from the config file. (issue #9900 + reported by arkadia) ........ + + * main/file.c: Properly handle cases where a stream can't be + written to. (issue #9757 reported by junky) + +2007-06-06 22:08 +0000 [r67862-67872] Russell Bryant + + * res/res_snmp.c: Disable reload functionality in res_snmp. It is + not possible to initialize the snmp library more than once + without completely unloading the module and loading it again. + (issue #9571, reported by hristo, additional helpful debug + information from festr, patch from me) + + * channels/chan_sip.c: Fix a crash when doing call pickups with SIP + phones. The code unlocked the channel when it should not have. + (issue #9652, reported by corruptor, fixed by me) + +2007-06-06 19:26 +0000 [r67804] Mark Michelson + + * apps/app_voicemail.c: Fix for Issue 9810. There was a segfault + under a specific set of circumstances: 1. VoiceMailMain was + configured in the dialplan with an extension as its argument 2. A + message was left for this mailbox 3. Tried to call VoiceMailMain + but hung up before entering password. This was fixed by checking + that a pointer was non-null prior to trying to dereference it. + (Issue 9810, reported by xmarksthespot, patched by Corydon76 with + modifications by me). + +2007-06-06 16:55 +0000 [r67716] Russell Bryant + + * main/channel.c, /, include/asterisk/linkedlists.h: Merged + revisions 67715 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | + 5 lines We have some bug reports showing crashes due to a double + free of a channel. Add a sanity check to ast_channel_free() to + make sure we don't go on trying to free a channel that wasn't + found in the channel list. (issue #8850, and others...) ........ + +2007-06-06 13:30 +0000 [r67594-67650] Joshua Colp + + * main/rtp.c, /: Merged revisions 67649 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 + lines Reinvite the RTP back to the Asterisk machine when the + timeout happens. (issue #9888 reported by gasparz) ........ + + * main/translate.c: Fix plc_samples warning when registering a + translator. (issue #9897 reported by xylome) + + * apps/app_directed_pickup.c: Include macroexten while searching + for a channel to pick up in case they are in a macro. (issue + #9491 reported by jamesb63) + + * res/res_agi.c: Make the new "agi debug off" CLI command work. + (issue #9890 reported by eliel) + + * /, main/devicestate.c: Merged revisions 67593 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2 + lines Revert channel name splitting fix for Zap. The moral of the + story is don't use - in your user/peer names. (issue #9668 + reported by stevedavies) ........ + +2007-06-05 23:01 +0000 [r67558] Russell Bryant + + * apps/app_meetme.c: Fix some crashes related to the use of the + "meetme" CLI command. The code for this command was not locking + the conference list at all. (issue #9351, reported by and patch + submitted by Junk-Y, committed patch is different and by me) + +2007-06-05 21:30 +0000 [r67526] Steve Murphy + + * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug + 9883, wherein macros were not allowing the includes construct. + fixed and tested, looks OK. Now includes can serve as an adjunct + to catch. + +2007-06-05 20:53 +0000 [r67457-67492] Russell Bryant + + * include/asterisk/linkedlists.h: This bug has been hanging over my + head ever since I wrote this SLA code. Every time I tried to go + debug it by adding some debug output, the behavior would change. + It turns out I wasn't crazy. I had the following piece of code: + if (remove) AST_LIST_REMOVE_CURRENT(...); Well, + AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my + conditional statement didn't do much good at all. It always ran + at least all of the macro minus the first statement, so I was + seeing list entries magically disappear when they weren't + supposed to. After many hours of debugging, I have come to this + extremely irritating fix. :) (issues #9581, #9497) + + * channels/chan_zap.c: Suppress a bunch of debug output unless + option_debug is on + +2007-06-05 18:32 +0000 [r67424] Mark Michelson + + * apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails + saved to IMAP storage using extensions other than gsm were unable + to be played over the phone. (Issue 9786, reporter: + xmarksthespot, Patched by xmarksthe spot with revisions by me, + reviewed by Russell Bryant). + +2007-06-05 18:18 +0000 [r67421] Jason Parker + + * channels/chan_skinny.c: Correctly update date/time on devices + throughout the life of the device, instead of just at + registration. Issue 9152, yet another patch by DEA. + +2007-06-05 18:17 +0000 [r67420] Steve Murphy + + * pbx/pbx_ael.c: Added code to automatically add a default case to + switches that don't have one. In some cases, rather than fall + thru, it results in a goto with -1 result, which terminates the + extension; a sort of dialplan seqfault, sort of. This was + required to fix bug reported in 9881 + +2007-06-05 17:07 +0000 [r67360-67372] Russell Bryant + + * main/channel.c: Handle a failure in malloc() in + ast_safe_string_alloc() + + * main/channel.c: Fix a problem that showed itself by causing Zap + channel names to be completely bogus on my machine. + ast_safe_string_alloc() was broken. It called vsnprintf() on a + va_args list twice without re-initializing it. After the first + usage, va_end() and va_start() must be called again. + +2007-06-05 16:14 +0000 [r67329-67334] Christian Richter + + * /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) | + 1 line briding is a bool, fixed copy and paste issue. ........ + + * channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 + Jun 2007) | 1 line simplified the EVENT_SETUP handling in the + cb_events function a lot. Commented the different possibilities a + bit and made functions of shared code. When the dialed extension + does not exist in the extensions.conf we'll jump into the 'i' + extension if this does exist, else we disconnect the call with + the cause:1 = No Route to Destination. ........ + +2007-06-05 15:51 +0000 [r67308] Russell Bryant + + * main/asterisk.c, main/loader.c, include/asterisk/module.h: When + shutting down "gracefully", go through and run the unload() + callbacks for all of the modules. "stop now" is considered a + non-graceful shutdown and will not go through this process. + (issue #9804, reported by chrisost, patch by me) + +2007-06-05 15:22 +0000 [r67304] Joshua Colp + + * channels/chan_iax2.c: Only muck with the thread structure if an + idle one was found/created. + +2007-06-05 14:35 +0000 [r67270] Kevin P. Fleming + + * channels/chan_iax2.c: ensure that a burst of full frames + (AST_FRAME_DTMF being the prime example) will not be processed + out of order... this is a brute force fix, but seems to be the + safest fix for now (thanks to the Digium PQ department for + finding this bug) + +2007-06-05 10:25 +0000 [r67210] Christian Richter + + * channels/misdn_config.c, channels/chan_misdn.c, /, + channels/misdn/chan_misdn_config.h: Merged revisions 67209 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | + 1 line added possibility to deactivate bridging per port ........ + +2007-06-04 23:43 +0000 [r67162] Tilghman Lesher + + * /, funcs/func_math.c: Merged revisions 67161 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007) + | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops. + ........ + +2007-06-04 23:31 +0000 [r67158] Russell Bryant + + * channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt + structure can disappear and the code did not account for it and + crashes. (issues #9642, #9569, #9666, probably others ... based + on the work by stevedavies and mihai, with additional changes + from me) + +2007-06-04 23:26 +0000 [r67121-67156] Jason Parker + + * channels/chan_skinny.c: Fix for skinny keepalives. If there is no + traffic from the phone for (keep_alive * 1100) ms (arbitrarily + adding 10% for network issues, etc), unregister the device. Issue + 8394, patch by DEA. + + * channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar + to the recent fix for chan_skinny) Issue 9855, patch by DEA. + +2007-06-04 22:28 +0000 [r67119] Russell Bryant + + * channels/chan_iax2.c: Add comments for two functions that get + called with the appropriate call locked, but perform operations + that could result in the pvt structure getting destroyed before + returning again, causing numerous seg faults all over the module. + (inspired by issues #9642, #9569, and #9666, and the work done by + stevedavies and mihai) + +2007-06-04 21:59 +0000 [r67073] Steve Murphy + + * main/cdr.c: This typo has been here since 1.4 forked. It has been + the source of heartburn to many a dialplan/CDR programmer. + +2007-06-04 21:47 +0000 [r67071] Russell Bryant + + * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC) + +2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp + + * channels/chan_sip.c: Better handle SIP devices that say they have + SDP content... but really don't. (issue #9398 reported by + mthomasslo) + + * apps/app_dial.c: Initialize cidname variable to nothing since it + may be used without having been touched. (issue #9661 reported by + dimas) + + * res/res_features.c: Returning a value that indicates the parking + of a call was a success when it really wasn't (because the + parking slot selected was in use) is the wrong thing to do. + (issue #9723 reported by mdu113) + +2007-06-04 17:11 +0000 [r67061] Tilghman Lesher + + * contrib/init.d/rc.debian.asterisk, + contrib/init.d/rc.mandrake.asterisk, /, + contrib/init.d/rc.redhat.asterisk, + contrib/init.d/rc.gentoo.asterisk, + contrib/init.d/rc.mandrake.zaptel, + contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007) + | 2 lines Add revision Id tags (by request of tzafrir) ........ + +2007-06-04 16:02 +0000 [r67026] Russell Bryant + + * configure, configure.ac: Change the configure script to build a + test program against libcurl to make sure the results from + curl-config can be used to compile successfully. This is intended + to help prevent a situation where you are cross compiling, and + the configure script finds the curl library installed on the + host. (issue #9865, reported and patched by zandbelt) + +2007-06-04 15:50 +0000 [r67021] Tilghman Lesher + + * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash + +2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant + + * channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When + handling an implicit ACK to a frame that was marked as the final + transmission for a call, don't call iax2_destroy() for that call + while the global frame queue is still locked. There is a very + nice explanation of the deadlock in the report. (issue #9663, + thorough report and patch from stevedavies, additional positive + test reports from mihai and joff_oconnell) + + * include/asterisk/stringfields.h: Fix some compiler warnings in + C++ modules. (issue #9866, reported by osk, patch by Corydon76) + +2007-06-01 21:45 +0000 [r66919] Tilghman Lesher + + * funcs/func_odbc.c: On some drivers, deallocating the statement + handle isn't enough. We also have to clear the cursor (nice, + Oracle) + +2007-06-01 21:31 +0000 [r66897-66917] Mark Michelson + + * apps/app_voicemail.c: Removing extraneous debugging lines from + revision 66897. Sorry :) + + * apps/app_voicemail.c: Submitting a fix for voicemail with IMAP + storage. Attachments with format specified as gsm were duplicated + (i.e. two attachments) were left. Thank you very much to + xmarksthespot for submitting the patch that fixed this. (Issues + 9787 and 8873, Reported by xmarksthespot and jerjer, patched by + xmarksthespot) + +2007-06-01 19:41 +0000 [r66879-66881] Russell Bryant + + * channels/chan_skinny.c: Changes to the way DTMF is handled in the + core broke dialing in chan_skinny. This patch makes chan_skinny + usable again. I did not end up testing this, but there are + multiple positive test reports listed in the bug report. (issue + #9596, reported by pj, testing by pj and mvanbaak, and the fix + was written by DEA) + + * apps/app_page.c: List app_meetme as a module that app_page + depends on. + +2007-05-31 23:03 +0000 [r66821] Tilghman Lesher + + * doc/asterisk.8: Issue 9850 - update preferred command line syntax + +2007-05-31 18:41 +0000 [r66775] Russell Bryant + + * res/res_speech.c, include/asterisk/app.h, + include/asterisk/speech.h: Change a couple of header files to not + use "new", which is a reserved keyword in C++. (issue #9830, + reported by osk) + +2007-05-31 17:15 +0000 [r66770] Tilghman Lesher + + * /, apps/app_macro.c: Merged revisions 66744 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007) + | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime. + Issue 8329 will remain unfixed for pbx_realtime, but only because + we lack core API to do it. ........ + +2007-05-31 16:14 +0000 [r66768] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 + lines It is now possible for this path of execution to have the + frame pointer be NULL, therefore we need to check for it before + trying to access it. (issue #9836 reported by barthpbx) ........ + +2007-05-30 23:26 +0000 [r66671] Mark Michelson + + * apps/app_voicemail.c: Fixed seg-faults when recording greetings + in voicemail with IMAP enabled. (Issue No. 9735, reported by + xmarksthespot, patched by me) + +2007-05-30 17:28 +0000 [r66602-66639] Joshua Colp + + * channels/chan_sip.c: Silly me for having out of date source! Oh + well... I'm still leaving my comment. + + * channels/chan_sip.c: When calling some peer/host that may not + exist/reply back... don't keep the dialog in memory for all of + eternity. + + * channels/chan_zap.c, channels/chan_features.c: Change how channel + names are generated a bit. (issue #9825 reported by eldadran) + +2007-05-29 21:56 +0000 [r66538] Tilghman Lesher + + * /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007) + | 2 lines If the value of a variable passed to FIELDQTY is blank, + then FIELDQTY should return 0, not 1. ........ + +2007-05-29 19:32 +0000 [r66474-66503] Olle Johansson + + * channels/chan_sip.c: Properly handle 408 request timeout - + according to the RFC, the dialog dies if a request in a dialog + gets this response. + + * channels/chan_sip.c: Don't issue hangup on hangup on hangup on + hangup (for jcmoore) + +2007-05-29 16:44 +0000 [r66437] Joshua Colp + + * main/rtp.c: Handle cases where a frame may have no data. (issue + #9519 reported by dmb) + +2007-05-29 16:07 +0000 [r66404-66414] Olle Johansson + + * channels/chan_sip.c: Don't reset hangupcause if we already have + one + + * channels/chan_sip.c: Tracking down hanging channels, killing them + one by one. Issue #9235 and related + +2007-05-29 15:43 +0000 [r66398] Joshua Colp + + * doc/datastores.txt: Update datastores documentation. (issue #9801 + reported by mnicholson) + +2007-05-29 09:41 +0000 [r66363] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 + lines Issue #9802 - Change inuse counter on CANCEL ........ + +2007-05-28 23:16 +0000 [r66312] Joshua Colp + + * channels/chan_zap.c: Make the usedistinctiveringdetection option + work again. (issue #9823 reported by premeau) + +2007-05-27 04:12 +0000 [r66244] Jason Parker + + * channels/chan_zap.c: I don't know what this was trying to do, but + it's clearly incorrect. Issues 9808 and 9809. + +2007-05-25 14:43 +0000 [r66160] Kevin P. Fleming + + * configure, configure.ac: have to check for OSP toolkit _after_ + checking for OpenSSL + +2007-05-25 14:41 +0000 [r66159] Tilghman Lesher + + * /, main/say.c: Merged revisions 66127 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007) + | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch + ........ + +2007-05-25 14:28 +0000 [r66157] Kevin P. Fleming + + * configure, configure.ac, channels/chan_gtalk.c, makeopts.in, + res/res_jabber.c: handle the GNUTLS library properly in the + configure script and build system don't build in OSP support + unless we have found and are allowed to use SSL support + +2007-05-24 22:23 +0000 [r66076] Russell Bryant + + * main/channel.c: if the string field init fails, clean up the + stuff that was allocated already + +2007-05-24 22:16 +0000 [r66074] Joshua Colp + + * main/slinfactory.c: Fix slinfactory logic when dealing with + frames coming in that may already be in the signed linear format. + +2007-05-24 22:07 +0000 [r66068-66070] Russell Bryant + + * main/channel.c: Check the result of ast_string_field_init() in + ast_channel_alloc() + + * main/rtp.c: Make 1.4 build on my machine, too.. + +2007-05-24 20:54 +0000 [r66029-66030] Jason Parker + + * configure: Rebuild configure script for previous ar fix. + + * configure.ac: Following moving strip to AC_PATH_TOOL, we need to + do something similar for ar. + +2007-05-24 20:42 +0000 [r65978-66026] Russell Bryant + + * configure, include/asterisk/autoconfig.h.in, configure.ac: + Checking for the strip application needs to be done with + AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross + compilation environments. + + * Makefile: Clear CFLAGS before running make for menuselect. (issue + #9784, reported by ovi, patch by me) + +2007-05-24 18:28 +0000 [r65965-65967] Kevin P. Fleming + + * channels/chan_gtalk.c: oops, use #ifdef instead of #if + + * channels/chan_gtalk.c: don't reference GnuTLS headers and + functions unless the configure script found it + + * main/rtp.c: don't use uninitialized variables + +2007-05-24 15:27 +0000 [r65902] Joshua Colp + + * main/manager.c: Add the ability to blacklist certain commands + from being executed using the Command AMI action. (issue #9240 + reported by junky) + +2007-05-24 15:26 +0000 [r65892-65901] Olle Johansson + + * channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk + core dump since the GnuTLS interface did not support + multithreading correctly. + + * channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN. + Patch by phsultan. Thanks! + +2007-05-24 15:16 +0000 [r65877-65883] Jason Parker + + * .cleancount: Update cleancount for that last commit - just for + good measure. + + * include/asterisk/translate.h, codecs/codec_speex.c, + main/translate.c, codecs/codec_ilbc.c: Fix handling of + zero-length frames when a codec is capable of native PLC. Issue + 9183, patch by Mihai. + +2007-05-24 15:08 +0000 [r65866] Dwayne M. Hubbard + + * funcs/func_math.c: merged qwell's func_math patch for issue 9507 + +2007-05-24 15:08 +0000 [r65863] Joshua Colp + + * main/rtp.c: I like it when the RTP stack compiles myself... + +2007-05-24 15:05 +0000 [r65857] Olle Johansson + + * channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues + when calling from gtalk to SIP over nat. + +2007-05-24 15:04 +0000 [r65842-65853] Russell Bryant + + * apps/app_festival.c: Ensure that frames are fully initialized. + This will probably fix getting weird timestamp log messages in + logs when using the Festival app. (issue #9781, patch by me) + + * main/rtp.c: Fix the calculation of the RTT for RTCP. The previous + code would result in oscillating and incorrect data. + Additionally, the RTT would sometimes report negative values due + to incorrect calculations. (issue #9601, patch from davetroy) + +2007-05-24 14:48 +0000 [r65841] Olle Johansson + + * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for + jingle + +2007-05-24 14:42 +0000 [r65839] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 + lines Allow RFC2833 to be negotiated when an INVITE comes in + without SDP and is not matched to a user or peer. (issue #9546 + reported by mcrawford) ........ + +2007-05-24 14:38 +0000 [r65836] Olle Johansson + + * channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan - + Fix "login" as component to jabber server. ...and, by accident, + fix a bug in chan_sip for stopping a loop on retransmits of BYE + requests. + +2007-05-24 09:37 +0000 [r65768] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 + Mai 2007) | 1 line we should only activate the generator in + chan_misdn, when asterisk hask not yet taken the call + (WAITING4DIGS state). Alerting audio will be generated fomr + asterisk for example. ........ + +2007-05-23 20:59 +0000 [r65677-65685] Kevin P. Fleming + + * channels/chan_iax2.c: start the delayed PBX when receive voice or + video full frames as well, and comment this delayed-PBX activity + + * /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) + | 2 lines ensure that variables are set on a newly created + channel before we start a PBX on it ........ + + * channels/chan_iax2.c: clear the 'delay PBX' flag when we are + ready to start the PBX + + * channels/chan_iax2.c: don't start a PBX on a new incoming IAX2 + channel until we have some sort of response to our ACCEPT (ACK or + anything else) + + * /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) + | 2 lines if we are going to set variables on a newly created + channel, it should be done *before* we start the PBX on it + ........ + +2007-05-23 13:07 +0000 [r65589] Russell Bryant + + * channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | + 3 lines Revert revision 62417 as someone reported problems with + it to Mark. This was related to issue #9588. ........ + +2007-05-22 20:25 +0000 [r65541] Kevin P. Fleming + + * build_tools/make_version: when building a version string for a + developer branch, include the base branch in the version string + +2007-05-22 18:40 +0000 [r65501] Russell Bryant + + * apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a + dependency for app_voicemail and chan_zap (Thanks to mnicholson + for pointing it out) + +2007-05-22 15:04 +0000 [r65452] Joshua Colp + + * apps/app_meetme.c: Remove a double const. + +2007-05-22 14:02 +0000 [r65408] BJ Weschke + + * apps/app_followme.c: Fix a problem with flag recognition. + +2007-05-22 13:09 +0000 [r65394] Russell Bryant + + * /, apps/app_queue.c: Merged revisions 65389 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) | + 4 lines Fix a memory leak that I just noticed in the device state + handling in app_queue. On most device state changes, it would + leak roughly 8 to 64 bytes (the length of the name of the + device). ........ + +2007-05-22 08:12 +0000 [r65342] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 + Mai 2007) | 1 line we stop the tones only when we're in the + pre-call phase, otherwise e.g. when in CONNECTED state we should + not stop tones when we receive an Information Message ........ + +2007-05-20 17:59 +0000 [r65250] Joshua Colp + + * res/res_agi.c: res_agi needs to export two symbols + (ast_agi_register and ast_agi_unregister) for usage by others. + (issue #9755 reported by mnicholson) + +2007-05-18 22:26 +0000 [r65200-65201] Steve Murphy + + * main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c + to main/cdr.c, and neither did I. This is the remainder of the + 9717 patch, the fix for the run-away FAIL status for a call + + * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions + 65172 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 + line This update will fix the situation that occurs as described + by 9717, where when several targets are specified for a dial, if + any one them reports FAIL, the whole call gets FAIL, even though + others were ringing OK. I rearranged the priorities, so that a + new disposition, NULL, is at the lowest level, and the + disposition get init'd to NULL. Then, next up is FAIL, and next + up is BUSY, then NOANSWER, then ANSWERED. All the related set + routines will only do so if the disposition value to be set to is + greater than what's already there. This gives the intended + effect. So, if all the targets are busy, you'd get BUSY for the + call disposition. If all get BUSY, but one, and that one rings is + not answered, you get NOANSWER. If by some freak of nature, the + NULL value doesn't get overridden, then the disp2str routine will + report NOANSWER as before. ........ + +2007-05-18 18:16 +0000 [r65041-65123] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 + lines Not getting an ACK to a 200 OK in the initial invite is + critical to the call. ........ + + * /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 + lines Issue 9235 - part of the problem, maybe not all. Please + retry with this patch (and no other patch) if you have problems + with hanging SIP channels. Thank you. A special Thank You to + WeBRainstorm that gave me access to his system. ........ + + * channels/chan_sip.c: - Adding support for putting calls OFF hold + with a re-invite with blank SDP. This was a bug found while doing + tests at SIPit in Antwerp. - In order to not duplicate code, I + restructured some of the code for putting calls on/off hold. + Thanks DEA for reminding me. This fix has been asleep in the + videocaps branch until now. + +2007-05-18 12:40 +0000 [r65039] Christian Richter + + * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged + revisions 65007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | + 1 line fixed a warning regarding Keypad encoding. encode the IE + sending_complete at the right position. ........ + +2007-05-18 10:37 +0000 [r64974] Olle Johansson + + * channels/chan_sip.c: Issue 9487 - stop media flows at hangup of + call + +2007-05-18 08:58 +0000 [r64904] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 + Mai 2007) | 1 line we *need* to send a PROCEEDING when + sending_complete is set, even if need_more_infos is requested. + ........ + +2007-05-18 02:48 +0000 [r64868] Russell Bryant + + * apps/app_queue.c: Fix a small bug I noticed while working on + something else. app_queue did not unregister its device state + monitoring callback in unload_module(). So, this would make + Asterisk crash on the first device state change after you unload + the module. + +2007-05-17 21:19 +0000 [r64820] Tilghman Lesher + + * /, include/asterisk/linkedlists.h: Merged revisions 64819 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007) + | 2 lines How is it that we never caught that this is returning + the opposite of our documentation, until now? ........ + +2007-05-17 16:53 +0000 [r64761] Jason Parker + + * apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4 + lines If we have a negative current message, we shouldn't go back + even further... Issue 9727. ........ + +2007-05-17 16:52 +0000 [r64756-64759] Russell Bryant + + * contrib/scripts/astxs (removed): Remove script that is no longer + functional since the build system was redone. (issue #9340, + reported by junky) + + * apps/app_dial.c: Increase the size of a buffer to support longer + dial strings for channels. (issue #9291, reported and fix + suggested by meni) + +2007-05-17 16:10 +0000 [r64720-64754] Joshua Colp + + * channels/chan_sip.c: Even more direct RTP setup fixes! Don't + allow a codec that isn't supported to creep into the SDP of + either side. (issue #9446 reported by marcelbarbulescu) + + * apps/app_voicemail.c: Fix authuser support. (issue #9740 reported + by xmarksthespot) + +2007-05-17 06:13 +0000 [r64686] Russell Bryant + + * README: Update the main README to reflect the new build process + for 1.4 and above. (issue #9725, patch by eliel) + +2007-05-16 11:01 +0000 [r64516-64609] Olle Johansson + + * /: Blocking patch already in this code + + * channels/chan_sip.c: Fix auth on BYE. (Different patch than for + 1.2) + + * channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE + + * channels/chan_sip.c: Final part of issue #9483 - fixing + transfer() of sip calls in the dial plan (twilson) + + * channels/chan_sip.c: Issue #9439 - properly handle username + parameters in SIP uri. + + * /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 + lines Support SIP uri's starting with SIP: and sip: (reported by + Tony Mountfield on the mailing list. Thanks!) ........ + + * /, channels/chan_sip.c: Merged following patch with a lot of + changes for 1.4 ------ Merged revisions 64514 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 + lines Issue #9726 - rlister - Better logging for ACL denials + While at it, also added better logging and handling of peers that + are not supposed to register. My patch, stole the issue report + from Russell. My apologies, Russell :-) ........ + +2007-05-16 08:44 +0000 [r64515] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 + Mai 2007) | 1 line in the case immediate=yes, we directly jump + into the dialplan, where people can use PlayTones to indicate a + Dialtone, so we don't need to to that by ourself. also we should + not do a dialtone_indicate for incoming calls on a TE port in + overlapdialmode. ........ + +2007-05-15 19:52 +0000 [r64353-64426] Russell Bryant + + * res/res_features.c: Properly fix a problem that occurs when you + set PARKINGEXTEN to an exten where a call is already parked. + (issue #9723, patch by me) + + * res/res_features.c: When someone requests a specific parking + space using the PARKINGEXTEN variable, ensure that no other + caller is already there. (issue #9723, reported by mdu113, patch + by me) + +2007-05-14 19:26 +0000 [r64324] Olle Johansson + + * channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit + more + +2007-05-14 19:13 +0000 [r64306] Russell Bryant + + * channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by + just ignoring it. An unknown indication will trigger an error and + cause sounds to stop, which in this case, is ringing. + +2007-05-14 18:52 +0000 [r64280] Olle Johansson + + * channels/chan_sip.c: Handle network errors, like host or network + unreachable, in a better way. This means that calls to hosts or + qualify (OPTION) messages will fail quicker if the TCP/IP stack + tells us that there is an issue. Since this is an unconnected UDP + socket, we will not get error messages directly in most cases, + but maybe on the second and third try. This is already + implemented in trunk. + +2007-05-14 18:48 +0000 [r64240-64278] Joshua Colp + + * codecs/codec_speex.c: Properly set datalen field when doing PLC + in codec_speex. (issue #9722 reported by mihai) + + * /, main/devicestate.c: Merged revisions 64275 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2 + lines Only perform stripping of - strings from the channel name + for Zap channels. Anywhere else we might remove a legitimate part + of a device name. (issue #9668 reported by stevedavies) ........ + + * main/channel.c: Fix scenario where if a phone that simply called + Echo() put itself on hold it could never get off hold. + +2007-05-14 13:58 +0000 [r64193] Steve Murphy + + * main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570, + worrisome CDR warnings have been removed, that are either not + helpful, or not relevant. + +2007-05-14 10:39 +0000 [r64157] Olle Johansson + + * main/channel.c: Add hangupcause when we lack codecs for + transcoding + +2007-05-12 22:27 +0000 [r64044-64114] Joshua Colp + + * channels/chan_sip.c: This concludes my final adventure with + bitmasks and the onhold flag. Would anyone care for some peanuts? + + * channels/chan_sip.c: Tweak hold flags some more. They can be of + three states when active: active, inactive, one direction. + + * channels/chan_sip.c: Ensure the onhold flag is set no matter what + when being put on hold. + +2007-05-11 20:16 +0000 [r63982] Jason Parker + + * main/manager.c: Hide manager password from "manager show user + foo". I realize that there are other ways to get this, but we + really don't need to just show it in plain text so easily. Issue + 9273, patch by junky + +2007-05-11 16:35 +0000 [r63905] Tilghman Lesher + + * contrib/scripts/safe_asterisk, Makefile, /: Merged revisions + 63903 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007) + | 2 lines Issue 9121 - fixups for safe_asterisk script ........ + +2007-05-11 16:05 +0000 [r63886] Russell Bryant + + * main/manager.c: When MD5 authentication is not possible because + there is no challenge present, either because the Challenge + action was never issued, or some other reason, give a proper + error message and return an error instead of claiming that the + user wasn't found. (reported by jsmith on IRC) + +2007-05-11 15:43 +0000 [r63872] Joshua Colp + + * res/res_features.c: Make the PARKINGEXTEN feature of parking + actually work. (issue #9708 reported by mdu113) + +2007-05-10 23:15 +0000 [r63830] Jason Parker + + * /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 + lines Fix an issue with trying to kill a thread before it gets + created. Issue 9709, patch by nic_bellamy. ........ + +2007-05-10 22:23 +0000 [r63804] Russell Bryant + + * main/manager.c: Strip terminal escape sequences from CLI command + output that is going to be sent out over the manager interface. + (issue #9659, reported by pari, fixed by me) + +2007-05-10 20:48 +0000 [r63750] Doug Bailey + + * main/callerid.c: Add test for negative offsets in cid data to + prevent infinite loops. + +2007-05-10 20:46 +0000 [r63749] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 + lines Do not allocate SIP pvt's for PEERs we can not reach. This + was seen as a lot of dialogs being created then immediately + destroyed at reload/restart of the SIP channel. ........ + +2007-05-09 19:22 +0000 [r63656-63698] Joshua Colp + + * main/channel.c: Use the DTMF frame on the channel when returning + a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE. + + * channels/chan_sip.c: Do not prematurely go on hold if sendonly + was not actually set. + +2007-05-09 17:25 +0000 [r63654] Matthew Fredrickson + + * channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 + lines Make sure we only create a DSP if it's requested on + SUB_REAL ........ + +2007-05-09 16:55 +0000 [r63612] Russell Bryant + + * main/channel.c: Modify ast_senddigit_begin() to use the same + assumptions used elsewhere in the code in that if a channel does + not have a send_digit_begin() callback, it only cares about DTMF + END events. (pointed out by Michael Neuhauser on the asterisk-dev + list) + +2007-05-09 16:54 +0000 [r63611] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 + lines Properly handle hints that point to multiple devices in + chan_sip. Why chan_sip is even doing this I have no idea but I + would rather not go into a rant. (issue #9536 reported by + rlister) ........ + +2007-05-09 16:43 +0000 [r63608] Russell Bryant + + * main/channel.c: Only call ast_senddigit_begin() in + ast_senddigit() if the channel has a send_digit_begin() callback. + Checking the END_DTMF_ONLY flag was the wrong thing to do, + because that flag indicates that a *bridged* channel only wants + DTMF END events coming from this channel. + +2007-05-09 14:50 +0000 [r63566] Tilghman Lesher + + * /, apps/app_directory.c: Merged revisions 63565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007) + | 2 lines Replicate fix from 51158 (app_voicemail) to + app_directory (Issue 9224) ........ + +2007-05-09 13:24 +0000 [r63535] Russell Bryant + + * Makefile: I have seen multiple people post questions trying to + figure out what the message "The configure script must be + executed before running 'make'" means. So, add another like that + says to specifically run ./configure. If this isn't obvious + enough, then they should be using something like AsteriskNOW and + not installing from source. + +2007-05-09 13:17 +0000 [r63534] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, + channels/misdn/isdn_msg_parser.c: Merged revisions + 62945,63402,63519 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | + 1 line when we're in state WAITING4DIGS, we use the asterisk + tone-generator which prods us, so we can't just return -1 in + misdn_write in this case. Added a MISDN_KEYPAD channel variable, + and fixed the sending of keypad. this enables us to modify the + call forward parameters in the switch. ........ r63402 | crichter + | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added + application misdn_check_l2l1 which tries to pull up the L1/L2 on + all ports that have the layers down in a group. It waits then for + a timeout. This helps for scenarios where multiple PMP BRIs are + grouped together, or where a provider has a faulty PTP + Implementation, that looses the L2 after a while. ........ r63519 + | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line + release_chan frees ch, so we should never touch ch after + release_chan, this may cause segfaults. ........ + +2007-05-09 13:04 +0000 [r63532] Olle Johansson + + * channels/chan_sip.c: Don't retransmit 200 OK's on ignore status. + (Reported on asterisk-users) + +2007-05-08 22:38 +0000 [r63478] Tilghman Lesher + + * /, apps/app_macro.c: Merged revisions 63477 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007) + | 2 lines Issue 9602 - segfault in app_macro ........ + +2007-05-08 16:53 +0000 [r63403-63448] Russell Bryant + + * res/res_features.c: I mixed up the use of the find_feature() + function, so I renamed it find_dynamic_feature, and changed the + code to use the correct lock when using it. + + * res/res_features.c: Use a read/write lock when accessing the + built-in features. + + * contrib/scripts/realtime_pgsql.sql (added), + contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to + contrib/scripts to be with the rest of the sql examples. (issue + #9676, suretec) + +2007-05-08 06:22 +0000 [r63360] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007) + | 2 lines Issue 9527 - upon entering a folder, no message is + selected (curmsg == -1), so deleting causes memory corruption + (beyond bounds) ........ + +2007-05-07 22:28 +0000 [r63329] Russell Bryant + + * configs/res_pgsql.conf.sample (added), + configs/extconfig.conf.sample, contrib/realtime_pgsql.sql + (added): Add a sample configuration file and example tables for + use with res_config_pgsql. (issue #9676, suretec) + +2007-05-07 21:45 +0000 [r63283-63286] Joshua Colp + + * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged + revisions 63285 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 + lines Properly handle what happens during a masquerade in + relation to group counting. (issue #9657 reported by ramonpeek) + ........ + + * channels/chan_sip.c: Minor backport of revision 59083 in trunk. + Don't queue an unhold frame up if the call was never on hold to + begin with. + +2007-05-07 20:05 +0000 [r63196-63254] Olle Johansson + + * main/config.c: Don't remove configuration from memory just + because one section failed. + + * /: Guess svnmerge doesn't handle files that move around. Blocking + patch to ./config.c + +2007-05-06 12:28 +0000 [r63152] Olle Johansson + + * main/file.c: Stop the video stream when you stop playback of all + streams for a call + +2007-05-04 20:03 +0000 [r63099] Jason Parker + + * res/res_jabber.c: Fix a crash when checking version attribute in + an incoming XML caps element. Issue 9667, patch by phsultan. + +2007-05-04 16:45 +0000 [r63047] Pari Nannapaneni + + * configs/manager.conf.sample: explanation for httptimeout in + manager.conf + +2007-05-03 16:44 +0000 [r62989] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 + lines When a peer is seeded or built tell the devicestate core to + update it's status. This is easier then having chan_sip load + before pbx_config. (issue #9658 reported by dlynes) ........ + +2007-05-03 16:38 +0000 [r62986] Kevin P. Fleming + + * main/loader.c: improve loader a bit, by avoiding trying to + initialize embedded modules twice and avoiding trying to load + modules from disk when they have been loaded already during the + 'preload' pass (reported by blitzrage on IRC, patch by me) + +2007-05-03 15:23 +0000 [r62942] Russell Bryant + + * main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell + Bryant, 2007, TM, Patent Pending). This set of changes came from + a debugging session I had with Dwayne Hubbard. When he called + into his home FXO, ran the Echo application, and pressed a digit, + the digit would be echoed back and would never end. This is + fixed, along with a couple other little improvements. * When + chan_zap is in the middle of playing a digit to a channel, it + feeds back null frames, not voice frames. So, I have modified + ast_read to check the timing on emulated DTMF when it receives + null frames, in addition to where it was doing this on voice + frames. * Make a tweak to setting the duration on emulated DTMF + digits. If there was no duration specified, it set it to be the + minimum, instead of the default. * Instead of timing the emulated + digits off of the number of samples in audio frames that pass + through, just use time values. Now there is no code in this + section that assumes 8kHz audio. + +2007-05-03 14:41 +0000 [r62913] Steve Murphy + + * pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19 + (added), pbx/ael/ael-test/ael-test18/extensions.ael, + pbx/ael/ael-test/ael-test19/extensions.ael (added), + pbx/ael/ael-test/ael-test19 (added), + pbx/ael/ael-test/ref.ael-test20 (added), + pbx/ael/ael-test/ael-test20/extensions.ael (added), + pbx/ael/ael-test/ael-test20 (added): updated the ael regressions + to match what's in trunk + +2007-05-03 14:36 +0000 [r62912] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, + channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged + revisions 61357,61770,62885 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | + 1 line some fixes for PMP Hold/Retrieve, it should work now, when + briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200 + (Di, 24 Apr 2007) | 1 line added lock for sending messages to + avoid double sending. shuffled some empty_chans after the + cb_event calls, this avoids that a release_complete from a quite + different call releases a fresh created setup by accident. + ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 + Mai 2007) | 1 line fixed the problem that misdn_write did not + return -1 when called with 0 samples in a frame this resultet in + a deadlock in some circumstances, when the call ended because of + a busy extension. added encoding of keypad. ........ + +2007-05-03 13:54 +0000 [r62883] Steve Murphy + + * pbx/ael/ael-test/ref.ael-test18 (added), + pbx/ael/ael-test/ref.ael-vtest13, + pbx/ael/ael-test/ael-test18/extensions.ael (added), + pbx/ael/ael-test/ael-test18 (added), + pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c, + pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7: + These mods fix bug 9623, where an '@' in the eswitch contents + causes a syntax error. I also updated the regressions. + +2007-05-03 00:23 +0000 [r62797-62842] Kevin P. Fleming + + * res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02 + May 2007) | 2 lines doh... initializing the pointer variable will + work just a bit better ........ + + * res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02 + May 2007) | 7 lines increase reliability and efficiency of static + Realtime config loading via ODBC: don't request fields we aren't + going to use don't request sorting on fields that are pointless + to sort on explicitly request the fields we want, because we + can't expect the database to always return them in the order they + were created (reported by blitzrage in person (!), patch by me) + ........ + + * res/res_config_pgsql.c: improve static Realtime config loading + from PostgreSQL: don't request sorting on fields that are + pointless to sort on use ast_build_string() instead of snprintf() + don't request the list of fieldnames that resulted from the query + when we both knew what they were before we ran the query _AND_ we + aren't going to do anything with them anyway (patch by me, + inspired by blitzrage's bug report about res_config_odbc) + +2007-05-02 22:59 +0000 [r62739-62789] Russell Bryant + + * main/channel.c: Merge changes from team/russell/inband_dtmf ... + Fix some issues related to generating inband DTMF. There are two + changes here: 1) The list of DTMF tones in the senddigit_begin() + function explicitly specified 100ms of the tone followed by 100ms + of silence. This really broke things with the way that Asterisk + now wants complete control over when the digit begins and ends. + So, regardless of what Asterisk really wanted to do, this was + going to play out the tone at the length it wanted to. This + caused various problems like DTMF translation to inband to be + extremely unreliable. The list of tones has been changed so that + the correct DTMF tone is played indefinitely until Asterisk tells + it to stop. 2) ast_write() had to be modified to let a DTMF_END + frame get processed even when a generator is present. This is how + the tone will finally get stopped. (issues #8944, #9250, #9348, + maybe others. Thanks to mdu113 from #8944 for the testing and + feedback!) + + * main/manager.c: Backport the change that only went in to trunk + that fixes the command manager action over http. (reported + internally by pari and bkruse) + +2007-05-02 20:46 +0000 [r62738] Steve Murphy + + * main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May + 2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being + in 'h' extension louses up the dst field ........ + +2007-05-02 17:43 +0000 [r62692] Tilghman Lesher + + * /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) + | 4 lines Issue 9638 - if a text frame is sent with no + terminating NULL through a bridged IAX connection, the remote end + will receive garbage characters tacked onto the end. ........ + +2007-05-02 17:10 +0000 [r62689] Steve Murphy + + * configs/extensions.conf.sample, main/channel.c, main/pbx.c, + channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the + clid, src fields in channel_alloc call. b)in the channel_alloc + func, set the cid_num and name fields from the arglist[blush]. c) + don't update the channel app & app data fields if you are in the + 'h' extension. d)the load_module func in cdr_radius needs to + return DECLINE, SUCCESS. + +2007-05-02 06:15 +0000 [r62624] Olle Johansson + + * channels/chan_sip.c: Don't unlock a channel that we already know + does not exist (propably isue 8228) + +2007-05-01 21:57 +0000 [r62548] Russell Bryant + + * /, res/res_features.c: Merged revisions 62547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | + 4 lines Remove an unnecessary check that makes it so if you hang + up after doing an attended transfer before the target extension + answers the channel, the transfer is not successful. (issue + #9338, patch by svanlund) ........ + +2007-05-01 21:34 +0000 [r62545] Tilghman Lesher + + * apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user() + (found by rayjay, different fixes by me) + +2007-05-01 16:26 +0000 [r62497] Russell Bryant + + * /, configs/indications.conf.sample: Merged revisions 62496 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | + 3 lines Add indications.conf information for the Philippines. + (issue #9525, reported and patched by loloski) ........ + +2007-04-30 15:58 +0000 [r62414-62419] Russell Bryant + + * channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | + 4 lines This patch fixes an issue where depending on the cause + code, when the network sends a PRI disconnect, the call may not + be properly hung up. (issue #9588, reported and patched by + softins) ........ + + * include/asterisk/http.h, main/http.c: When serving dynamic + content, include a Cache-Control header to instruct the browsers + to not store the resulting content. (issue #9621, reported by + Pari, patch by me) + +2007-04-30 14:52 +0000 [r62371] Jason Parker + + * configs/iax.conf.sample: Remove unused (and potentially + confusing) jitterbuffer options from sample config. + +2007-04-30 14:36 +0000 [r62369] Joshua Colp + + * main/asterisk.c, /: Merged revisions 62368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2 + lines Update copyright notice. It's now the year 2007! ........ + +2007-04-29 05:50 +0000 [r62299-62331] Russell Bryant + + * channels/chan_zap.c: Fix a bug that made the "language" setting + in zapata.conf not functional. (issue #9626, reported and fixed + by sergee) + + * apps/app_meetme.c: Note that the "talker optimization" option + will be enabled by default in 1.6 + +2007-04-27 Russell Bryant + + * Asterisk 1.4.4 released. + +2007-04-27 21:10 +0000 [r62218] Russell Bryant + + * channels/chan_agent.c: Fix a weird problem where when a caller + talking to someone sitting behind an agent channel sent a digit, + the digit would be played to the agent for forever. This is + because chan_agent always returned -1 from its send_digit_begin + and _end callbacks. This non-zero return value indicates to the + Asterisk core that it would like an inband DTMF generator put on + the channel. However, this is the wrong thing to do. It should + *always* return 0, instead. When the digit begin and end + functions are called on the proxied channel, the underlying + channel will indicate whether inband DTMF is needed or not, and + the generator will be put on that one, and not the Agent channel. + (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed + by me) + +2007-04-27 16:17 +0000 [r62174] Jason Parker + + * /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 + lines This transcoder message needn't be a NOTICE. I've seen it + cause confusion more than a few times. ........ + +2007-04-27 16:14 +0000 [r62171] Russell Bryant + + * main/pbx.c: If no variables were passed into + pbx_substitute_variables_helper_full(), then don't even bother + creating a temporary bogus channel, since that is only for + allowing certain functions to operate on the variables as if they + were on a channel. Most importantly, this fixes a crash. (issue + #9613, reported by callguy, fixed by me) + +2007-04-27 14:04 +0000 [r62095-62137] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 + lines Issue #7351 - SIP Cancel fails due to the wrong contact + uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka + - THANKS!!!! THis was a hard one to catch. ........ + + * channels/chan_zap.c, main/manager.c: Issue #9608 - fix some + annoying DEBUG messages not controlled by option_debug (DEA). + Thanks! + +2007-04-26 16:33 +0000 [r61959-62038] Joshua Colp + + * /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 + lines Revert previous fix for when the IAX2 channel goes funky + (that's the technical term). This is causing legit calls to be + prematurely hung up. (issue #9600 reported by justdave) ........ + + * main/channel.c: Missed an ast_app_group_discard during merge. + Thanks blitzrage! + + * res/res_monitor.c: Don't always say that the channel is being + paused if it is actually being unpaused in the Manager ack + message. (reported by jsmith in #asterisk-bugs) + + * main/config.c, /: Merged revisions 61958 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2 + lines Don't count failed include attempts against the + configuration include level. (issue #9593 reported by mostyn) + ........ + +2007-04-25 22:29 +0000 [r61914] Kevin P. Fleming + + * channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) + | 2 lines handle a very bizarre race condition with channels + being redirected before a simple switch can be started on them + (issue #9286) ........ + +2007-04-25 21:59 +0000 [r61863-61870] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | + 2 lines If the callerid= option is specified, but empty, clear + any previous data. ........ + + * /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | + 2 lines Ensure that callerid settings are reset on a reload. + ........ + +2007-04-25 19:21 +0000 [r61805] Joshua Colp + + * main/cli.c, main/channel.c, include/asterisk/app.h, + funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 + lines Merge rewritten group counting support. No more storing + data on the variable list of the channels. That was bad, mmmk? + (issue #7497 reported by sabbathbh) ........ + +2007-04-25 16:22 +0000 [r61799] Russell Bryant + + * channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | + 3 lines Fix a typo where cid_num got copied instead of cid_ani. + (issue #9587, reported and patched by xrg) ........ + +2007-04-24 Russell Bryant + + * Asterisk 1.4.3 released. + +2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant + + * main/manager.c, /: Merged revisions 61786 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) | + 4 lines Don't crash if a manager connection provides a username + that exists in manager.conf but does not have a password, and + also requests MD5 authentication. (ASA-2007-012) ........ + + * main/channel.c, include/asterisk/channel.h: Improve DTMF handling + in ast_read() even more in response to a discussion on the + asterisk-dev mailing list. I changed the enforced minimum length + of a digit from 100ms to 80ms. Furthermore, I made it now enforce + a gap of 45ms in between digits. These values are not + configurable in a configuration file right now, but they can be + easily changed near the top of main/channel.c. + +2007-04-24 18:43 +0000 [r61779] Dwayne M. Hubbard + + * channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) + | 1 line removed #if 0 block from chan_phone, chan_zap, and + chan_modem restart_monitor() ........ + +2007-04-24 16:16 +0000 [r61774] Russell Bryant + + * main/dial.c: Add a few more state changes in + handle_frame_ownerless() so that the SLA code will get notified + of these changes even when an owner channel is not provided. This + isn't from a specific bug report, it's just something I noticed + while poking around. + +2007-04-24 16:07 +0000 [r61772] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 + lines Allow RFC2833 to be sent in the response SDP when an INVITE + comes in without SDP. (issue #9546 reported by mcrawford) + ........ + +2007-04-23 18:17 +0000 [r61763-61765] Russell Bryant + + * main/pbx.c: Some dialplan functions, such as CUT(), expect to + operate on variables on a channel. So, this little hack lets them + work in places where a channel doesn't exist, such as within + DUNDi configuration. (issue #9465, reported and patched by + Corydon76, testing by blitzrage) + + * main/channel.c: Ensure that digits passing through Asterisk have + a reasonable minimum length. It is currently 100 ms. If someone + thinks this should be different, feel free to speak up. (related + to issues #8944, #9250, and #9348) + +2007-04-20 21:35 +0000 [r61705-61707] Jason Parker + + * main/rtp.c: Avoid invalid seqno cycling detection. Per comment + from Dave Troy: This adds back in some simple typecasting I had + in an earlier version which I realize now may be breaking things. + Issue #9554. + + * main/loader.c, /: Merged revisions 61704 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4 + lines Fix an issue that I noticed while looking over issue 9571. + The reload timestamp was getting set after reloading the built-in + stuff, and before the modules. ........ + +2007-04-20 20:42 +0000 [r61697] Russell Bryant + + * main/rtp.c: Remove a stray debug message introduced by a recent + commit. + +2007-04-20 19:51 +0000 [r61694] Jason Parker + + * /, apps/app_queue.c: Merged revisions 61692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5 + lines If the '* to hangup' option is not enabled, we don't need + to disable * as a valid exit key. If it was enabled, this + statement would've never been checked in the first place. Issue + #9552 ........ + +2007-04-20 18:19 +0000 [r61690] Russell Bryant + + * main/config.c, apps/app_voicemail.c, main/manager.c, + include/asterisk/config.h: Fix the UpdateConfig manager action to + properly treat "variables" and "objects" differently (a=b versus + a=>b). (issue #9568, reported by pari, patch by me) + +2007-04-19 08:37 +0000 [r61686] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3 + lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by + Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........ + +2007-04-19 04:36 +0000 [r61681-61683] Tilghman Lesher + + * main/manager.c: Bug 9557 - simple reason why reading a function + always returned NULL + + * funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c, + funcs/func_groupcount.c, /, funcs/func_timeout.c, + funcs/func_cdr.c: Merged revisions 61680 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007) + | 5 lines Bug 9557 - Specifying the GetVar AMI action without a + Channel parameter can cause Asterisk to crash. The reason this + needs to be fixed in the functions instead of in AMI is because + Channel can legitimately be NULL, such as when retrieving global + variables. ........ + +2007-04-18 22:10 +0000 [r61678] Kevin P. Fleming + + * sounds/Makefile: allow external build systems to extract the + required sound file versions + +2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson + + * main/rtp.c: Clean upp formatting, add some doxygen stuff while + we're in cleaning mode... Thanks Kevin! + + * main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy) + +2007-04-16 14:47 +0000 [r61664-61666] Olle Johansson + + * channels/chan_sip.c: #9483, half of patch by twilson to solve 302 + redirect issues + + * /: Blocking AstHoloPatch from 1.2 + +2007-04-13 21:17 +0000 [r61658] Steve Murphy + + * main/cdr.c: This is a fix to the way CDR merge handles the data + that results from ForkCDR. + +2007-04-13 19:17 +0000 [r61648-61656] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 61655 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 + lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves + the same as OUTBOUND_GROUP except it will get unset after use so + it won't get accidentally inherited. (issue #BE-140) ........ + + * apps/app_speech_utils.c: Do not bother looking for a result if + none are present. + + * channels/chan_sip.c: For those very verbose SIP implementations + that attach tons of info to the Contact header... let's increase + our variable sizes. (issue #9535 reported by jeffg) + +2007-04-13 17:10 +0000 [r61645] Russell Bryant + + * apps/app_voicemail.c: Eliminate a compiler warning with + ODBC_STORAGE enabled so that it will build under dev-mode. + +2007-04-13 17:01 +0000 [r61644] Steve Murphy + + * channels/chan_oss.c: A fix for chan_oss that resulted from the + CDR changes; it helps to use the right info. + +2007-04-13 16:32 +0000 [r61641] Joshua Colp + + * channels/chan_sip.c: Don't assume the callid of a dialog will be + set, as in some circumstances it may not. (issue #9534 reported + by tecnoxarxa) + +2007-04-11 16:05 +0000 [r61477] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | + 5 lines If someone sets the "useragent" option in sip.conf to be + empty, then don't add the User-Agent header at all. It is an + optional header, anyway. Also, the bug report says that some of + Japan's SIP providers don't allow it for some weird reason. + (issue #9488, reported by makoto, fixed by me) ........ + +2007-04-11 15:39 +0000 [r61443] Nadi Sarrar + + * channels/chan_misdn.c: Don't export AOCD variables on + misdn_hangup anymore, this was mainly a fix for trunk.. + +2007-04-11 15:09 +0000 [r61377-61427] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | + 6 lines Fix a bug with switching between host=dynamic and using + specific hosts for peers. The code would only reset the peer's + address when it is dynamic if it was a new peer structure. Now, + it will also reset the address if it was already in the peer + list, but before the reload, it was not dynamic. (issue #9515, + reported by caio1982, fixed by me) ........ + + * main/http.c: Add "svgz" to the mimetypes table. (issue #9510, + bkruse) In passing, constify the elements of the mimetypes table. + + * /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | + 5 lines Remove the attempt at reporting configuration errors in + sip.conf. This can cause a bunch of improper messages when using + realtime. I give up. As oej tried to convince me when I put this + in, there is just no easy way to do it. (inspired by a message on + the -dev list) ........ + +2007-04-11 13:40 +0000 [r61342-61373] Nadi Sarrar + + * channels/chan_misdn.c: Export AOCD variables on misdn_hangup. + + * channels/chan_misdn.c: Ignore facility messages in case we don't + have a corresponding channel object. + + * channels/chan_misdn.c: AOCD's are now exported to asterisk + channel variables. + +2007-04-10 16:05 +0000 [r61220] Russell Bryant + + * main/Makefile, main/http.c, main/minimime (removed): File upload + support was added to solve some needs for the Asterisk GUI. + However, after much discussion, it has been decided that adding + this to 1.4 is not in the best interests of the project. It has + been removed here, but will remain in trunk. + +2007-04-10 12:43 +0000 [r61183] Nadi Sarrar + + * channels/misdn_config.c, /: Merged revisions 61170 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr + 2007) | 2 lines msns config parameter defaults to '*' ........ + +2007-04-10 05:18 +0000 [r61136] Steve Murphy + + * apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a + previous fix to overcome a compiler warning; the app NoCDR() has + been updated to mark the channel CDR as POST_DISABLED instead of + destroying the CDR; this way its flags are propagated thru a + bridge and the CDR is actually dropped. The cases where only one + channel in a bridge has a CDR was cleaned up. + +2007-04-09 19:58 +0000 [r61072] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 + lines - Don't send ActionID before Response: header. - Don't use + a blank in an AMI header ........ + +2007-04-09 19:55 +0000 [r61062-61070] Kevin P. Fleming + + * main/minimime/mm_envelope.c, res/res_features.c: fix up some + warnings found using --enable-dev-mode + + * main/minimime/Doxyfile (removed), + main/minimime/tests/messages/CVS (removed), + main/minimime/tests/CVS (removed): remove some more stuff we + don't need + +2007-04-09 19:41 +0000 [r61042-61044] Russell Bryant + + * main/minimime/test (removed): Remove another directory that + should no longer be there + + * main/minimime/Make.conf (removed), main/minimime/mytest_files + (removed), main/minimime/.cvsignore (removed), main/minimime/sys + (removed), main/minimime/mm-docs (removed): Remove various files + that I thought I already removed. + +2007-04-09 19:05 +0000 [r61022] Jason Parker + + * apps/app_queue.c: Use the appropriate interface name with + COMPLETECALLER. Issue 9395. + +2007-04-09 18:32 +0000 [r60989] Steve Murphy + + * channels/chan_oss.c, main/channel.c, main/cdr.c, + channels/chan_phone.c, channels/chan_misdn.c, + channels/chan_skinny.c, channels/chan_features.c, + channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c, + channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, + channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c, + channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, + include/asterisk/channel.h, channels/chan_gtalk.c, + channels/chan_iax2.c: This is a big improvement over the current + CDR fixes. It may still need refinement, but this won't have as + many folks bothered. + +2007-04-09 18:02 +0000 [r60984] Olle Johansson + + * res/res_jabber.c: Add final new line after JabberEvent + +2007-04-09 17:22 +0000 [r60936] Jason Parker + + * /, apps/app_directory.c: Merged revisions 60935 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5 + lines Allow matching on names shorter than 3 chars. This also + fixes the case where somebody wants to match on less then 3 + chars. Issue 9071 ........ + +2007-04-09 03:01 +0000 [r60847-60850] Tilghman Lesher + + * main/asterisk.c, include/asterisk.h, /: Merged revisions 60849 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) + | 2 lines Don't check for error when lowering priority (according + to the manpage, it should never happen anyway). It might could + happen, though, if another thread messed with the priority, so + safeguard against that (reported via -dev list). ........ + + * channels/chan_local.c, /: Merged revisions 60846 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 + Apr 2007) | 2 lines Bug 9505 - If the return value for + local_queue_frame is set, then p->lock is no longer valid. + ........ + +2007-04-09 01:03 +0000 [r60762-60798] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 60797 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2 + lines When calling a device that then forwards us elsewhere... we + have to make our channels compatible if it is the only channel + being dialed. (issue #9445 reported by marcelbarbulescu) ........ + + * apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if + MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy) + +2007-04-08 14:14 +0000 [r60661-60713] Tilghman Lesher + + * /, apps/app_macro.c: Merged revisions 60711 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007) + | 2 lines Gosub called within a Macro resets the arguments + improperly and causes general weirdness. (Issue 8329) ........ + + * main/http.c: Fix --enable-dev-mode + + * channels/chan_oss.c: Off by one error, resulting in a crash + (Issue 9500) + + * /, main/file.c: Merged revisions 60660 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007) + | 2 lines Bug 9486 - memory leak when opening a filestream + ........ + +2007-04-06 20:58 +0000 [r60603] Russell Bryant + + * main/minimime/sys/mm_queue.h, main/minimime/Doxyfile, + main/minimime/mimeparser.yy.c, main/minimime/minimime.c, + main/manager.c, main/minimime/mm_mimepart.c, + main/minimime/test.sh, configure, include/asterisk/compat.h, + main/strcompat.c, main/minimime/mm_internal.h, main/http.c, + main/minimime/tests/parse.c, main/minimime/mm_base64.c, + main/minimime/mm_mimeutil.c, main/minimime/mm.h, + main/minimime/tests, main/minimime/mm_header.c, + main/minimime/mm_error.c, main/Makefile, + main/minimime/mm_codecs.c, main/minimime/mm_param.c, + configure.ac, main/minimime/Makefile, main/minimime/mm_init.c, + include/asterisk/manager.h, main/minimime/strlcpy.c, + configs/http.conf.sample, main/minimime/mm_parse.c, + main/minimime/tests/create.c, main/minimime/mm_contenttype.c, + main/minimime/mm_util.c, main/minimime/mm_envelope.c, + main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c, + main/minimime/tests/messages/test2.txt, + main/minimime/tests/messages/test3.txt, + main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c, + main/minimime/tests/messages/test4.txt, + main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h, + main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c, + main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt, + main/minimime/mimeparser.l, main/minimime/mm_context.c, + main/minimime/mimeparser.tab.h, main/minimime (added), + main/minimime/mm_warnings.c, main/minimime/mm_queue.h, + main/minimime/tests/messages, include/asterisk/autoconfig.h.in, + main/minimime/mimeparser.y, Makefile.moddir_rules, + main/minimime/sys, main/minimime/tests/Makefile: To be able to + achieve the things that we would like to achieve with the + Asterisk GUI project, we need a fully functional HTTP interface + with access to the Asterisk manager interface. One of the things + that was intended to be a part of this system, but was never + actually implemented, was the ability for the GUI to be able to + upload files to Asterisk. So, this commit adds this in the most + minimally invasive way that we could come up with. A lot of work + on minimime was done by Steve Murphy. He fixed a lot of bugs in + the parser, and updated it to be thread-safe. The ability to + check permissions of active manager sessions was added by Dwayne + Hubbard. Then, hacking this all together and do doing the + modifications necessary to the HTTP interface was done by me. + +2007-04-06 20:32 +0000 [r60568-60572] Dwayne M. Hubbard + + * UPGRADE.txt: clarified a sentence in the format_wav section + + * UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and + plan to remove GAIN code from trunk + +2007-04-06 19:50 +0000 [r60521-60565] Russell Bryant + + * apps/app_meetme.c: When a station picks up a trunk that was on + hold, make the hints reflect that nobody has the trunk on hold + anymore. + + * apps/app_meetme.c: Fix a few problems with SLA. (issue #9459, + reported by francesco_r, fixed by me) * The original behavior was + that if one station put a call on hold, another one picked it up, + and then hung up, the code would still consider the call on hold + by the first station, so the trunk would not be hung up. However, + to better comply with what most people seem to expect it to + behave, it will now hang up the trunk. * Fix a problem with + "barge=no". This was only intended to prevent people from joining + calls that are in progress. However, it also prevented other + people from picking up a call that was on hold. This has been + fixed. * When there are no active stations on a trunk and it is + on hold, the code now indicates the HOLD and UNHOLD conditions to + the trunk channel. This allows music on hold to be played to the + trunk when it is on hold. + +2007-04-06 18:21 +0000 [r60459-60485] Matt Frederickson + + * channels/chan_zap.c: Make sure we check the faxdetect option + before doing fax processing + + * channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 + lines There should only be one code path for doing DTMF + conditionals on channels. This fixes it. ........ + +2007-04-06 14:49 +0000 [r60399] Kevin P. Fleming + + * /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007) + | 2 lines remove undocumented 'cardsmode' parameter and stop + searching for transcoders during reload() ........ + +2007-04-06 01:14 +0000 [r60361] Joshua Colp + + * res/res_speech.c, apps/app_speech_utils.c, + include/asterisk/speech.h: Add support for returning different + types of results (ie: NBest). + +2007-04-05 22:58 +0000 [r60325] Dwayne M. Hubbard + + * formats/format_wav.c: modified default GAIN for issue 5823, + thanks jrwalliker + +2007-04-05 22:35 +0000 [r60323] Steve Murphy + + * configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added + some clarification to the example configs for CDRs, on how to + select a backend. Also, made cdr-csv the default if you 'make + samples', and no other changes. + +2007-04-05 16:10 +0000 [r60268] Jason Parker + + * apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5 + lines Just because we can't find the voicemail configuration + file, doesn't mean that the module failed to load. The user could + be using realtime. Issue #9473 ........ + +2007-04-05 15:47 +0000 [r60265] Russell Bryant + + * main/http.c: Add the MIME type for gif by request from Pari + +2007-04-05 12:55 +0000 [r60214] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 + lines Only unlock our pvt and net locks if we are actually going + to try to lock the owner again. (issue #9472 reported by zoa) + ........ + +2007-04-04 17:40 +0000 [r60013-60137] Russell Bryant + + * main/manager.c, /: Merged revisions 60134 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) | + 6 lines It is valid to redirect channels via the manager + interface that are not in the UP state. Instead of checking for + that to prevent to ensure a dead channel doesn't get redirected, + just use the ast_check_hangup() API call. (issue #9457, reported + by Callmewind, patch by me) (related to issue #8977) ........ + + * channels/chan_sip.c: Add a Content-Length of 0 to the response + built by transmit_response_with_unsupported(). (issue #9454, + reported by makoto, fixed by me) + + * /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | + 4 lines Fix the return value of handle_common_options() so that + it always properly indicates whether it handled the option or + not. (issue #9455, reported by Netview, fixed by me) ........ + + * apps/app_meetme.c: Fix a problem where if a trunk was hung up + while it was on hold, all of the hints would reflect the line + still on hold, even though it should reflect that it is back to + not in use. (issue #9459, reported by francesco_r, fixed by me) + + * /: Blocked revisions 60016 via svnmerge ........ r60016 | russell + | 2007-04-03 18:23:23 -0500 (Tue, 03 Apr 2007) | 3 lines Add a + missing "\r\n" in the body of the NOTIFY that is sent to indicate + the status of a transfer. (issue #9388, reported by rarritt) + ........ + + * /: Blocked revisions 60014 via svnmerge ........ r60014 | russell + | 2007-04-03 18:00:10 -0500 (Tue, 03 Apr 2007) | 3 lines Use the + more generic check for "sed -r" support that was already present + in 1.4. (related to issue #9399) ........ + + * /: Blocked revisions 60012 via svnmerge ........ r60012 | russell + | 2007-04-03 17:54:49 -0500 (Tue, 03 Apr 2007) | 3 lines On + Darwin, the -r argument to sed is not valid. It has to be -E. + (issue #9399, reported by jcovert) ........ + +2007-04-03 19:40 +0000 [r59963] Joshua Colp + + * apps/app_speech_utils.c: Don't clash when a person both speaks + and uses DTMF. + +2007-04-03 19:16 +0000 [r59853-59939] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | + 4 lines Don't attempt to report configuration errors in + build_user(). oej pointed out that for a "friend" entry, this + won't work, because all user options are valid for peers, but not + the other way around. ........ + + * /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | + 3 lines Make chan_sip report when it encounters an unknown + option. (issue #9440, reported by nightcrawler) ........ + + * /, main/app.c: Merged revisions 59886 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) | + 5 lines When doing a built-in blind or attended transfer, restore + the ability to use '#' to terminate the number and immediately do + the transfer instead of having to dial the number and just wait + for the feature digit timeout. (issue #8366, xueliangliang) + ........ + + * Makefile: Ensure that menuselect gets executed in dependency + check mode every time you run make. + +2007-04-03 11:02 +0000 [r59804] Nadi Sarrar + + * channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h: + Merged revisions 59788,59803 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 + lines Use the new sysfs way of mISDN 1.2 to check if a port is NT + or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, + 03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........ + +2007-04-03 07:20 +0000 [r59774] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h: + Merged revisions 59623-59624,59639 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | + 1 line we can now make 30 channels on a PRI (before we forgot + chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200 + (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........ + r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | + 1 line added option which allows us to accept incoming SETUP + Messages without automatically sending Proceeding or Setup + Acknowledge, this is useful with some broken switches and if you + want to Release incoming calls without previously having + acknowledged them. The new option is + noautorespond_on_setup=yes|no default is no, so we don't break + the existing behaviour ........ + +2007-04-02 18:58 +0000 [r59724] Joshua Colp + + * apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2 + lines Increase the maximum size for a string of mailboxes to + 1024. (issue #9270 reported by rtucker) ........ + +2007-04-02 17:31 +0000 [r59688] Steve Murphy + + * pbx/pbx_ael.c: continue in for-loop should go to the incrementer, + not the test. As per 9435, thanks to marcelbarbulescu + +2007-04-02 15:39 +0000 [r59654] Russell Bryant + + * main/netsock.c, /: Merged revisions 59608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) | + 6 lines Add the SO_REUSEADDR flag to sockets handled by netsock. + This is needed by the patch that went in for issue 7874. + chan_iax2 needs to be able to create socket that is lisetning on + INADDR_ANY, but also be able to bind sockets to specific + addresses. (Thanks to Stevenson on the asterisk-dev mailing list + for explaining why this flag was needed.) ........ + +2007-03-30 22:50 +0000 [r59573] Jason Parker + + * configure, main/Makefile, acinclude.m4: Add linux-uclibc host + arch..."thingy". Sorry, I don't know what it's called... + +2007-03-30 17:51 +0000 [r59452-59522] Steve Murphy + + * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c, + include/asterisk/cdr.h: several changes via kpflemings review + + * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c, + include/asterisk/cdr.h: These mods fix CDR issues from 8221, + 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated + from transfer situations. + + * configs/extensions.conf.sample: A small clarification to keep + bugs from being filed, and confusion from rising, if + clearglobalvars is set, and globals are set in the AEL file. + (9419) + +2007-03-29 17:43 +0000 [r59363] Russell Bryant + + * res/res_jabber.c: When building a response to a subscription, the + "from" must be the full Jabber ID. This fixes some problems where + jabber users are not able to add their Asterisk account to their + user list, since they are unable to get Asterisk to approve their + subscription. (issue #8210, reported by caspy, and verified by + bradtem) + +2007-03-29 17:38 +0000 [r59361] Joshua Colp + + * /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2 + lines Keep a global array of variables indicating whether certain + conference rooms are in use. This ensures that two people going + into a new dynamic conference when the 'e' option is set don't go + into the same conference room. (issue #8835 reported by eliel) + ........ + +2007-03-29 17:17 +0000 [r59304-59358] Russell Bryant + + * main/rtp.c, /: Merged revisions 59357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | + 5 lines If an error occurs when reading from an RTP socket, and + the error code does not indicate that we should try again, then + return NULL instead of a "null frame". This will prevent Asterisk + from trying over and over again, and eventually causing the + system to crash. (issue #8285, john) ........ + + * /: Blocked revisions 59355 via svnmerge ........ r59355 | russell + | 2007-03-29 12:10:28 -0500 (Thu, 29 Mar 2007) | 3 lines Backport + the change to chan_iax2 to return NULL instead of a "null frame" + from its read callback. See revision 59341 to the 1.4 branch for + more info. ........ + + * channels/chan_iax2.c: When the IAX2 read callback gets called, + return NULL instead of a "null frame". This will cause Asterisk + to hangup the call instead of keep trying whatever it was doing. + Under normal conditions, this function would *never* be called. + However, the author of this patch says an error will occur that + will cause it to get called every 100 thousand calls or so. When + this does happen, it puts the channel in a loop that eventually + brings down the system. So, hangup up the call is certainly a + better alternative. (issue #8286, john) + + * Makefile: Export the GTK2 library and include information to sub + Makefiles. + +2007-03-29 16:07 +0000 [r59300-59302] Tilghman Lesher + + * /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007) + | 3 lines Issue 9415 - No point to getting a diagnostic field if + we aren't doing anything with the information. (Plus, it tends to + crash the Postgres ODBC driver.) ........ + + * /: Blocked revisions 59299 via svnmerge ........ r59299 | + tilghman | 2007-03-29 10:33:10 -0500 (Thu, 29 Mar 2007) | 2 lines + Change ENV section to use setenv, instead of putenv (Alexandru + Pirvulescu , reported via -dev list) ........ + +2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher + + * res/res_odbc.c: Another crash that I thought we had fixed already + - Issue 9396 + + * apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007) + | 2 lines Oops ........ + + * apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007) + | 2 lines Fix a few remaining bad mmap(2) return values ........ + +2007-03-27 23:20 +0000 [r59262-59278] Russell Bryant + + * /, apps/app_directory.c: Merged revisions 59277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) | + 3 lines Fix the check of the return value from mmap(). Thanks to + Corydon for catching this one. ........ + + * apps/app_directory.c: Fix app_directory to actually compile with + ODBC_STORAGE, and update the code to the latest res_odbc API. + + * apps/Makefile: Fix app_directory when ODBC_STORAGE is being used. + The Makefile did not properly ensure that this information got + copied from what was selected for app_voicemail. (issue #9224) + + * channels/chan_sip.c: Fix the check that ensures that the CHANNEL + function's first argument is "rtpqos". Thanks, Corydon. :) + +2007-03-27 18:16 +0000 [r59261] Steve Murphy + + * pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes + asterisk), kpfleming pointed on asterisk-dev, that DECLINE in + this case the proper thing to do. This change now has it doing + the proper thing. + +2007-03-27 18:05 +0000 [r59256-59259] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | + 4 lines Fix the use of the "sourceaddress" option when "bindaddr" + is set to 0.0.0.0 instead of having each interface explicitly + listed. (issue #7874, patch by stevens) ........ + + * channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS + function to just be additional parameter of the CHANNEL function. + This way, it will be possible for other RTP based channel drivers + to expose this information in the future. + +2007-03-27 15:00 +0000 [r59254] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 + Mär 2007) | 1 line fixed #9355 ........ + +2007-03-26 21:45 +0000 [r59230] Tilghman Lesher + + * channels/chan_sip.c: Oops, this should be case insensitive + +2007-03-26 21:41 +0000 [r59228] Steve Murphy + + * pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes + asterisk). I turned a duplicate context from a WARNING to an + ERROR. Now you get a module load failure, and asterisk just + exits. That's better than a crash, right\? + +2007-03-26 21:37 +0000 [r59227] Tilghman Lesher + + * channels/chan_sip.c: Change this to a single dp function to make + oej happy. + +2007-03-26 20:06 +0000 [r59225] Steve Murphy + + * main/config.c: Fix for 9257; by eliminating the globals in + main/config.c, we make it thread-safe, which is a minimum + requirement. + +2007-03-26 19:34 +0000 [r59223] Joshua Colp + + * apps/app_speech_utils.c: Add ability to specify no timeout. This + means as soon as the prompt is done playing it moves on to the + next priority. + +2007-03-26 18:33 +0000 [r59215-59217] Russell Bryant + + * apps/app_voicemail.c: Somehow the code for building the email for + voicemail got out of sync. This change makes a few tweaks to get + 1.4 in sync with trunk. (issue #9301) + + * apps/app_meetme.c: Fix some codec negotiation problems when + CallerID support is not enabled in SLA. (issue #9308, reported by + twilson) + +2007-03-26 18:13 +0000 [r59213] Joshua Colp + + * apps/app_speech_utils.c: Make SpeechBackground obey the digit + timeout value. + +2007-03-26 17:53 +0000 [r59207-59209] Russell Bryant + + * channels/chan_sip.c: Rename the new dialplan functions to match + the variable name + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The + AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in + some because they get set in sip_hangup. So, there are common + situations where the variables will not be available in the + dialplan at all. So, this patch provides an alternate method for + getting to this information by introducing AUDIORTPQOS and + VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, + with some testing by blitzrage) + +2007-03-26 17:38 +0000 [r59206] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, + pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE, + and STANDALONE_AEL + +2007-03-26 15:25 +0000 [r59202] Nadi Sarrar + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure, + include/asterisk/autoconfig.h.in, channels/misdn/Makefile, + channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2 + provides a dsp pipeline for i.e. echo cancellation modules, make + chan_misdn use it. * add a check for linux/mISDNdsp.h to + configure.ac and update the autogenerated files: 'configure', + 'autoconfig.h.in' (the 'configure' script was not in sync with + the latest configure.ac, so the diff is a bit bigger than + expected). + +2007-03-26 15:16 +0000 [r59200] Joshua Colp + + * pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the + aelparse binary! DONT_OPTIMIZE should now work once again. + +2007-03-24 01:39 +0000 [r59195] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 + lines Only try to handle a response if it has a response code. + (ASA-2007-011) ........ + +2007-03-23 16:11 +0000 [r59188-59189] Steve Murphy + + * /: blocking out the fix in 59187... already incorporated here + + * /, apps/app_macro.c: Merged revisions 59186 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1 + line Added a few words in the Macro doc strings about the + behavior of macros with hangups (et al.), as per 9337 ........ + +2007-03-22 23:40 +0000 [r59180-59182] Kevin P. Fleming + + * channels/chan_sip.c: don't allow string input to overrun the + buffer to hold it (ASA-2007-010) + + * channels/chan_misdn.c: remove variables that are no longer used + (--enable-dev-mode is good, developers should be using it) + +2007-03-22 14:40 +0000 [r59145] Steve Murphy + + * utils/Makefile: The stuff in utils was compiling with -O6 even if + DONT_OPTIMIZE is set in menuconfig. Added the include to fix that + +2007-03-21 18:08 +0000 [r59081-59089] Joshua Colp + + * main/http.c: Add svg mimetype for pari. + + * res/res_monitor.c, /: Merged revisions 59086 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2 + lines Indicate the filename changed when it is changed. (issue + #9311 reported by jsmith) ........ + + * channels/chan_sip.c: Until we can do media level parsing for + sendrecv/etc just use the first value found. This crept up when a + phone was offered audio+video and returned an inactive video + stream. chan_sip thought the phone said to put the person on hold + but that was totally wrong. (issue #9319 reported by benbrown) + +2007-03-20 21:04 +0000 [r59078] Tilghman Lesher + + * main/logger.c: Fix defines for inline stack backtraces (only used + by developers anyway) + +2007-03-20 20:42 +0000 [r59076] Joshua Colp + + * channels/iax2-parser.c: Copy len variable as well, should fix + remaining IAX2 DTMF issues. + +2007-03-20 17:48 +0000 [r59069-59070] Steve Murphy + + * apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should + return it to its previous, untouched, state. + + * apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h: + The fix for the AEL <> (bug 9316) is here... + +2007-03-20 13:16 +0000 [r59064] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, + channels/misdn/chan_misdn_config.h: Merged revisions + 58849-58850,59062-59063 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | + 1 line added method standard_dec for dialing out on groups, to + avoid conflicts, which caused issues with some ISDN providers + ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 + Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 | + crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line + avoid sending a disconnect when we already received one. ........ + r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | + 1 line modified a loglevel ........ + +2007-03-19 Jason Parker + + * Asterisk 1.4.2 released. + +2007-03-19 22:29 +0000 [r59049] Tilghman Lesher + + * funcs/func_strings.c: Oops, this should have been a %d all along + +2007-03-19 15:52 +0000 [r59042] Joshua Colp + + * funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295 + reported by ajohnson) + +2007-03-19 15:42 +0000 [r59040] Tilghman Lesher + + * configs/sip_notify.conf.sample: Fix unescaped semicolon (reported + via -dev list) + +2007-03-18 20:37 +0000 [r59037] Olle Johansson + + * channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return + code 0 (reported by qwerty1979) + +2007-03-18 16:36 +0000 [r59035] BJ Weschke + + * apps/app_followme.c: Don't return a non-zero return code if the + profile doesn't exist, to match what the documentation says it + already does. (#9307 Reported by kkiely) + +2007-03-16 16:12 +0000 [r58992] Joshua Colp + + * apps/app_page.c: Wait for the async thread to exit when hanging + up all of the paged phones under all circumstances. (issue #9181 + reported by PhilSmith) + +2007-03-16 01:42 +0000 [r58947-58957] Russell Bryant + + * configs/sla.conf.sample: fix a couple SLA documentation + references + + * doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex + (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added), + doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added), + doc/channelvariables.txt (added), doc/ael.txt (added), + doc/billing.tex (removed), build_tools/prep_tarball, + doc/callingpres.txt (added), doc/enum.txt (added), + doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added), + doc/cdrdriver.tex (removed), build_tools/make_buildopts_h, + doc/security.txt (added), doc/imapstorage.txt (added), + doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed), + doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac, + doc/iax.txt (added), doc/ael.tex (removed), + doc/channelvariables.tex (removed), doc/enum.tex (removed), + doc/security.tex (removed), doc/math.txt (added), Makefile, + doc/imapstorage.tex (removed), doc/privacy.tex (removed), + doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt + (added), apps/app_voicemail.c, doc/cliprompt.txt (added), + doc/chaniax.txt (added), doc/app-sms.txt (added), + doc/ast_appdocs.tex (removed), doc/realtime.tex (removed), + doc/ices.txt (added), doc/dundi.tex (removed), + doc/linkedlists.txt (added), doc/queuelog.txt (added), + doc/extconfig.txt (added), doc/radius.txt (added), + doc/cliprompt.tex (removed), doc/chaniax.tex (removed), + doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex + (removed), doc/ices.tex (removed), doc/asterisk.tex (removed), + doc/queuelog.tex (removed), doc/configuration.txt (added), + doc/asterisk-conf.txt (added), doc/sla.pdf (added), + doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt + (added), doc/mp3.tex (removed), doc/configuration.tex (removed), + doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added), + doc/channels.txt (added), doc/ip-tos.tex (removed), + doc/extensions.txt (added), doc/queues-with-callback-members.txt + (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added), + doc/misdn.txt (added), doc/manager.txt (added), + doc/jitterbuffer.tex (removed), doc/extensions.tex (removed), + doc/billing.txt (added), doc/localchannel.txt (added), + doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt + (added), doc/00README.1st (added): Making these documentation + changes in the 1.4 branch upset various people, so these chanes + will only be done in the trunk. + + * build_tools/prep_tarball: Add the --pdf option to the usage of + rubber in prep_tarball + + * Makefile, build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add + configure script checking for GTK2 and some additional Makefile + targets to support gmenuselect + +2007-03-15 23:52 +0000 [r58946] Tilghman Lesher + + * main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match + common syntax and update the resulting appdocs TeX file + +2007-03-15 23:24 +0000 [r58941] Russell Bryant + + * doc/asterisk.tex: add a link to the rubber homepage + +2007-03-15 23:11 +0000 [r58939] Tilghman Lesher + + * apps/app_setcdruserfield.c, main/pbx.c, + apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c: + Expand deprecation warnings from simply warning on use to the + builtin documentation. + +2007-03-15 22:51 +0000 [r58935-58937] Russell Bryant + + * doc/asterisk.tex, Makefile: Add Asterisk version information to + the generated PDF + + * build_tools/prep_tarball: have prep_tarball attempt to build + asterisk.pdf + +2007-03-15 22:32 +0000 [r58933] Tilghman Lesher + + * funcs/func_realtime.c: Function works fine, but the documentation + is backwards. + +2007-03-15 22:25 +0000 [r58931] Russell Bryant + + * doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex + (added), doc/freetds.txt (removed), doc/odbcstorage.txt + (removed), configure, doc/sla.tex, doc/cygwin.txt (removed), + doc/model.txt (removed), doc/channelvariables.txt (removed), + doc/ael.txt (removed), doc/billing.tex (added), + doc/callingpres.txt (removed), doc/enum.txt (removed), + doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed), + doc/cdrdriver.tex (added), build_tools/make_buildopts_h, + doc/security.txt (removed), doc/imapstorage.txt (removed), + doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added), + doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac, + doc/iax.txt (removed), doc/ael.tex (added), + doc/channelvariables.tex (added), doc/enum.tex (added), + doc/security.tex (added), doc/math.txt (removed), Makefile, + doc/imapstorage.tex (added), doc/privacy.tex (added), + doc/realtime.txt (removed), doc/dundi.txt (removed), + doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt + (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed), + doc/ast_appdocs.tex (added), doc/realtime.tex (added), + doc/ices.txt (removed), doc/dundi.tex (added), + doc/linkedlists.txt (removed), doc/queuelog.txt (removed), + doc/extconfig.txt (removed), doc/radius.txt (removed), + doc/cliprompt.tex (added), doc/chaniax.tex (added), + doc/hardware.txt (removed), doc/mp3.txt (removed), + doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex + (added), doc/queuelog.tex (added), doc/configuration.txt + (removed), doc/asterisk-conf.txt (removed), doc/sla.pdf + (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added), + doc/h323.txt (removed), doc/mp3.tex (added), + doc/configuration.tex (added), doc/asterisk-conf.tex (added), + doc/jitterbuffer.txt (removed), doc/channels.txt (removed), + doc/ip-tos.tex (added), doc/extensions.txt (removed), + doc/queues-with-callback-members.txt (removed), doc/apps.txt + (removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt + (removed), doc/manager.txt (removed), doc/jitterbuffer.tex + (added), doc/extensions.tex (added), doc/billing.txt (removed), + doc/localchannel.txt (removed), + doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt + (removed), doc/00README.1st (removed): Merge changes from + svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc + directory into a single LaTeX formatted document so that we can + generate a PDF, HTML, or other formats from this information. * + Add a CLI command to dump the application documentation into + LaTeX format which will only be include if the configure script + is run with --enable-dev-mode. * The PDF turned out to be close + to 1 MB, so it is not included. However, you can simply run "make + asterisk.pdf" to generate it yourself. We may include it in + release tarballs or have automatically generated ones on the web + site, but that has yet to be decided. + +2007-03-15 18:13 +0000 [r58923] Joshua Colp + + * channels/chan_iax2.c: Don't assume that the pvt structure will + still exist after calling schedule_delivery as it may not. (issue + #9278 reported by fmachado) + +2007-03-14 19:18 +0000 [r58894-58906] Russell Bryant + + * channels/chan_sip.c: Some people like to put "limitonpeer" + instead of "limitonpeers" in their configuration. While we're at + it, support "limitonpeerz" and "limitonpeerssssss". (inspired by + issue #9172) + + * doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the + examples section + + * doc/security.txt, /: Merged revisions 58896 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | + 3 lines Add a note to the security file that the Asterisk CLI and + log files may contain sensitive information, and that people + should keep this in mind. ........ + + * configs/sla.conf.sample, apps/app_meetme.c: By default, don't + attempt to do any CallerID handling at all with SLA because it is + known to not work properly in some situations. However, add an + option to enable it for those that would like to use it anyway. + The short story behind this is that to properly handle CallerID + with SLA, we need the ability to change the CallerID on an + existing call, and we are not ready to handle that. + +2007-03-14 01:47 +0000 [r58880] Tilghman Lesher + + * funcs/func_strings.c: Issue 9162 - + pbx_substitute_variables_helper assumes the buffer is initialized + to all zeroes. This fixes a case where it wasn't. + +2007-03-13 23:19 +0000 [r58870-58872] Russell Bryant + + * apps/app_meetme.c: Ensure that the blinky lights show that the + trunk stopped ringing when the trunk hangs up before a station + has answered it. (issue #9234, reported by francesco_r) + + * configs/sla.conf.sample: fix the reference to the SLA + documentation + +2007-03-13 11:49 +0000 [r58843-58848] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 + lines Issue #9229 - No port in request URI on register to non + default SIP ports (neelakantan) ........ + + * channels/chan_sip.c: Don't hangup the call on OK or errors on + MESSAGE and INFO inside of a dialog (like video update requests). + + * channels/chan_sip.c: Issue #9251 - Clear From URI from user + attributes (tgrman) + +2007-03-12 16:52 +0000 [r58833] Joshua Colp + + * /: Blocked revisions 58832 via svnmerge ........ r58832 | file | + 2007-03-12 12:49:49 -0400 (Mon, 12 Mar 2007) | 2 lines We can't + use the assembler version of fetchadd_int under Intel Macs. + (issue #9254 reported by darrell budic) ........ + +2007-03-12 13:08 +0000 [r58825-58826] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 57034,57523,57753,58558 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | + 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com + bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02 + 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........ + r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | + 1 line fixed another place where the out_cause was hardcoded to + 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 + Mar 2007) | 1 line we can free channel 31 as well, since we can + occupy it ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, channels/misdn/ie.c, + channels/misdn/isdn_msg_parser.c: added UU transceiving and + corect handling for rdnis + +2007-03-12 01:21 +0000 [r58779-58783] Joshua Colp + + * main/rtp.c: Allow RFC2833 compensation to compensate for even + stupider implementations by queueing up the end frame at the + start, not the actual end. (issue #8963 reported by AndrewZ) + + * channels/chan_sip.c, configs/sip.conf.sample: Add + matchexterniplocally setting which only substitutes your + externip/externhost setting if it matches the localnet setting. I + know of at least two people who need opposite settings, so I made + it an option! (issue #8821 reported by kokoskarokoska) + +2007-03-10 18:11 +0000 [r58638-58705] Russell Bryant + + * channels/chan_iax2.c: Fix a few more places in chan_iax2 where + the ast_frame used for receiving a frame was not properly + initialized. - Interpolating a frame when the jitterbuffer is in + use - decrypting a frame when IAX2 encryption is on - frames in + an IAX2 trunk + + * apps/app_meetme.c: Make the compiler happy and initialize a + variable. + + * doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added): + Merge some updates to the SLA documentation. I plan to keep + working on this to explain all of the expected behavior with call + handling, configuration details for specific phones, and other + things. However, I got tired of doing it in plain text, so I + switched to using LaTeX. I have included the PDF version. I + haven't been able to get a nice looking plain text version out of + it yet, but I'm not terribly concerned since this is supposed to + be more of the manual, while the plain text sample configuration + file is the reference. + +2007-03-09 21:08 +0000 [r58584-58604] Joshua Colp + + * apps/app_voicemail.c: Fix spelling of unavailable in voicemail + documentation. (issue #9248 reported by tensai) + + * /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 + lines If we are unable to lookup the host in a c line we have to + abort, otherwise the previous data is gone and we will + (potentially) have no data when all is said and done. ........ + +2007-03-08 22:15 +0000 [r58510-58512] Russell Bryant + + * apps/app_meetme.c: Hang up the channel that put the call on hold + in the event processing thread to avoid a race condition. Also, + if the station originated the call that it is putting on hold, + don't hang up the trunk if it was the only station on the call + and it is hanging up due to hold and not a normal hangup. + + * channels/chan_zap.c: Add a missing break statement so that + handling the above event does not incorrectly destroy the + channel. (issue #9242, andrew) + +2007-03-08 21:33 +0000 [r58479] Tilghman Lesher + + * res/res_odbc.c: Fix segfault (Issue 9236) + +2007-03-08 20:54 +0000 [r58474] Russell Bryant + + * apps/app_meetme.c: Refactor hold handling a bit so that it does + not require keeping the call up when a call is put on hold. + +2007-03-08 18:01 +0000 [r58389-58436] Joshua Colp + + * main/rtp.c: Make early SDP seeding even smarter! We have to check + codecs in the make_compatible function too. (issue #9221 reported + by marcelbarbulescu) + + * main/dsp.c, /: Merged revisions 58388 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 + lines Only print out debug message if the definition that makes + the variables shows up was actually defined. (issue #9233 + reported by serginuez) ........ + +2007-03-08 13:23 +0000 [r58351-58354] Kevin P. Fleming + + * main/http.c: this change was not needed; fclose() handles closing + the file descriptor already + + * apps/app_meetme.c: fix a compiler warning, and overwriting 'res' + value + + * main/http.c: fix two cases where HTTP session file descriptors + would not be closed + +2007-03-08 01:01 +0000 [r58243-58320] Russell Bryant + + * channels/chan_zap.c, configure, configure.ac: If we receive + ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, + tzafrir) Also, update the configure script to make sure that we + don't try to build chan_zap if the installed version of zaptel + does not include ZT_EVENT_REMOVED. + + * /, channels/chan_iax2.c: (This bug was reported to me by Kinsey + Moore) Merged revisions 58242 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | + 7 lines Fix a problem where the Asterisk channel name could be + that of the wrong IAX2 user for a call. This is because the first + step of choosing this name is to look for an IAX2 peer that + happens to have the same IP/port number that this call is coming + from and assuming that is it. However, this is not always + correct. So, I have made it change this name after authentication + happens since at that point, we have an exact match. ........ + +2007-03-07 17:52 +0000 [r58240] Joshua Colp + + * main/rtp.c, channels/chan_sip.c: Ensure we have (or should have) + at least one matching codec before attempting early bridge SDP + seeding. (issue #9221 reported by marcelbarbulescu) + +2007-03-07 00:27 +0000 [r58165-58168] Russell Bryant + + * /: Blocked revisions 58167 via svnmerge ........ r58167 | russell + | 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines Fix a + misplaced block of code in the 1.2 version of the patch to fix + issue #8977 ........ + + * main/manager.c, /: Merged revisions 58164 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | + 4 lines If the channels acquired using the manager Redirect + action are not up, then don't attempt to do anything with them. + It could lead to weird behavior, including crashes. (issue #8977) + ........ + +2007-03-06 23:10 +0000 [r58121] Steve Murphy + + * /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 + line Fix for 9220: Eyebeam cannot renew subscriptions for + presence info. Reason: re-SUBSCRIBE requests don't include Accept + headers, which the rfc says are optional (to put it tersely), (it + uses MAY), and luckily, the sip_pvt struct has the format info + stored, so we simply leave it if the format is set, and the + accept header null. ........ + +2007-03-06 23:00 +0000 [r58119] Russell Bryant + + * configs/voicemail.conf.sample: Clarify the documentation of the + dialout and sendvoicemail options. (issue #9000, caio1982 and + serge-v) + +2007-03-06 20:37 +0000 [r58053] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 + lines Change error message to proper message ........ + +2007-03-06 18:01 +0000 [r58023] Russell Bryant + + * channels/chan_skinny.c: Return an error of transmit_response is + called without a session. (issue #9002) + +2007-03-05 19:19 +0000 [r57870-57914] Joshua Colp + + * channels/chan_iax2.c: Since chan_iax2 does not support reception + of DTMF with duration ensure that it is set to 0 on the frame. + (issue #8521 reported by gdhgdh) + + * apps/app_meetme.c: Don't create a listen channel and record the + conference unless the option is turned on. (issue #9204 reported + by francesco_r) + + * apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 + lines Make create_dirpath use our standard for return values. -1 + is failure, 0 is success. (issue #9205 reported by ballares) + ........ + +2007-03-05 15:20 +0000 [r57826] Steve Murphy + + * main/pbx.c, /: Merged revisions 57825 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 + line Fixed a typo introduced via 9156 (either the gotos or their + doc strings are wrong) ........ + +2007-03-05 04:19 +0000 [r57768-57798] Joshua Colp + + * main/slinfactory.c: Don't allow a NULL pointer to reach + ast_frdup. (issue #9155 reported by cmaj) + + * res/res_jabber.c: Don't reference a potentially NULL pointer. + (issue #9199 reported by klolik) + + * main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198 + reported by edgreenberg) + +2007-03-03 15:31 +0000 [r57707] Steve Murphy + + * pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2, + pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7: + Updated the regression tests + +2007-03-03 06:45 +0000 [r57649] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) + | 2 lines Memory leak of a list, if call recording was abandoned + ........ + +2007-03-03 00:59 +0000 [r57620] Dwayne M. Hubbard + + * main/say.c: submitted patch for Georgian language, issue 9010, + submitted by Alexander Shaduri + +2007-03-03 00:02 +0000 [r57591] Russell Bryant + + * configs/sla.conf.sample: add missing configuration template. + Thanks to Lacy Moore on asterisk-users for pointing this out\! + +2007-03-02 Russell Bryant + + * Asterisk 1.4.1 released. + +2007-03-02 23:03 +0000 [r57556] Russell Bryant + + * configure, configure.ac: Update the check that is used to + determine whether zaptel transcoder support is present. The + interface has changed. + +2007-03-02 17:06 +0000 [r57477] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 + lines If a SIP message comes in and goes to a method handler that + requires additional values that may not be present then send back + an error. ........ + +2007-03-02 16:55 +0000 [r57426-57473] Steve Murphy + + * main/pbx.c, /: Merged revisions 57458 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 + line further refinement in wording of goto documentation, as per + 9156, goto not proceeding to next instruction ........ + + * pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes + right, but 9184 points out the problem-- the escape is removed by + pbx_config, and pbx_ael should also, before sending it down into + the pbx engine. Also, you have to insert it back in, if you are + generating extensions.conf code from the AEL. + +2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant + + * main/file.c: Return the correct digit that interrupted the + stream. This fixes exiting the Background application when using + the m option. (issue #9176, mjagdis) + + * configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt, + include/asterisk/channel.h: Merge changes from + svn/asterisk/team/russell/sla_updates * Originally, I put in the + documentation that only Zap interfaces would be supported on the + trunk side. However, after a discussion with Qwell, we came up + with a way to make IP trunks work as well, using some things + already in Asterisk. So, here it is, this now officially supports + IP trunks. * Update the SLA documentation to reflect how to setup + IP trunks. * Add a section in sla.txt that describes how to set + up an SLA system with voicemail. * Simplify the way DTMF + passthrough is handled in MeetMe. * Fix a bug that exposed itself + when using a Local channel on the trunk side in SLA. The + station's channel needs to be passed to the dial API when dialing + the trunk. * Change a WARNING message to DEBUG in channel.h. This + message is of no use to users. + +2007-03-01 22:21 +0000 [r57318] Joshua Colp + + * channels/chan_local.c, /: Merged revisions 57317 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar + 2007) | 2 lines Don't even attempt to optimize things when a + proxy channel is involved. It will just explode in weird and + unexplaineable ways. (issue #9175 reported by + clegall_proformatique) ........ + +2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development + + * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick. + +2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant + + * configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla + docs + + * configs/sla.conf.sample, apps/app_meetme.c: Merge more changes + from svn/asterisk/team/russell/sla_updates * Add support for + private hold. By setting "hold=private" for a trunk, only the + station that put the call on hold will be able to retrieve it + from hold. Also, by setting "hold=private" for a station, any + call that station puts on hold can only be retrieved by that + station. + + * apps/app_meetme.c: Minor formatting change + + * configs/sla.conf.sample, apps/app_meetme.c: Merge changes from + svn/asterisk/team/russell/sla_updates * Add support for the + "barge=no" option for trunks. If this option is set, then + stations will not be able to join in on a call that is on + progress on this trunk. + +2007-02-28 19:23 +0000 [r57139] Steve Murphy + + * main/pbx.c, /: Merged revisions 57118 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 + line a small documentation update, to reflect reality in the goto + doc strings, as per 9156, Goto does not proceed to next prio if + jump fails ........ + +2007-02-28 18:57 +0000 [r57093] Joshua Colp + + * /, channels/chan_agent.c: Merged revisions 57092 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb + 2007) | 2 lines Fix a few more issues with the agent logoff CLI + command. (issue #9123 reported by arbrandes) ........ + +2007-02-28 18:20 +0000 [r57089] Russell Bryant + + * configs/sla.conf.sample, apps/app_meetme.c: Merge current set of + changes from svn/asterisk/team/russell/sla_updates * Add support + for station ring delays. Ring delays can be set globally for a + station or for specific trunks on the station. * Fix a few bugs + in existing code. * Restructure and Reorganize code to improve + readability and maintainability. * Improve formatting of the "sla + show (trunks|stations)" CLI commands. + +2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp + + * apps/app_meetme.c: Picky compiler... + + * apps/app_speech_utils.c: Better handle timeouts when the + individual speaks after everything has been played but before the + timeout ends. + +2007-02-28 17:15 +0000 [r57049] Steve Murphy + + * pbx/pbx_ael.c: I was surprised that I had not yet downgraded + missing goto targets and macro call defs to a warning, in case + they are in extensions.conf; I rectified this problem. Also, A + goto in a macro to a target in a catch block was not being found; + I fixed this too; the cause was that I needed to treat catch + statements like an extension in the find_match code. + +2007-02-27 17:36 +0000 [r56975] Russell Bryant + + * apps/app_voicemail.c: Fix voicemail email attachments. I missed + the conversion of one of the line endings and there was an extra + one where it should not have been. (issue #9128) + +2007-02-26 22:01 +0000 [r56922] Tilghman Lesher + + * apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky, + picky... show deprecation warning in application help, too + (reported via list) + +2007-02-26 20:42 +0000 [r56888] Russell Bryant + + * channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where + if a device was not specified in alsa.conf, then we just use the + system default, instead of creating our own default of hw:0,0. + (issue #9139) + +2007-02-26 20:07 +0000 [r56856] Joshua Colp + + * /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 + lines Obey the clearglobalvars option in extensions reload (or + dialplan reload depending on your version). (issue #9146 reported + by ramonpeek) ........ + +2007-02-26 20:04 +0000 [r56847] Russell Bryant + + * channels/chan_iax2.c: Fix a crash in my last change to + iax2_indicate(). (issue #9150) + +2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp + + * apps/app_record.c: Update app_record documentation to use new CLI + command, core show file formats. (issue #9151 reported by junky) + + * main/pbx.c: Use ast_strlen_zero to see if the language and/or + context argument is not present for Background instead of just + checking if it is NULL. (issue #9141 reported by mjagdis) + +2007-02-26 16:51 +0000 [r56785] Russell Bryant + + * channels/chan_iax2.c: Do more complete locking of the + chan_iax2_pvt struct in the indicate callback. (Problem brought + up by Ben Smithurst on the asterisk-dev list) + +2007-02-26 16:36 +0000 [r56783] Joshua Colp + + * main/asterisk.c: Allow both of the show version files and core + show file versions CLI commands to work. (issue #9135 reported by + mvanbaak) + +2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant + + * apps/app_meetme.c: Move a comment to be in the correct struct. + + * /: Blocked revisions 56729 via svnmerge ........ r56729 | russell + | 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines Ensure + that lock.h is included in utils.c with AST_API_MODULE defined so + that the implementations will be properly included when the + AST_INLINE_API functions are not going to be inlined. (issue + #9124, festr) ........ + +2007-02-25 14:46 +0000 [r56685] Tilghman Lesher + + * main/channel.c, /: Merged revisions 56684 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) + | 3 lines Issue 9130 - If prev is the last item on the channel + list, then evaluating additional conditions (e.g. name prefix) + will cause a NULL dereference. ........ + +2007-02-24 02:02 +0000 [r56569] Jason Parker + + * channels/chan_skinny.c: Make sure to set a speeddials parent on + creation. Don't crash if hold is pressed when no call is active. + Don't return in places that we shouldn't.. + +2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming + + * codecs/codec_zap.c: update to match zaptel 1.4 API change that + was committed a few minutes ago + +2007-02-23 23:24 +0000 [r56505] Russell Bryant + + * main/asterisk.c, /: Merged revisions 56504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | + 8 lines Fix up a couple more signal handlers to not do bad things + that could cause various undesirable results. The other day, I + made Asterisk deadlock by hitting Control-C because of a bad + signal handler. Now, signal handlers just set a flag and write to + an alert pipe for the flag to be handled. Then, there is another + thread that is monitoring for these flags. If being run in + console mode, it is just the main thread. If Asterisk is in the + background, a thread is created to do it. ........ + +2007-02-23 21:53 +0000 [r56457] Joshua Colp + + * main/sched.c: Change log notice to debug. It is possible for a + scheduled item to execute and be deleted at close to the same + time and unavoidable. If this happens this message creeps up. + +2007-02-23 20:20 +0000 [r56407] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | + 4 lines Don't destroy mutexes before unregistering all of the + entry points from the core. Also, fix a potential memory leak + from not destroying the locks for all of the possible call + numbers (about 32k of them). ........ + +2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming + + * build_tools/make_version_h: build special version strings for + AADK/S800i builds + +2007-02-23 17:58 +0000 [r56341] Russell Bryant + + * apps/app_voicemail.c: The IMAP storage code uses the same code to + build the email that is used when voicemail is sent via email + using something like sendmail. In the patch from bug 8033 to fix + various IMAP storage problems, the line endings in the email file + were changed in the code from "\n" to "\r\n". However, this + breaks sending regular voicemail to email. So, this change + conditionally sets line endings to "\r\n" only if IMAP_STORAGE is + enabled. (issue #9128, patch by jarjarbinks, modified by me to + not break IMAP storage) + +2007-02-22 23:25 +0000 [r56280] Joshua Colp + + * /: Blocked revisions 56279 via svnmerge ........ r56279 | file | + 2007-02-22 18:19:25 -0500 (Thu, 22 Feb 2007) | 2 lines Always + defer Agent logoff if any channels are up until they hang up. + (issue #9123 reported by arbrandes) ........ + +2007-02-22 23:08 +0000 [r56277] Russell Bryant + + * configs/sla.conf.sample, main/dial.c, apps/app_meetme.c, + doc/sla.txt: Merge changes from team/russell/sla_updates. This + batch of changes to the SLA code does a few different things. * I + made the SLA code event driven instead of having to act in a lot + of busy loops while dialing things to wait for state changes. + This makes the code more efficient and readable at the same time. + * I have implemented a couple of new features. The first is + inbound trunk ringing timeouts. This is an option that defines + how long to let an incoming call on a trunk to ring. * I have + also implemented ring timeouts for stations. They may be + specified for the entire station, meaning it is how long to let + the station ring before giving up. You can also specify a ring + timeout for a specific trunk on a station. So, you can say that + you only want a specific station to ring 5 seconds if it is line1 + ringing, but otherwise, there is no timeout. + +2007-02-22 18:49 +0000 [r56231] Joshua Colp + + * main/channel.c, /, channels/chan_sip.c: Merged revisions 56230 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 + lines Only change the original or clone channel if it's the + channel behind the proxy channel, not if it's just a regular + bridged channel. ........ + +2007-02-22 14:06 +0000 [r56169] TransNexus OSP Development + + * doc/osp.txt: Update OSP documentation for v1.4. + +2007-02-22 10:33 +0000 [r56125] Olle Johansson + + * channels/chan_sip.c: Move message from verbose to debug + +2007-02-22 02:39 +0000 [r56094] Steve Murphy + + * sounds/Makefile: updated the sound tarball versions in Makefile + +2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant + + * channels/chan_sip.c: Restructure a little bit of code to reduce + nesting. There is no functionality change here. + + * /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | + 3 lines If we receive a frame that is not in any of the + negotiated formats, then drop it. (potentially issue #8781 and + SPD-12) ........ + +2007-02-22 00:35 +0000 [r56008] Joshua Colp + + * main/cli.c: Print out deprecation notice on usage output of CLI + commands. (issue #8925 reported by blitzrage) + +2007-02-22 00:08 +0000 [r56006] Kevin P. Fleming + + * main/loader.c: disable unloading of embedded modules... there is + a fundamental problem with doing so that will not be fixed in + this version of Asterisk due to its invasiveness + +2007-02-21 20:35 +0000 [r55957] Joshua Colp + + * /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 + lines Change naughty warning message to provide useful + information. If a write now fails on a channel in meetme it will + tell you the channel name instead of spitting out the wrong error + message. ........ + +2007-02-21 20:27 +0000 [r55954] Jason Parker + + * channels/chan_gtalk.c: Fix locking issue, and accept + "transport-accept" as a valid accept message. This should solve + issues 8970 and 8503. + +2007-02-21 20:22 +0000 [r55951] Russell Bryant + + * apps/app_meetme.c: Simplify the last change to app_meetme, and + move the call to dispose_conf() up into the block where we know a + conf exists. + +2007-02-21 20:16 +0000 [r55914-55949] Joshua Colp + + * apps/app_meetme.c: Only dispose of the conference if one was + created. + + * apps/app_speech_utils.c: Only start playing the next file if we + have not been quieted. + + * channels/chan_sip.c: Add a flag that indicates whether a SIP + dialog is an outgoing call or not. SIP_OUTGOING originally did it + but it was repurposed to the direction of the last transaction, + which can cause update_call_counter to falsely decrease the wrong + counters. (please don't hurt me oej) (issue #8943 reported by + mdu113) + +2007-02-21 14:06 +0000 [r55869] Kevin P. Fleming + + * /, build_tools/make_version: Merged revisions 55868 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21 + Feb 2007) | 2 lines use new tag version script ........ + +2007-02-21 08:32 +0000 [r55834] Olle Johansson + + * channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly + after transfer (decrement inuse early on transferer's call leg) + +2007-02-21 02:01 +0000 [r55799] Jason Parker + + * channels/chan_gtalk.c: Fix segfault when buddy couldn't be found. + Issue 7764, patch by sailer + +2007-02-21 01:03 +0000 [r55751-55758] Russell Bryant + + * apps/app_meetme.c: Improve the reference counting to fix bugs + where people report seeing conferences listed that have no + members. (issue #9073) + + * /: Blocked revisions 55750 via svnmerge ........ r55750 | russell + | 2007-02-20 18:19:14 -0600 (Tue, 20 Feb 2007) | 9 lines Fix + random crashes when using the MeetMe application. This patch + converts list handling to use the linked list macros and most + importantly, implements reference counting on the ast_conference + objects. The reference counting was first backported from 1.4. + However, that code has some problems that caused the reference + count to never hit zero. Those problems are fixed in this patch + and will be resolved in 1.4 and trunk next, with a different + patch. (issues #7647, #9073, #9106, BE-115). ........ + +2007-02-21 00:11 +0000 [r55670-55741] Joshua Colp + + * apps/app_voicemail.c: Better handle dropped IMAP connections. + (issue #9054 reported by bsmithurst) + + * channels/chan_sip.c: Return behavior I removed. I did not + remember that you could just add a localnet entry to make it + work. + + * channels/chan_sip.c: Don't test our own address against the + localnet settings. At least one person has had issues as a result + of this from #7051 so I'm reversing it. (issue #8821 reported by + kokoskarokoska) + + * /, channels/chan_agent.c: Merged revisions 55669 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb + 2007) | 2 lines Defer clearing callback information if channels + are up until they are hung up. This ensures the hangup process + goes smoothly and no channels get hung in limbo. (issue #8088 + reported by kebl0155) ........ + +2007-02-20 20:26 +0000 [r55589-55634] Russell Bryant + + * main/http.c: Add the Asterisk version information to the Server + header in HTTP responses. (requested by Pari) + + * include/asterisk/manager.h: Increase the maximum number of + manager headers to 128, at the request of Pari. + + * /: Blocked revisions 55588 via svnmerge ........ r55588 | russell + | 2007-02-20 13:49:50 -0600 (Tue, 20 Feb 2007) | 3 lines Convert + a tab to spaces so that the documentation is printed out properly + aligned. ........ + +2007-02-20 16:53 +0000 [r55555] Jason Parker + + * channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free + with strdupa (thanks file) 55555! + +2007-02-20 16:41 +0000 [r55553] Russell Bryant + + * configs/sla.conf.sample: Change the formatting of sla.conf.sample + to make it more readable. (issue #9112, blitzrage) + +2007-02-19 21:12 +0000 [r55483] Olle Johansson + + * res/res_jabber.c: - Not sending arguments to an application is + not "out of memory" - Making error messages a bit more clear + +2007-02-19 18:11 +0000 [r55435] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007) + | 2 lines forcename and forcegreetings options should check to + see if the recording already exists ........ + +2007-02-19 14:52 +0000 [r55397] Doug Bailey + + * channels/chan_iax2.c: Changed iax2 process thread to detached to + correct memory leak due to left over thread context on thread + exit. Modified module unload process to avoid deadlocks on + pthread cancels + +2007-02-18 12:35 +0000 [r55250-55278] Olle Johansson + + * /, apps/app_record.c: Merged revisions 55277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 + lines Documentation update (#9053, jsmith) ........ + + * /: Block patch that was made only for 1.2 (already implemented in + 1.4 and trunk) + +2007-02-17 17:39 +0000 [r55219] Joshua Colp + + * apps/app_queue.c: Add missing membername option to AddQueueMember + documentation. (issue #9088 reported by seanbright) + +2007-02-17 17:10 +0000 [r55217] Jason Parker + + * channels/chan_skinny.c: Fix an issue where callerid would not be + displayed on some phones. Issue 8995, initial patch and research + done by wedhorn + +2007-02-17 03:55 +0000 [r55086-55154] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 55153 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 + lines Answer the channel before recording privacy information. + (issue #8926 reported by lmamane) ........ + + * apps/app_queue.c: Make the 'i' option of Queue actually work. + (issue #8986 reported by utis) + + * /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 + lines Allow chan_sip to handle attended transfers from a SIP + phone that is sitting behind chan_agent. Yes folks, all it took + was one line of code. (issue #8784 reported by pzieba) ........ + +2007-02-17 00:40 +0000 [r55006-55052] Russell Bryant + + * configure, include/asterisk/autoconfig.h.in, configure.ac: If the + pg_config application is found, but there is probably executing + it, then consider postgres unavailable. (issue #8637) + + * codecs/gsm/Makefile: Filter out yet another architecture that + does not work with the optimizations in the built-in libgsm. + (issue 8637, ovi) + + * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged + revisions 55005 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | + 9 lines Revert the change I did in revisions 54955, 54969, and + 54970, in 1.2, 1.4, and trunk. I decided that once a conference + is created from meetme.conf, it is acceptable behavior that the + pin can not be changed until the conference goes away. I also + added a note in meetme.conf to describe this behavior. We still + have another issue in 1.4 and trunk where some conferences with + no users don't go away. That is the real bug that needs to be + addressed here. ........ + +2007-02-16 22:18 +0000 [r55002] Joshua Colp + + * /, channels/chan_agent.c: Merged revisions 54999 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb + 2007) | 2 lines Do not send indications through ast_indicate in + chan_agent but instead go directly to the technology. This way + when indications are emulated they happen on the Agent channel + and do not screw up formats on the channels. (issue #8439 + reported by punkgode) ........ + +2007-02-16 21:12 +0000 [r54969] Russell Bryant + + * /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | + 5 lines For conferences that are configured in meetme.conf, check + the configuration file every time someone joins the conference + instead of only when the conference is first created. This is to + ensure that changes to the pin numbers in the config file are + always honored. (issue #9073) ........ + +2007-02-16 18:51 +0000 [r54924] Joshua Colp + + * apps/app_dial.c: Need to check macro extension as well as macro + context for directed pickup. + +2007-02-16 18:03 +0000 [r54888-54898] Russell Bryant + + * pbx/pbx_config.c: Fix setting "autofallthrough" to yes by + default. It was set to enabled in pbx.c. However, if the option + was not present in extensions.conf, then pbx_config.c would set + it back to disabled. + + * res/res_features.c: Clean up a few coding guidelines issues - + spaces to tabs, use sizeof() to pass the size of a static buffer, + add spaces ... + +2007-02-16 17:25 +0000 [r54886] Jason Parker + + * main/asterisk.c: Clarify a restart message. It's silly, but the + reporter had a very valid point. Issue 9079 + +2007-02-16 17:02 +0000 [r54884] Joshua Colp + + * apps/app_dial.c: Allow directed pickup to pick up the real + context instead of the macro context if a Macro is used. (issue + #8984 reported by jamesb63) + +2007-02-16 12:06 +0000 [r54772-54787] Olle Johansson + + * channels/chan_sip.c: Issue #7541 - Handle multipart attachments + to SIP messages - even if boundary is quoted. + + * /, res/res_agi.c: Merged revisions 54771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2 + lines Issue #9069 - If we open with TH we should not close with + /TD. (seanbright) ........ + +2007-02-16 00:48 +0000 [r54481-54714] Joshua Colp + + * apps/app_speech_utils.c: Don't let dtmf leak over into the engine + and let it skew the results... also give DTMF results priority. + (issue #9014 reported by surftek) + + * apps/app_dial.c, /: Merged revisions 54622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 + lines Use a separate variable to indicate execution should + continue instead of the return value. (issue #8842 reported by + pluto70) ........ + + * apps/app_dial.c: Forward begin DTMF frames as well as end. (issue + #9068 reported by mhardeman) + +2007-02-14 18:44 +0000 [r54439] Olle Johansson + + * /: Block patch only needed in 1.2 + +2007-02-14 16:56 +0000 [r54375] Matt Frederickson + + * channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 + lines When handling glare on a PRI, move the requested channel + rather than hang up the old one. Fix for 8957 and 9011. ........ + +2007-02-14 01:09 +0000 [r54290] Joshua Colp + + * main/channel.c: Add G722 to ast_best_codec. If anyone disagrees + with it's placement, feel free to change it. (issue #9045 + reported by gork) + +2007-02-13 21:31 +0000 [r54204-54235] Russell Bryant + + * channels/chan_sip.c: Remove a couple of leftover debug messages + + * include/asterisk/devicestate.h: Fix the documentation on the + return values from device state provider registration and + deletion. + + * channels/chan_sip.c: If we fail to create the SIP socket, then + return -1 from reload_config() so that load_module() will return + AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get + spammed with error messages every time chan_sip tries to send a + message. + +2007-02-13 18:41 +0000 [r54180] Olle Johansson + + * /: Blocking patch for 1.2 only + +2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant + + * main/dial.c, include/asterisk/dial.h: Change + ast_set_state_callback() to ast_dial_set_state_callback() + + * main/dial.c, apps/app_meetme.c, apps/app_page.c, + include/asterisk/dial.h: - Add the ability to register a callback + to monitor state changes in an asynchronous dial operation. - + Rename the various references to "status" to "state" in the dial + API + +2007-02-12 16:34 +0000 [r54026] Joshua Colp + + * configure, configure.ac: Make the --without-oss argument work. + (issue #9026 reported by puzzled) + +2007-02-12 15:38 +0000 [r54002] Russell Bryant + + * configs/users.conf.sample: Fix a typo where "vmpassword" should + be "vmsecret" + +2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach + + * channels/chan_h323.c: Fix VLDTMF reception + + * apps/app_echo.c: Much simpler than previous one ;-) + + * main/channel.c: Provide correct DTMF duration + + * main/cli.c: Bring deprecated 'debug channel ' command back + +2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming + + * configure, configure.ac, acinclude.m4: don't display the + --with-imap message unless --with-imap was specified without a + path use '-n' instead of '! -z' for tests + +2007-02-10 01:02 +0000 [r53783-53821] Russell Bryant + + * apps/app_meetme.c: Add some output for "show application + SLAStation/SLATrunk" + + * channels/chan_sip.c: Change some text to properly state "On + Hold", which was already done in trunk. + + * configs/sla.conf.sample, include/asterisk/app.h, + include/asterisk/utils.h, main/dial.c, apps/app_meetme.c, + channels/chan_sip.c, doc/sla.txt (added), + include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge + team/russell/sla_rewrite This is a completely new implementation + of the SLA functionality introduced in Asterisk 1.4. It is now + functional and ready for testing. However, I will be adding some + additional features over the next week, as well. For information + on how to set this up, see configs/sla.conf.sample and + doc/sla.txt. In addition to the changes in app_meetme.c for the + SLA implementation itself, this merge brings in various other + changes: chan_sip: - Add the ability to indicate HOLD state in + NOTIFY messages. - Queue HOLD and UNHOLD control frames even if + the channel is not bridged to another channel. linkedlists.h: - + Add support for rwlock based linked lists. dial.c: - Add the + ability to run ast_dial_start() without a reference channel to + inherit information from. + + * apps/app_echo.c: When the Echo() application receives the digit + '#', echo that back as well. Since we already sent the BEGIN + frame for that digit, it makes sense to send the END as well. + +2007-02-09 23:52 +0000 [r53779-53781] Kevin P. Fleming + + * channels/chan_gtalk.c: another dependency + + * apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c, + funcs/func_odbc.c, res/res_adsi.c: add some inter-module + dependencies + + * build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk + scripts to work when both MODULEINFO and MAKEOPTS are present in + a source file + +2007-02-09 19:33 +0000 [r53749] Joshua Colp + + * apps/app_dial.c: Temporarily change musicclass on channel to one + specified in Dial so that the 'm' option functions properly. + (issue #8969 reported by christianbee) + +2007-02-09 16:42 +0000 [r53715] Kevin P. Fleming + + * doc/imapstorage.txt, configure, configure.ac: clarify the fact + that voicemail IMAP storage cannot be built against a distro's + binary c-client library package (at least not at this time) + +2007-02-08 23:18 +0000 [r53672] Olle Johansson + + * main/acl.c: Don't output debug unless we asked for it + +2007-02-08 17:54 +0000 [r53601] Joshua Colp + + * apps/app_speech_utils.c: Fix timeout issue when utterance is + longer then timeout itself. + +2007-02-08 13:47 +0000 [r53530-53532] Tilghman Lesher + + * main/loader.c: Issue 9007 - Mutex not released on early return + + * apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007) + | 2 lines Issue 9003 - If fullname is empty, quote() passes back + "\"" ........ + +2007-02-07 23:52 +0000 [r53464-53497] Russell Bryant + + * main/db1-ast/Makefile: When building libdb1.a, put the additional + flags needed at the beginning of ASTCFLAGS, instead of at the + end. This way, we ensure that we find the local headers first + before accidentally trying to use headers that exist in locations + specified in the ASTCFLAGS passed from the main Makefile. (issue + #8637, ovi) + + * main/Makefile: The clean target actually needs to run "distclean" + on editline. This is because we need to make sure that its + configure script gets executed again, because the CFLAGS we want + to pass to editline may have changed. + +2007-02-07 17:53 +0000 [r53434] Joshua Colp + + * main/rtp.c: We can not reliably do P2P bridging with DTMF passing + back with compensation if we need to listen for DTMF frames. + (issue #8962 reported by caio1982) + +2007-02-07 17:39 +0000 [r53429] Russell Bryant + + * main/rtp.c: When parsing the NTP timestamp in a sender report + message, you are supposed to take the low 16 bits of the integer + part, and the high 16 bits of the fractional part. However, the + code here was erroneously taking the low 16 bits of the + fractional part. It then shifted the result 16 bits down, so the + result was always zero. This fix makes it grab the appropriate + high 16 bits, instead. (issue #8991, pointed out by + andre_abrantes) + +2007-02-07 17:04 +0000 [r53358-53399] Joshua Colp + + * apps/app_playback.c: Directly load say.conf in load_module + instead of calling the reload function. (issue #8946 reported by + junky) + + * /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 + lines Fix a few potential memory leaks with realtime users and + peers. (issue #8999 reported by bsmithurst) ........ + +2007-02-07 15:33 +0000 [r53355] Tilghman Lesher + + * /, apps/app_macro.c: Merged revisions 53354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007) + | 2 lines Issue 7440 - Macro called from Macro from the h + extension exits prematurely ........ + +2007-02-07 09:22 +0000 [r53324] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged + revisions 52843 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | + 1 line fixed some possible segfaults. also fixed an very + important bug which occurs on high load (when calls are very fast + generated) ........ + +2007-02-07 05:24 +0000 [r53246-53294] Tilghman Lesher + + * res/res_jabber.c: Text fix for jabber reload command (reported by + bkruse via IRC) + + * main/manager.c, /: Merged revisions 53245 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007) + | 2 lines Issue 8987 - Status could return two responses + (mnicholson) ........ + +2007-02-05 23:43 +0000 [r53222] Olle Johansson + + * channels/chan_sip.c: Formatting + +2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp + + * apps/app_playback.c: Ensure say_cfg is NULL when the module is + loaded. (issue #8946 reported by junky) + + * apps/app_playback.c: Unregister Playback CLI commands as well as + dialplan application. (issue #8946 reported by junky) + +2007-02-05 00:18 +0000 [r53143] Olle Johansson + + * channels/chan_sip.c: Add some comments on queue system behaviour + and how it affects the SIP channel + +2007-02-03 21:05 +0000 [r53138] Joshua Colp + + * channels/chan_sip.c: Make SIPDtmfMode application work with + recent capability changes, and also fix an RTP stack issue when + the auto option was used. (issue #8972 reported by mdu113) + +2007-02-03 20:44 +0000 [r53135-53136] Russell Bryant + + * apps/app_dial.c, /: Merged revisions 53133 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | + 4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when + the dial application exits early because of invalid arguments + instead of just leaving it empty. (issue #8975) ........ + + * /: Blocked revisions 53134 via svnmerge ........ r53134 | russell + | 2007-02-03 14:39:45 -0600 (Sat, 03 Feb 2007) | 2 lines Revert + some changes that accidentally got committed as a part of another + fix. ........ + +2007-02-03 10:02 +0000 [r53131] Paul Cadach + + * channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string + because due to compatibilities with CS1000 reported at + www.voip-info.org + +2007-02-02 21:26 +0000 [r53129] BJ Weschke + + * UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a + warning to the console that things might possibly be + misconfigured when queue member's states are still 'Not in Use' + when we're about to bridge them with a caller from queue. Also, + put some documentation quoted from oej's queues.txt efforts + started in /trunk today. This commit puts #7433 into feedback + state for 1.4, and pending no further negative feedback, it will + finally be closed. + +2007-02-02 17:15 +0000 [r53114-53120] Joshua Colp + + * main/rtp.c: Correct a copy/pasted error message line for RTCP. + + * main/config.c, /: Merged revisions 53117 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2 + lines Pass the glob expanded filename to process_text_line so + that error messages contain the actual filename, not the original + include one. (issue #8959 reported by tzafrir) ........ + + * Makefile: Add systemname to asterisk.conf generation per recent + discussions about it. (issue #8968 reported by blitzrage) + +2007-02-02 00:24 +0000 [r53109] Olle Johansson + + * channels/chan_sip.c, configs/sip.conf.sample: Disable the direct + p2p RTP call setup in SIP. You can enable it in sip.conf, but it + is now considered experimental until we solve the + AST_CONTROL_ANSWER with payload and videocaps stuff. + +2007-02-01 23:16 +0000 [r53108] Jason Parker + + * /: Blocked revisions 53107 via svnmerge ........ r53107 | qwell | + 2007-02-01 17:14:09 -0600 (Thu, 01 Feb 2007) | 2 lines Fix a + small typo. Synopsis lines shouldn't have a newline ........ + +2007-02-01 22:24 +0000 [r53097-53104] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 + lines Copy noncodeccapability over to the joint variable so that + telephone-event will get transmitted in the sent INVITE. ........ + + * main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile + here as well, but it apparently required both dev mode and no + optimizations to creep up. + + * /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 + lines Don't negotiate RFC2833 when not configured to do so. + (issue #8799 reported by mdu113) ........ + +2007-02-01 21:24 +0000 [r53093] Russell Bryant + + * funcs/func_strings.c: Fix the FIELDQTY function to not crash. + (reported by blitzrage and Corydon on IRC) + +2007-02-01 21:15 +0000 [r53091] Olle Johansson + + * /: Going backwards, blame file. + +2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp + + * /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb + 2007) | 2 lines Return previous behavior of having MOH pick up + where it was left off. (issue #8672 reported by sinistermidget) + ........ + + * funcs/func_strings.c: Make func_strings build under dev mode. + Didn't I do this today already in the berkeley DB? + +2007-02-01 21:05 +0000 [r53079-53085] Olle Johansson + + * channels/chan_sip.c: - Clean INC_COUNT flag when we decrement + call counter - If it's still set at time of dialog destruction, + make sure we decrement the device call counter properly before we + destroy the dialog + + * apps/app_queue.c: Change debug level for state change message + that is not really informative when debugging app_queue + + * channels/chan_sip.c: Cleaning up the devicestate callback + function + +2007-02-01 20:13 +0000 [r53075-53077] Tilghman Lesher + + * funcs/func_strings.c: Oops. + + * /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007) + | 2 lines Bug 8965 ........ + +2007-02-01 19:33 +0000 [r53072] Joshua Colp + + * main/asterisk.c: Add missing 'F' letter to getopt so it magically + becomes a valid option. (issue #8960 reported by tzafrir) + +2007-02-01 19:21 +0000 [r53070] Tilghman Lesher + + * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007) + | 2 lines No wonder FIELDQTY doesn't work with functions... the + documentation in pbx.c was wrong ........ + +2007-02-01 17:37 +0000 [r53064] Joshua Colp + + * channels/chan_sip.c: Fix silly logic. We really want to write + UDPTL frames out when the call is up. + +2007-02-01 16:35 +0000 [r53062] Olle Johansson + + * configs/sip.conf.sample: Add explanation of port= in combination + with defaultip= (thanks jsmith) + +2007-02-01 13:17 +0000 [r53060] Christian Richter + + * channels/chan_misdn.c: we update the name on any first reply of + our setup + +2007-02-01 11:07 +0000 [r53057] Paul Cadach + + * channels/chan_h323.c: chan_h323 is very stable, so let it built + by default + +2007-02-01 00:24 +0000 [r53050-53052] Joshua Colp + + * main/rtp.c: When going on hold have the side that was put on hold + reinvite back to Asterisk. When going off hold have the side that + was taken off hold reinvited back to the other party. + + * main/rtp.c: Add more frame types to forward in the RTP bridge + loops. + +2007-01-31 21:32 +0000 [r52859-53046] Russell Bryant + + * main/cdr.c, main/manager.c, pbx/pbx_spool.c, + channels/chan_skinny.c, channels/chan_h323.c, main/http.c, + pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c, + main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c, + channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c: + Merged revisions 53045 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | + 3 lines Fix a bunch of places where pthread_attr_init() was + called, but pthread_attr_destroy() was not. ........ + + * apps/app_userevent.c: Remove an extra \r\n from manager user + events. (issue #8955, mnicholson) + + * main/rtp.c, /: Merged revisions 53039 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | + 3 lines Use the proper format string to print unsigned values in + the rtp debug output. (issue #8954, wmis) ........ + + * apps/app_queue.c: Only changed the paused status in an existing + queue member if the paused column exists. + + * apps/app_queue.c: Instead of always creating a realtime queue + member as unpaused, read the "paused" column and use that value + for the paused status of the member. (issue #8949, jmls) + + * contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10. + (issue #8363, johnlange) + + * doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue + #8942, lters) + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + codecs/codec_gsm.c: When we are checking for a system installed + version of libgsm, we need to check for gsm.h as well. + Furthermore, when checking for this header, it may be located in + a gsm/ sub directory, so check for that, as well. (issue #8773) + + * /: Blocked revisions 52954 via svnmerge ........ r52954 | russell + | 2007-01-30 13:41:52 -0600 (Tue, 30 Jan 2007) | 4 lines Don't + print a message indicating that we don't know what to do with a + proceeding control frame in ast_request_and_dial(). We just need + to ignore it. (reported by JerJer on #asterisk-dev) ........ + + * channels/chan_sip.c: Only set the DTMF flag on the rtp structure + if the DTMF mode is actually RFC2833, not just that it is not + INFO. This makes it get set for inband DTMF as well, which is not + valid. (issue #8936) + + * main/asterisk.c, /: Merged revisions 52903 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | + 9 lines The SIGHUP handler was implemented to allow admins to + send SIGHUP to a running Asterisk process to reload the + configuration. However, doing the actual reload in the signal + handler itself is a very bad thing to do, because the reload + process includes calling non-reentrant functions such as + malloc/calloc/etc. If Asterisk is running in the background, then + the reload will happen immediately. However, if running in + console mode, the reload doesn't work until something is typed at + the console. That sort of defeats the purpose, but I don't see an + easy way to get around it at this point. ........ + + * /: Blocked revisions 52857 via svnmerge ........ r52857 | russell + | 2007-01-30 09:35:23 -0600 (Tue, 30 Jan 2007) | 5 lines Comment + out the parts in the Makefile that make codec_zap get built. It + will not yet build against zaptel 1.2, so I am disabling it to + prevent further bug reports until it gets merged. (issue #8940) + ........ + +2007-01-30 15:29 +0000 [r52856] Joshua Colp + + * channels/chan_iax2.c: Drop the deprecated show commands since the + original ones were changed back. (issue #8937 reported by + PCadach) + +2007-01-30 08:46 +0000 [r52807-52809] Paul Cadach + + * channels/chan_h323.c: Revert reprecation of h.323 gk cycle + command from pre-1.4 version instead of duplicated h323 cycle gk + + * res/res_odbc.c: Don't play with free()'d pointers + + * configure, acinclude.m4: Handle non-standard OpenH323/PWLib + library names + +2007-01-30 00:15 +0000 [r52763] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | + 5 lines Fix the extraction of the timestamp from video frames. It + was using the mapping for a mini-frame instead of a video-frame, + which caused it to get invalid data. (issue #8795, mihai) + ........ + +2007-01-29 23:43 +0000 [r52717] Joshua Colp + + * apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan + 2007) | 2 lines Now that filename is part of the structure and + since it comes before postprocess... we have to add it to our + postprocess line. (reported on asterisk-dev by Boris Bakchiev) + ........ + +2007-01-29 22:58 +0000 [r52688-52695] Russell Bryant + + * main/Makefile: Add a missing quotation mark. This was pointed out + by jcmoore on #asterisk-dev. + + * main/manager.c: Remove a recursive lock of the manager session. + This was pointed out by zandbelt in issue #8711. + +2007-01-29 22:12 +0000 [r52679] Tilghman Lesher + + * pbx/pbx_config.c: Argument number correction + +2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant + + * main/Makefile: ASTLDFLAGS needs to be passed to the editline + configure script as LDFLAGS. (issue #8928, zandbelt) + + * main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF + mode translation. P2P bridging can only be used when the DTMF + modes don't match if the core is monitoring DTMF in both + directions. Then, the core will handle the translation. + Otherwise, this bridging method can not be used. (issue #8936) + + * main/manager.c: The session lock can not be held while calling + action callbacks. If so, then when the WaitEvent callback gets + called, then no event can happen because the session can't be + locked by another thread. Also, the session needs to be locked in + the HTTP callback when it reads out the output string. This fixes + the deadlock reported in both 8711 and 8934. Regarding issue + 8711, there still may be an issue. If there is a second action + requested before the processing of the first action is finished, + there could still be some corruption of the output string buffer + used to build the result. (issue #8711, #8934) + +2007-01-29 18:59 +0000 [r52572] Joshua Colp + + * apps/app_voicemail.c: Use ast_calloc instead of malloc. + +2007-01-29 17:57 +0000 [r52535] Steve Murphy + + * apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR + backport to 1.4). It was committed to trunk via 7663. But it + wasn't so much an enhancement as a fix for the bad language + output for portuguese in Brazil, so, after a lot of prodding from + patient Brazilians, here is the same fix for 1.4 + +2007-01-29 17:33 +0000 [r52523] Joshua Colp + + * apps/app_voicemail.c: Set quota information to 0 when creating a + vm_state. (issue #8924 reported by neutrino88) + +2007-01-29 16:54 +0000 [r52506] Russell Bryant + + * main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in + the last commit to the adaptive jitterbuffer code. - Specifically + indicate to the compiler that the "dropem" variable only needs + one but. - Change formatting to conform to coding guidelines. + +2007-01-29 04:18 +0000 [r52494] Jim Dixon + + * main/jitterbuf.c, include/jitterbuf.h: Fixed problem with + jitterbuf, whereas it would not complain about, and would allow + itself to be overfilled (per the max_jitterbuf parameter). Now it + rejects any data over and above that size, and complains about + it. + +2007-01-28 05:15 +0000 [r52462] Tilghman Lesher + + * configure, configure.ac: Suggested change to fix normal usage of + --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing + list) + +2007-01-27 02:13 +0000 [r52335-52416] Joshua Colp + + * /, apps/app_queue.c: Merged revisions 52415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2 + lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log + follow documentation. (issue #7677 reported by amilcar) ........ + + * main/manager.c: Have the manager interface send back an "Already + logged in" message instead of "Invalid/Unknown Command" when the + client authenticates for a second time. (issue #8509 reported by + pari) + + * /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 + lines Make the last context entry read in the dominant one. + (issue #8918 reported by pj) ........ + + * main/file.c: Fix core show file formats CLI command. + +2007-01-25 19:18 +0000 [r52163-52265] Joshua Colp + + * /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2 + lines Allow dequeueing of frames with negative timestamp by + moving jitterbuffer frames check to jb_next. (issue #8546 + reported by harmen) ........ + + * channels/chan_sip.c: Drop out variables I accidentally put in. + + * channels/chan_sip.c: Decrement onHold count if we are hung up on + and still on hold. (issue #8909 reported by alexh42) + + * apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan + 2007) | 2 lines Add another note about audio files being played + back to each bridged party. (issue #8718 reported by ppyy) + ........ + +2007-01-25 01:37 +0000 [r52107-52160] Russell Bryant + + * apps/app_voicemail.c, configs/users.conf.sample: By suggestion + from kpfleming last week, change "vmpassword" to "vmsecret". + + * configure, configure.ac: Remove libnsl as a required lib for + libiksemel to work. This change was already made in the trunk. + (issue #8762) + + * /: Blocked revisions 52137 via svnmerge ........ r52137 | russell + | 2007-01-24 18:39:50 -0600 (Wed, 24 Jan 2007) | 3 lines Fix a + seg fault when running this application with no arguments from + AGI. (issue #8905, junky) ........ + + * include/asterisk/dial.h: Fix the formatting of doxygen comments + to properly indicate that the comment documents the previous + entity, as opposed to the next one. + +2007-01-24 18:26 +0000 [r52052] Steve Murphy + + * utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1 + line updated check_expr via 8322 (refactoring of expression + checking impl); elfring contributed a nice code reorg, I + contributed some time to get it working again, better messages + ........ + +2007-01-24 18:20 +0000 [r52016-52049] Joshua Colp + + * main/dial.c (added), apps/app_page.c, main/Makefile, + include/asterisk/dial.h (added): Merge in dialing API and the + app_page that uses it. (issue #BE-118) + + * channels/chan_sip.c: Fix changing channel formats when joint + capability changes and there are no audio formats... I didn't + break it originally! (issue #8535 reported by ivoc) + +2007-01-24 17:14 +0000 [r52000] Russell Bryant + + * configure: rebuild configure script to reflect last chan_h323 + related changes. + +2007-01-24 12:57 +0000 [r51979-51989] Christian Richter + + * channels/chan_misdn.c: added fix from #8899 + + * channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24 + Jan 2007) | 1 line fixed the busy problem (dialstatus was not + busy when we called a busy extension) ........ + +2007-01-24 09:30 +0000 [r51931] Olle Johansson + + * channels/chan_sip.c: Show capabilities *and* preference in + general settings in "sip show settings" (reported by Clona/Telio + - Thanks!) + +2007-01-24 08:04 +0000 [r51895] Paul Cadach + + * acinclude.m4: Allow x64 builds of H.323 (please, rebuild + configure) + +2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant + + * main/channel.c, /: Merged revisions 51843 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | + 6 lines Fix an issue related to synchronization of recordings + when using Monitor(). The bug is a miscalculation of the amount + to seek the stream for writing to disk when the number of samples + coming in and out of a channel do not match up. (issue #8298, + #8887, report and patch by guillecabeza, patch files created and + testing done by whoiswes) ........ + + * apps/app_while.c, /: Merged revisions 51828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) | + 4 lines Don't set a new value for the END_ variable on the + channel before using the old value. If you do, it will lead to + accessing a memory address that has been free()'d. (issue #8895, + arkadia) ........ + +2007-01-23 22:46 +0000 [r51788] Joshua Colp + + * channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c, + channels/chan_sip.c, channels/chan_skinny.c, + channels/chan_features.c, channels/chan_alsa.c, + channels/chan_gtalk.c, channels/chan_iax2.c: Update channel + drivers to use module referencing so that unloading them while in + use will not result in crashes. (issue #8897 reported by junky) + +2007-01-23 22:04 +0000 [r51750-51781] Russell Bryant + + * main/manager.c: Fix some bugs in process_message(). The manager + session lock needs to be held when sending some sort of response, + or calling one of the manager action callbacks. This resolves an + issue where people using the GUI would get random crashes when + they start clicking around a lot. (issue #8711, reported and + debugged by zandbelt) + + * main/http.c: Fix setting the default port of 8088 on 64-bit or + big-endian machines. + + * main/manager.c: When traversing the list of manager actions, the + iterator needs to be initialized to the list head *after* locking + the list. Also, lock the actions list in one place it is being + accessed where it was not being done. + +2007-01-23 20:32 +0000 [r51683-51716] Steve Murphy + + * res/res_features.c: this mod from 8593 (dstchannel in cdr is + empty when transfer call). + + * main/callerid.c: via 8748 (callerid.c loses name when returning + PRIVATE_NUMBER flag), the user suggested this mod, saying it + would allow 'WITHHELD' to appear in the name field, which would + be useful + +2007-01-23 10:28 +0000 [r51648-51649] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, + channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) | + 6 lines * more additions to make the RESTART message work * added + fix for misdn_call to allow SETUPs with empty extensions, + replaced the strtok_r functions with strsep for that (inspired by + Sandro Cappellazzo, thanks) ........ r50506 | crichter | + 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get + L2 UP, the L1 is UP definitely too, so we set the L1 state up as + well. ........ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c: manually merged r49922 and r50335, because + of conflicts. this commint includes addition of the ISDN RESTART + Message + +2007-01-23 06:51 +0000 [r51615] Paul Cadach + + * channels/chan_h323.c, channels/Makefile: Do not abort Asterisk + startup if h323 configuration file not found (reported by + mithraen) + +2007-01-23 03:00 +0000 [r51513-51558] Joshua Colp + + * channels/chan_sip.c: Only change audio formats on the channel if + we have an audio format to change to. (issue #8535 reported by + ivoc) + + * /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan + 2007) | 2 lines Yield before reading from zaptel timing source + under Solaris so that other threads get a chance to do things. + (issue #7875 reported by bob) ........ + +2007-01-22 19:41 +0000 [r51411] Russell Bryant + + * /: Blocked revisions 51410 via svnmerge ........ r51410 | russell + | 2007-01-22 13:39:30 -0600 (Mon, 22 Jan 2007) | 3 lines Merge + codec_zap support for the transcoder card. This is a standalone + codec module so it will not affect anything else. ........ + +2007-01-22 19:28 +0000 [r51409] Steve Murphy + + * pbx/pbx_ael.c: This fixes 8836, according to dnatural + +2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp + + * apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan + 2007) | 2 lines Move filestream creation to Mixmonitor loop. This + will prevent a blank file from being created if no frames ever + pass through to be recorded. (issue #7589 reported by + steve_mcneil) ........ + + * /: Blocked revisions 51359 via svnmerge ........ r51359 | file | + 2007-01-22 11:23:03 -0500 (Mon, 22 Jan 2007) | 2 lines Explicitly + declare what codecs are supported by default globally since using + a bitmask for all may include ones we don't need. (issue #8357 + reported by gknispel_proformatique) ........ + +2007-01-20 06:53 +0000 [r51348-51350] Jason Parker + + * configs/say.conf.sample: Fix Italian numeral support in say.conf + for "_[2-9]00" case. "2131" would've translated to something + along the lines of (pardon my..Italian {or lack thereof}) + "duecentocentotrentuno", which makes no sense at all. + + * configs/say.conf.sample: Fix German language support in say.conf + Properly support 21, 31, 41, 51, 61, 71, 81, and 91. + einundzwanzig has the same format as zweiundzwanzig (as do all + other "_ZX" spoken numerals) Fix support for numbers in the + 10,000,000 to 99,999,999 range. Add support for numbers in the + 100,000,000 to 999,999,999 range. + +2007-01-20 00:13 +0000 [r51302-51343] Russell Bryant + + * apps/app_meetme.c: Remove an unused instance of an unnamed enum. + + * apps/app_meetme.c: Remove another duplicated definition + + * apps/app_meetme.c: Remove a variable that was declared twice. + + * codecs/gsm/Makefile: Add a couple more processors that need + optimizations excluded. (issue #8637) + + * channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk. + AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same + thing. So, a digit would have been interpreted incorrectly here. + Since the channel driver will always have the begin and end + callbacks called for a digit, only support the button-down and + button-up messages. + + * .cleancount: Bump the cleancount since my last commit changed the + channel structure. + + * channels/chan_oss.c, main/rtp.c, main/channel.c, + channels/chan_phone.c, channels/chan_misdn.c, + channels/chan_skinny.c, channels/chan_features.c, + channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c, + channels/chan_zap.c, channels/chan_local.c, main/frame.c, + channels/chan_sip.c, channels/chan_agent.c, + include/asterisk/channel.h, channels/chan_gtalk.c, + channels/chan_iax2.c: Merge the changes from the + /team/group/vldtmf_fixup branch. The main bug being addressed + here is a problem introduced when two SIP channels using SIP INFO + dtmf have their media directly bridged. So, when a DTMF END frame + comes into Asterisk from an incoming INFO message, Asterisk would + try to emulate a digit of some length by first sending a DTMF + BEGIN frame and sending a DTMF END later timed off of incoming + audio. However, since there was no audio coming in, the DTMF_END + was never generated. This caused DTMF based features to no longer + work. To fix this, the core now knows when a channel doesn't care + about DTMF BEGIN frames (such as a SIP channel sending INFO + dtmf). If this is the case, then Asterisk will not emulate a + digit of some length, and will instead just pass through the + single DTMF END event. Channel drivers also now get passed the + length of the digit to their digit_end callback. This improves + SIP INFO support even further by enabling us to put the real + digit duration in the INFO message instead of a hard coded 250ms. + Also, for an incoming INFO message, the duration is read from the + frame and passed into the core instead of just getting ignored. + (issue #8597, maybe others...) + + * main/asterisk.c: Merged revisions 51300 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) | + 4 lines Fix a memory leak on command line tab completion. The + container for the matches was freed, but the individual matches + themselves were not. (issue #8851, arkadia) ........ + +2007-01-19 00:17 +0000 [r51272-51274] Dwayne M. Hubbard + + * channels/chan_zap.c: chan_zap compiles without libpri after + committing 7877 patch + + * channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) + | 3 lines issue 7877: chan_zap module reload does not use + default/initialized values on subsequent loads. Reset + configuration variables to default values prior to parsing + configuration file. ........ + +2007-01-18 23:36 +0000 [r51270] Kevin P. Fleming + + * /: block this patch since it is already here + +2007-01-18 22:50 +0000 [r51265] Jason Parker + + * apps/app_voicemail.c, main/channel.c, main/pbx.c, + funcs/func_strings.c, main/app.c: Add some more checks for + option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832, + patch(es) by tgrman + +2007-01-18 21:54 +0000 [r51262] Russell Bryant + + * Makefile, configure, main/Makefile, acinclude.m4, makeopts.in: + Ensure that the locations given to the Asterisk configure script + for ncurses, curses, termcap, or tinfo are further passed along + to the editline configure script. This fixes some + cross-compilation environments. (issue #8637, reported by ovi, + patch by me) + +2007-01-18 21:14 +0000 [r51256] Tilghman Lesher + + * /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18 + Jan 2007) | 2 lines If a timezone is not specified, assume + localtime (instead of gmtime) (Issue #7748) ........ + +2007-01-18 19:17 +0000 [r51251] Joshua Colp + + * apps/app_speech_utils.c: Only start timeout once we reach the end + of the files to play back. + +2007-01-18 18:42 +0000 [r51245] Jason Parker + + * main/cli.c: Fix an issue with file name completion in "module + load" and "load". Issue 8846 + +2007-01-18 18:36 +0000 [r51243] Joshua Colp + + * channels/chan_sip.c: Copy MOH settings when calling a peer so + that if they put someone on hold or get put on hold themselves + they get the right music class. (issue #8840 reported by mdu113) + +2007-01-18 18:28 +0000 [r51241] Jason Parker + + * main/channel.c: Fix an issue with deprecated commands + +2007-01-18 17:49 +0000 [r51236] Tilghman Lesher + + * contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18 + Jan 2007) | 2 lines Document all the fields, including the + indication that "uniqueid" should not be renamed. ........ + +2007-01-18 17:18 +0000 [r51233] Russell Bryant + + * main/manager.c: Make the "hasmanager" option in users.conf + actually have an effect. (issue #8740, LnxPrgr3) + +2007-01-18 00:48 +0000 [r51211-51213] Joshua Colp + + * apps/app_voicemail.c: Build the IMAP remote directory string + better and properly. Fix an issue with encoding the GSM voicemail + when attaching to the voicemail. (issue #8808 reported by + akohlsmith) + + * main/rtp.c: Pass data as well for hold/unhold/vidupdate frames. + (issue #8840 reported by mdu113) + +2007-01-17 23:31 +0000 [r51198-51205] Russell Bryant + + * funcs/func_odbc.c: Fix some instances where when loading + func_odbc, a double-free could occur. Also, remove an unneeded + error message. If the failure condition is actually a memory + allocation failure, a log message will already be generated + automatically. + + * channels/chan_zap.c: Instead of dividing the offset by 2 + directly, make it more clear that the offset is being scaled by + the size of the elements in the buffer. (Inspired by a discussing + on the asterisk-dev list about this code) + + * /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | + 3 lines Move the check for a failure of ast_channel_alloc() to + before locking the pvt structure again. Otherwise, on a failure, + this will cause a deadlock. ........ + +2007-01-17 20:56 +0000 [r51195] Tilghman Lesher + + * /, main/utils.c: Merged revisions 51194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007) + | 4 lines When ast_strip_quoted was called with a zero-length + string, it would treat a NULL as if it were the quoting character + (and would thus return the string in memory immediately following + the passed-in string). ........ + +2007-01-17 17:36 +0000 [r51186] Jason Parker + + * apps/app_voicemail.c: re-add "password" for realtime voicemail + +2007-01-17 06:36 +0000 [r51182] Joshua Colp + + * main/rtp.c: Return the correct result when directly writing out a + packet so that the core doesn't then decide to handle it the + regular way again. (issue #8833 reported by rcourtna) + +2007-01-17 01:29 +0000 [r51176] Kevin P. Fleming + + * apps/app_voicemail.c: a few more coding style cleanups and one + bug fix (from AnthonyL) + +2007-01-17 00:46 +0000 [r51172] Joshua Colp + + * channels/chan_iax2.c: Move rescheduling of lagrq/pings into the + scheduler callback. + +2007-01-17 00:20 +0000 [r51165-51170] Jason Parker + + * main/rtp.c: Fix issue with dtmf continuation packets when the + dtmf digit is 0... Issue 8831 + + * apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with + IMAP storage and realtime voicemail. Also update the vmdb sql + script for IMAP specific options. Issue 8819, initial patches by + bsmithurst (slightly modified by me) + + * doc/voicemail_odbc_postgresql.txt: change documentation to + reflect new procedure in 1.4/trunk + +2007-01-16 21:51 +0000 [r51159-51162] Tilghman Lesher + + * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions + 51161 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007) + | 2 lines Add documentation walkthrough on getting Postgres to + work with voicemail (from Issue 8513) ........ + + * apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007) + | 2 lines Postgres driver doesn't like a NULL pointer when + retrieving the length (Bug 8513) ........ + +2007-01-16 17:46 +0000 [r51150] Matt O'Gorman + + * apps/app_voicemail.c: minor things i missed before i get jumped + on + +2007-01-16 17:39 +0000 [r51148] Joshua Colp + + * /, res/res_features.c: Merged revisions 51145 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2 + lines Return previous behavior. ParkedCalls will be able to do + DTMF based transfers again. trunk however will get an option to + allow this to be set on/off. (issue #8804 reported by nortex) + ........ + +2007-01-16 17:36 +0000 [r51146] Jason Parker + + * main/file.c: Display more useful output when streaming files. + Include the channel name to which the file is being played. Issue + 8828, patch by junky. + +2007-01-16 05:55 +0000 [r51087] Joshua Colp + + * channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 + lines Add none as a valid callgroup/pickupgroup option. I + consider it a bug that it would inherit it all the way down and + not have any way to reset it to nothing - so that's why it is in + 1.2. (issue #8296 reported by gkloepfer) ........ + +2007-01-16 01:15 +0000 [r51057] Russell Bryant + + * main/config.c: It is possible for the config pointer to be NULL + here, so it needs to be checked before dereferencing it. + +2007-01-16 00:22 +0000 [r51030] Matt O'Gorman + + * apps/app_voicemail.c, configs/users.conf.sample: Patch allows for + changing voicemail password in users.conf from voicemail main, + written by AnthonyL bug #8436 + +2007-01-15 23:49 +0000 [r50994] Russell Bryant + + * Makefile.rules: Filter out a few CFLAGS that are not valid + CXXFLAGS. + +2007-01-15 23:10 +0000 [r50988] Tilghman Lesher + + * /: Blocked revisions 50987 via svnmerge ........ r50987 | + tilghman | 2007-01-15 17:09:02 -0600 (Mon, 15 Jan 2007) | 2 lines + Check return value before dereferencing (Bug 8822) ........ + +2007-01-15 21:08 +0000 [r50957] Matt O'Gorman + + * apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946 + | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4 + lines Solves issue with forwarding voicemails from folders other + than inbox. patch by anthonyl. ........ + +2007-01-15 18:23 +0000 [r50921] Jason Parker + + * main/asterisk.c: re-add deprecated "show version" CLI command. + +2007-01-15 16:36 +0000 [r50895] Joshua Colp + + * main/manager.c: Move event processing into do_message so that it + gets executed again when events are tripped. + +2007-01-15 15:03 +0000 [r50867] Kevin P. Fleming + + * configure, include/asterisk/autoconfig.h.in, main/Makefile, + configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the + ACX_PTHREAD macro from the Autoconf macro archive for setting up + compiler pthreads support... should improve portability to + platforms with unusual pthreads requirements + +2007-01-14 21:59 +0000 [r50820] Joshua Colp + + * main/astmm.c: Add missing newlines for two memory CLI commands. + +2007-01-14 05:13 +0000 [r50782] Tilghman Lesher + + * main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c, + main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c, + main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c, + main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c, + main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c, + main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c, + main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c, + main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c, + main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c, + main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h, + main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c, + main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c, + main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c, + main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c, + main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c, + main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13 + Jan 2007) | 2 lines Bug 8814 - db should look for its header + using a relative path, instead of the system path (Fixes FreeWRT) + ........ + +2007-01-13 16:45 +0000 [r50754] Kevin P. Fleming + + * Makefile, build_tools/make_sample_voicemail (added): when + building the sample greetings for maibox 1234@default during + 'make samples', build a greeting for each language and file + format the user selected to install with menuselect (reported by + Brian Capouch on asterisk-dev) + +2007-01-13 06:00 +0000 [r50674-50727] Joshua Colp + + * main/channel.c: Only write a frame out to the channel if one + exists. There are cases where one may not and would therefore + cause the channel driver to segfault. (issue #8434 reported by + slimey) + + * res/res_snmp.c: Only join the snmp thread on an unload if the + thread is actually running. (issue #8810 reported by junky) + +2007-01-12 19:24 +0000 [r50647] Jason Parker + + * configs/voicemail.conf.sample: Update documentation to state that + you shouldn't use realtime static with voicemail.conf + +2007-01-12 16:42 +0000 [r50602] Joshua Colp + + * main/manager.c: We need to check for res being 0 in do_message + itself, otherwise our headers will get lost. + +2007-01-12 14:42 +0000 [r50562] Kevin P. Fleming + + * main/pbx.c, /: Merged revisions 50561 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007) + | 2 lines minor documentation clarification ........ + +2007-01-11 05:53 +0000 [r50377-50468] Joshua Colp + + * channels/chan_sip.c: Remove check for channel state as it can + definitely be something other then ring, and also clean up the + code a bit. This should solve the parking issues and maybe some + attended transfer issues people have been seeing. + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add + support to see whether NAT was detected (yay symmetric RTP) and + also add a check in chan_sip so that if NAT has been detected and + the reinvite behind nat option has been turned off, then just do + partial bridge. (issue #8655 reported by mnicholson) + + * apps/app_speech_utils.c: Merge speech-multi branch which adds + support for joining multiple sound files together to be played + one after another in SpeechBackground. + + * main/config.c: Fix parsing when using something like ldap + settings. (done by anthonyl) + + * channels/chan_sip.c: Fix chan_sip not working issue. Let's not + prematurely return 0. (issue #8783 reported by st41ker) + +2007-01-10 16:45 +0000 [r50346] Jason Parker + + * cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made + it fail to load if the config file existed. Issue 8777 + +2007-01-10 04:55 +0000 [r50266-50298] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 50295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2 + lines Add another return value to dial_exec_full that indicates + execution is going to continuing at a new + extension/context/priority and to just let it slide. (issue #8598 + reported by jon) ........ + + * main/pbx.c: Ensure data's existence before trying to access it. + (issue #8774 reported by rcourtna) + +2007-01-10 02:17 +0000 [r50228] Russell Bryant + + * Makefile, /: Merged revisions 50227 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) | + 6 lines Make the number that represents the major version number + a single digit instead of 2. Using two digits makes it an octal + number when put into version.h, which breaks the compilation of + any out of tree module that checks the version for any version + after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev + mailing list, who gave credit to vihai for pointing it out) + ........ + +2007-01-09 17:11 +0000 [r50186] Jason Parker + + * main/cli.c: Re-add CLI command that should have only been + deprecated in 1.4. Thanks kshumard! (reported in person, so no + associated issue #) + +2007-01-09 13:40 +0000 [r50151] Tilghman Lesher + + * apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007) + | 4 lines The advent of realtime has enabled people to use commas + in the fullname field. This could cause an issue with sending + voicemails, when the field is unquoted. (Issue 8595) ........ + +2007-01-09 11:25 +0000 [r50124] Olle Johansson + + * channels/chan_sip.c: - handle re-invites properly in sip_hangup() + - Add some invitestate status changes just to be sure + +2007-01-08 23:39 +0000 [r50098] Jason Parker + + * apps/app_voicemail.c: Fix an issue with voicemail and users.conf, + where it wouldn't ever parse a password, since it was using + "secret" instead of "password" Issue 8761, reported by and patch + suggestion from ssokol. + +2007-01-08 21:11 +0000 [r50073] Matt O'Gorman + + * apps/app_senddtmf.c: we can't unlock a channel if we cant find + it. - AnthonyL bug #8741 + +2007-01-08 18:21 +0000 [r50032] Joshua Colp + + * main/rtp.c: Disable the more intense packet2packet bridging until + the bugs can be worked out. + +2007-01-08 14:26 +0000 [r49925-50006] Olle Johansson + + * channels/chan_sip.c: Issue #8677 - Handle failure of T.38 + re-invite This is not a fix, but adding an error message to tell + the admin that we have a bad configuration. We should not send + T.38 re-invites to devices that can't handle it (with the current + architecture where you have to hard-code t.38 support per + device). To really fix this, we need to figure out a way to tell + the incoming call that the re-invite failed, so we can signal + failure on that end and go back to the original call. + + * channels/chan_sip.c: Issue #8524, support multiple via header + values (tardieu) Thanks! + + * channels/chan_sip.c: We only need one forward declaration + + * channels/chan_sip.c: Issue 8735: Terminate state when extension + is unavailable for subscription + +2007-01-08 05:11 +0000 [r49890] Joshua Colp + + * /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 + lines Ensure we use the default refresh value of 60 if the remote + server does not send one. (issue #8746 reported by maethor) + ........ + +2007-01-08 03:53 +0000 [r49866] Kevin P. Fleming + + * configure, configure.ac: since we use AC_PATH_TOOL to find tools, + we should use the results it provides for us (reported by Brian + Capouch on the asterisk-dev list) + +2007-01-07 21:44 +0000 [r49831-49834] Tilghman Lesher + + * /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007) + | 2 lines If openstream fails, then we crash (Issue 8564) + ........ + + * channels/chan_sip.c: Second condition was a subset of the first, + so hold was never decremented, thus hint stayed stuck (Issue + 8747) + +2007-01-06 00:24 +0000 [r49742] Jason Parker + + * main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping + byte of allocated memory! This looks like it may have been a + chicken/egg scenario.. You had to call a cleanup func, because + everything was allocated. Then since you had to call a cleanup + func, you were forced to allocate - ie; strdup(""). + +2007-01-05 23:51 +0000 [r49710-49715] Kevin P. Fleming + + * configure, acinclude.m4: one more time... + + * configure, acinclude.m4: proper fix for r49712 + + * configure, acinclude.m4: if --with-foo= is specific for a + configure option, ensure that it is used for header file checking + as well + + * main/manager.c: ast_func_read() needs a writable copy of the + function name to be passed + +2007-01-05 23:16 +0000 [r49705] Jason Parker + + * channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and + chan_zap also depend on zaptel. This fixes an issue (8727) with + zaptel being in a different directory, using --with-zaptel. + +2007-01-05 22:52 +0000 [r49676-49680] Kevin P. Fleming + + * main/manager.c: don't 'consume' the params list before we try to + use it again + + * res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c, + main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c, + main/db.c, channels/chan_zap.c, channels/chan_sip.c, + apps/app_meetme.c, res/res_features.c, channels/chan_agent.c, + utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c, + apps/app_queue.c, res/res_jabber.c: reduce stack consumption for + AMI and AMI/HTTP requests by nearly 20K in most cases + +2007-01-05 22:14 +0000 [r49675] Joshua Colp + + * main/channel.c: Don't keep repeating the warning over and over + when the end of the call is reached. (issue #8724 reported by + xrg) + +2007-01-05 17:09 +0000 [r49581-49636] Kevin P. Fleming + + * /, channels/chan_sip.c, channels/chan_skinny.c, + channels/chan_iax2.c: Merged revisions 49635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) + | 2 lines ensure that threads which are supposed to be detached + (because we aren't going to wait on them) are created properly + ........ + + * channels/chan_iax2.c: revert the dynamic_list insertion change... + that was not the right thing to do + + * channels/chan_iax2.c: create the IAX2 processing threads as + background threads so they will use smaller stacks when we create + a dynamic thread, put it on the dynamic_list right away so we + don't lose track of it + +2007-01-04 23:00 +0000 [r49568] Joshua Colp + + * channels/chan_iax2.c: It's possible for the iax2 pvt to + disappear, so if it has... don't bother looking for dpentries. + +2007-01-04 22:51 +0000 [r49553] Kevin P. Fleming + + * include/asterisk/threadstorage.h, main/asterisk.c, + build_tools/cflags.xml, include/asterisk.h, main/Makefile, + main/threadstorage.c (added), main/utils.c: add support for + tracking thread-local-storage objects that exist via + 'threadstorage' CLI commands + +2007-01-04 22:28 +0000 [r49551] Joshua Colp + + * main/config.c: Only free comments and line buffer once we reach + the first level. (issue #8678 reported by ssokol, fixed by + anthonyl) + +2007-01-04 21:58 +0000 [r49460-49536] Kevin P. Fleming + + * channels/iax2-parser.c, main/frame.c: don't mark these + allocations as 'cache' allocations when caching has been disabled + + * channels/iax2-parser.c: if we're going to decrement the frame + count when we free a frame, we should inrement it when we create + one :-) + + * channels/iax2-parser.c, channels/iax2-parser.h, + channels/chan_iax2.c: only do IAX2 frame caching for voice and + video frames + + * main/frame.c: don't do frame header caching in the core if + LOW_MEMORY is defined + + * channels/iax2-parser.c: don't define this type either if + LOW_MEMORY is enabled + +2007-01-04 18:11 +0000 [r49459] Matt O'Gorman + + * apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447 + | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2 + lines converted a lot of 256 to PATH_MAX and some white space + fixes. ........ + +2007-01-04 18:06 +0000 [r49457-49458] Kevin P. Fleming + + * channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode + + * codecs/Makefile: make building of codec_gsm against the system + GSM library actually work + +2007-01-04 16:50 +0000 [r49413] Matt O'Gorman + + * apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412 + | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3 + lines good catch russell sorry i missed that. fix magic number + with proper sizeof ........ + +2007-01-04 04:33 +0000 [r49388] Russell Bryant + + * funcs/func_realtime.c: Fix the REALTIME() dialplan function. + ast_build_string() advances the string pointer to the position to + begin the next write into the buffer. So, this pointer can not be + used to copy the contents of the string later. The beginning of + the buffer must be saved. Interestingly enough, this code could + not have ever worked. (Pointed out by Sebb on IRC, thanks!) + +2007-01-03 23:32 +0000 [r49355] Matt O'Gorman + + * apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from + https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354 + | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6 + lines When using ODBC_STORAGE VoicemailMain doesn't create the + subdirectories for a mailbox such as the INBOX directory. this + patch solves that problem, was written by anthony be-125 ........ + +2007-01-03 09:06 +0000 [r49313] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c, + /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, + configs/misdn.conf.sample: Merged revisions + 48319,48321,48467,48552,48576,49135,49303 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | + 1 line changed a few debugs to higher debug levels ........ + r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | + 1 line added the export and import of the MISDN_ADDRESS_COMPLETE + Variable to inidcate wether the extension is already completely + dialed or if there might come additional digits by information + elements. also added some docs for that. ........ r48467 | + crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line + removed FIXUP state. added check for channel allocation conflict + when we create a setup while the other site creates a setup on + the same channel, besides the check we resolve this conflict. + ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 + Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a + preselected channel we just accept it, even when we're NT. added + some checks for segfaults. ........ r48576 | crichter | + 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we + reject a channel, because it's in use already, we shouldn't + process the setup anymore. made the channel allocation a bit + easier and more understandable, removed a few unused lines + ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 + Jan 2007) | 1 line added check for channel ranges in the + set/empty channel functions. set pmp_l1_check default to no. + added misdn restart pid cli command. added cleaning of channel + when we send a RELEASE_COMPLETE. ........ r49303 | crichter | + 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added + check for bridging in misdn_call to avoid setting + echocancellation when 2 mISDN channels are involved and when + bridging is set. That lead to a kernel panic before under + different situations, because we switched about 2 times between + hardware bridging and echocancelation * readded MISDN_URATE + variable which got lost before, this should make app_v110 work + again * fixed typo ........ + +2007-01-03 03:21 +0000 [r49282] Kevin P. Fleming + + * Makefile, Makefile.rules: various Makefile improvements to get + chan_vpb (and any other C++ modules) to build properly + +2007-01-03 01:19 +0000 [r49259] Joshua Colp + + * channels/chan_iax2.c: Check pvt structure presence before passing + to send_command. This gets rid of the irritating message about a + packet without pvt structure. This happens because the scheduled + item is getting cancelled at almost the exact moment it is + getting executed. + +2007-01-02 22:30 +0000 [r49237] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, + pbx/ael/ael.flex: This is a slight modification to Josh's edits + for #8579; both files edited were the produced by flex; so the + source files need to be changed instead, and the generated files + regenerated. + +2007-01-02 19:58 +0000 [r49212] Olle Johansson + + * channels/chan_sip.c: Small cleanup of add_t38sdp - it's always + enabled at that point in the code + +2007-01-02 17:33 +0000 [r49189] Jason Parker + + * main/pbx.c: Allow fractions of a second in the Wait() + application, like it says it allows. + +2007-01-02 13:59 +0000 [r49165] Kevin P. Fleming + + * channels/chan_zap.c: remove comment that is unrelated to this + function + +2007-01-02 12:08 +0000 [r49145] Olle Johansson + + * configs/features.conf.sample: Adding note on effect of + applicationmap features on re-invites + +2007-01-01 23:34 +0000 [r49098-49102] Kevin P. Fleming + + * channels/chan_zap.c, build_tools/menuselect-deps.in, configure, + configure.ac, codecs/codec_zap.c: check specifically for VLDTMF + and transcoding support in the system's Zaptel installation, and + make only the modules that need those features dependent on them + (this will allow building the other Zaptel-using parts of + Asterisk against older versions of Zaptel or those on other + platforms that haven't caught up yet to the Linux version) + + * Makefile: use a simpler (and portable) method to ensure that + menuselect is built as a host binary + + * Makefile: revert this change until a better solution can be + found... 'env -i' was not being used properly, but even when + changed to do so, this process fails during cross-compilation + because the menuselect build still sees 'CC' as set to the + cross-compiler + +2007-01-01 20:14 +0000 [r49096] Olle Johansson + + * channels/chan_sip.c: remove incomplete implementation of dnsmgr. + Let's fix this in trunk. + +2006-12-30 18:31 +0000 [r49063-49073] Joshua Colp + + * pbx/pbx_config.c: IAX has been deprecated for quite some time so + we had better use IAX2 when creating the dial string for users. + (issue #8697 reported by ssokol) + + * channels/chan_zap.c: Use asprintf to build the channel names + instead of custom function. I believe the custom function is + doing some things that are not portable across all + implementations. (issue #8570 reported by hterag & issue #8692 + reported by nicolasg) + + * main/rtp.c: If the Packet2Packet bridge is being broken because + of a masquerade then attempt to read a frame in so the masquerade + actually happens. Otherwise weirdness will occur. (issue #8696 + reported by kjotte) + + * channels/chan_iax2.c: Initialize the packet queue in load_module + instead of just declaring the list with the default value. (issue + #8695 reported by ssokol) + +2006-12-30 00:40 +0000 [r49061] Steve Murphy + + * pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have + comma args converted to vertical bars. I hope this change does + little harm. + +2006-12-29 00:50 +0000 [r49042-49048] Kevin P. Fleming + + * /: put this value into the correct property + + * /, BUGS: Merged revisions 49045 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006) + | 2 lines location of the bug posting guidelines has changed + ........ + + * sample.call: simple commit to test CIA integration + +2006-12-28 21:26 +0000 [r49032-49035] Jason Parker + + * main/cli.c: Fix some deprecated commands. Issue 8682, patch by me + + * main/http.c: saw this in passing... fix a small typo + +2006-12-28 20:08 +0000 [r49028] Kevin P. Fleming + + * sounds/Makefile: new versions of sounds + +2006-12-28 19:52 +0000 [r49024] Jason Parker + + * main/http.c: make the uris_lock a rwlock instead of a mutex lock + - needs to be forward ported to trunk + +2006-12-28 19:43 +0000 [r49022] Joshua Colp + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/lock.h: Backport support for read/write locks. + +2006-12-28 19:21 +0000 [r49020] Steve Murphy + + * main/ast_expr2.fl, main/ast_expr2.c, main/frame.c, + pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, + pbx/ael/ael_lex.c, include/asterisk/ael_structs.h, + pbx/ael/ael.tab.h, utils/ael_main.c: removed as in trunk + from the ael stuff. Also, threw in a minor fix to frame.c to + avoid build-killing compiler warnings. + +2006-12-27 22:28 +0000 [r49009] Joshua Colp + + * main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not + available when LOW_MEMORY is used and things are being built in + the utils directory, so we need to resort to the old method of + strncpy. (issue #8579 reported by mottano) + +2006-12-27 22:06 +0000 [r48998-49006] Kevin P. Fleming + + * main/enum.c, main/asterisk.c, main/rtp.c, main/term.c, + main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c, + main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c, + main/http.c, main/logger.c: since these variables all have static + duration, none of them need initializers (they default to zero + anyway) + + * include/asterisk/options.h, main/asterisk.c, main/file.c: move + extern declaration for this option to a header file where it + belongs provide an initial value for 'languageprefix' option, + instead of relying on randomness to provide a useful value + +2006-12-27 21:06 +0000 [r48993-48997] Olle Johansson + + * channels/chan_sip.c: Only include acl.h and lock.h once + + * channels/chan_sip.c: Only set rfc2833compensate flag once + (handle_common_options) + + * channels/chan_sip.c: - Remove checking for T38 options twice. + Keeping them in handle_common_options + +2006-12-27 18:33 +0000 [r48987-48988] Kevin P. Fleming + + * channels/chan_sip.c: make the option actually match the + documentation + + * channels/iax2-parser.c, include/asterisk/utils.h, + include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show + memory' and 'show memory summary' to distinguish memory + allocations that were done for caching purposes, so they don't + look like memory leaks + +2006-12-27 17:59 +0000 [r48975-48985] Olle Johansson + + * channels/chan_sip.c, configs/sip.conf.sample: Be a bit more + politically correct + + * channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy + cisco MWI support. Normally we try not to change our software for + bugs in other devices. But in this case, the Cisco phones are so + widespread so we try to implement a fix while waiting for a + bugfix from Cisco. + + * channels/chan_sip.c: - Make sure handle_common_options return 1 + when we found a common option - Move uncommon (only global) + option away from handle_common_options Reported by rizzo. Thanks! + + * channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before + re-sending invite with auth. + + * /, channels/chan_sip.c: Fix bogus content-length in t38 sdp. + (rizzo, #8600) + +2006-12-26 05:20 +0000 [r48960-48966] Joshua Colp + + * apps/app_meetme.c: Get rid of a needless memory allocation and + only create a conference structure in find_conf_realtime if data + was read from realtime. (issue #8669 reported by robl) + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an + API call that initializes an RTP structure. We need this because + chan_sip is cheeky and uses a temporary RTP structure for codec + purposes, and the API calls that are used rely on the lock. + (Pointed out on asterisk-dev by Andy Wang) + + * configure, configure.ac: Clean up autoconf file (gets rid of + warnings seen when rebuilding configure) and rebuild configure. + +2006-12-25 05:21 +0000 [r48931-48956] Russell Bryant + + * /, funcs/func_math.c: Merged revisions 48955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) | + 6 lines Fix an error introduced by copying and pasting the + handling of the >= operator for the MATH function. If a single + equal sign was used as an operator, the function would treat it + is as if it were the >= operator. Now, it properly handles it as + an invalid operator. (issue #8665, patch by tempest1) ........ + + * channels/chan_oss.c: Fix a typo in an error message that + indicated that the MGCP channel type could not be registered, + instead of the correct type, OSS. + + * /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | + 3 lines Check for the proper return value on an error in a call + to mmap(). This was reported by Andy Wang on the asterisk-dev + list. Thanks! ........ + + * /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | + 3 lines Remove a couple of misplaced dots in log messages. This + was reported by Andrea Spadaccini on the asterisk-dev mailing + list. ........ + + * main/http.c: Implement locking for the list of URI handlers to + make it thread-safe. + +2006-12-23 Kevin P. Fleming + + * Asterisk 1.4.0 released. + +2006-12-22 22:33 +0000 [r48870-48906] Jason Parker + + * Makefile, main/stdtime/localtime.c: Minor fixes for Solaris. + + * channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia + +2006-12-21 20:26 +0000 [r48783] Joshua Colp + + * /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2 + lines Add new silence sound files to the spec for Redhat. (issue + #8652 reported by alvaro_palma_aste) ........ + +2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp + + * apps/app_voicemail.c: vms doesn't exist on non-IMAP storage + builds. + + * apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so + it is then passed to the IMAP store file function. (issue #8614 + reported by punknow) + + * doc/snmp.txt: find is not the same as bind when it comes to + documentation. (issue #8626 reported by johann8384) + +2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming + + * channels/Makefile: suppress compiler warnings in this module + until it can be improved + +2006-12-19 21:12 +0000 [r48585] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 48584 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2 + lines Free localuser structure when we fail to dial (issue #8612 + reported by rizzo) ........ + +2006-12-19 21:03 +0000 [r48583] Luigi Rizzo + + * apps/app_sms.c: fix a bogus datalen in the frames generated by + app_sms (causing noisy output if you listen to the output!) This + affects trunk as well, whereas 1.2 is ok. + +2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming + + * res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable + type for these unixODBC API calls, eliminating warnings on 64-bit + platforms that use the 'new' 64-bit types for ODBC API calls + +2006-12-19 03:46 +0000 [r48571] Joshua Colp + + * Makefile: Use env -i to start a fresh environment when going to + build menuselect. This is more portable then using unset. (issue + #8543 reported by jtodd) + +2006-12-18 17:23 +0000 [r48566] Luigi Rizzo + + * include/asterisk/channel.h: unbreak the macro used for + incrementing the frame counters. I don't know when the bug was + introduced, but with the typical usage c->fin = + FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects + trunk as well (fix coming). + +2006-12-18 17:15 +0000 [r48564] Joshua Colp + + * channels/chan_iax2.c: Put thread into proper list if we abort + handling due to an error, and also hold the lock while putting it + back into the proper idle list so we don't prematurely get a + signal. (issue #8604 reported by arkadia) + +2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming + + * codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile, + utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile, + utils/ael_main.c: remove some now-unnecessary explicit includes + of autoconfig.h clean up per-file dependencies during 'make + clean' + + * build_tools/prep_tarball: need an additional argument here to + make the downloads actually occur + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep + these calls from thinking they have multiple arguments + + * codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile, + funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast, + main, codecs/gsm, pbx, res, channels, codecs, utils, agi, + main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr: + simplify dependency tracking system, using the compiler's + built-in method for generating them, and only doing dependency + tracking if developer mode is enabled via the configure script + + * Makefile, include/asterisk.h, main/stdtime/localtime.c: since we + really, really have to have autoconfig.h included before all + other headers (especially system headers), the Makefile will now + force it to happen (this will fix build problems with files like + ast_expr2f.c, where we can't control the inclusion order in the + file itself) + + * funcs/func_curl.c: instead of initializing the curl library every + time the CURL() function is invoked, do it only once per thread + (this allows multiple calls to CURL() in the dialplan for a + channel to run much more quickly, and also to re-use connections + to the server) (thanks to JerJer for frequently complaining about + this performance problem) + +2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp + + * main/rtp.c: Turn payload_lock into bridge_lock and make it + encompass all RTP structure contents that may relate to bridge + information, including who we are bridged to. + + * channels/chan_iax2.c: Hold call structure lock in places where a + qualify or peer action can destroy it. + + * channels/chan_iax2.c: Lock network retransmission queue in all + places that it is used. + +2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson + + * /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported + from 1.2) + + * channels/chan_sip.c: Update to latest IANA spec + +2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp + + * channels/chan_iax2.c: Use a wakeup variable so that we don't wait + on IO indefinitely if packets need to be retransmitted. + + * main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP + structure can change AFTER a bridge has started. This comes from + the packet handling of the SIP response when indication that it + was answered has been sent. Therefore we need to protect this + data with a lock when we read/write. (issue #8232 reported by + tgrman) + + * main/rtp.c: Remove direct RTCP bridging. I've come to the + conclusion that we should handle this through the core and not + just forward it on. Should solve a few bugs. + +2006-12-12 Kevin P. Fleming + + * Asterisk 1.4.0-beta4 released. + +2006-12-12 04:13 +0000 [r48401] Joshua Colp + + * apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This + is the way it should have been done. + +2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman + + * sounds/Makefile: new sounds package with 100% more silence + + * /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge + from https://svn.digium.com/svn/asterisk/branches/1.2 ........ + r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) + | 4 lines app_externalivr needs a real silence file, and + additional changes to add silence files into core instead of + extra patch provided by bug 8177 with minor additions. ........ + +2006-12-11 21:31 +0000 [r48391] Joshua Colp + + * apps/app_voicemail.c: Return non-existant callerid handling to + that which it was before. In 1.4 and trunk callerid can be + allocated but not have any contents so we have to use + ast_strlen_zero before passing it to the relevant functions. + (issue #8567 reported by pabelanger) + +2006-12-11 05:37 +0000 [r48382] Tilghman Lesher + + * funcs/func_strings.c: STRFTIME() does not actually require an + argument (issue 8540) + +2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp + + * main/rtp.c: Merge in my latest RTP changes. Break out RTP and + RTCP callback functions so they no longer share a common one. + + * apps/app_meetme.c: Use the correct API call to say a device state + changed. (Yes, I'm a nub.) + + * apps/app_meetme.c: Don't access the conference structure after it + has been freed. + +2006-12-11 00:47 +0000 [r48375] Tilghman Lesher + + * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c, + res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c, + apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) + | 5 lines When doing a fork() and exec(), two problems existed + (Issue 8086): 1) Ignored signals stayed ignored after the exec(). + 2) Signals could possibly fire between the fork() and exec(), + causing Asterisk signal handlers within the child to execute, + which caused nasty race conditions. ........ + +2006-12-10 03:04 +0000 [r48372] Steve Murphy + + * channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 + line This version applies the patch suggested by stevens in bug + 7836 (make inbound channel RINGING state consistent with other + channels). ........ + +2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant + + * channels/chan_iax2.c: Use locking when accessing the + registrations list. This list is not actually used very often, so + the likelihood of there being a problem is pretty small, but + still possible. For example, if the CLI command to list the + registrations was called at the same time that a reload was + occurring and the registrations list was getting destroyed and + rebuilt, a crash could occur. In passing, go ahead and convert + this list to use the linked list macros. + + * /: Blocked revisions 48361 via svnmerge ........ r48361 | russell + | 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use + locking when accessing the registrations list. This list is not + actually used very often, so the likelihood of there being a + problem is pretty small, but still possible. For example, if the + CLI command to list the registrations was called at the same time + that a reload was occurring and the registrations list was + getting destroyed and rebuilt, a crash could occur. ........ + +2006-12-07 18:17 +0000 [r48357] Russell Bryant + + * /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07 + Dec 2006) | 3 lines Ensure that the file position is not + incremented beyond the total number of files available for + playback. (issue #8539, ulogic) ........ + +2006-12-07 15:33 +0000 [r48349] Steve Murphy + + * main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that + killed bug 8423 -- OriginateSuccess and OriginateError incomplete + channel name. May it rest in peace. + +2006-12-06 16:25 +0000 [r48326] Olle Johansson + + * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being + retransmitted to Asterisk + +2006-12-06 16:15 +0000 [r48323] Russell Bryant + + * configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 + Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option + in the sample configuration file. (issue #8526, arkadia) ........ + +2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson + + * /, channels/chan_sip.c: Don't send Contact on MESSAGE + +2006-12-05 20:42 +0000 [r48279] Jason Parker + + * configure.ac: Fix curl version number testing to be much more + friendly to non-bash shells. Issue 8508, patch by me. This + *SHOULD* be POSIX compliant now.. + +2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson + + * channels/chan_sip.c: Merging the invitestate-1.4 branch after + successful testing. Will check if I can solve this with less + changes in 1.2. + + * configs/sip.conf.sample: Add missing s from another repository. + (thanks jcmoore!) + + * configs/sip.conf.sample: Updating sip.conf.sample with + information about T38 not working when chan_local or chan_agent + is involved in the call. I don't know how big a fix that would be + to solve, but this is the current state of affairs. (Chan_sip + currently checks if the other side of the bridge has a SIP tech. + We could/should implement another check, possibly for udptl_write + or some flag in the ast_channel structure). + +2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher + + * apps/app_voicemail.c: Oops, forgot to release the odbc handle + + * apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) + | 6 lines If the recording in the database is too large, it will + fail to retrieve with an mmap error. Not too sure why this + doesn't happen when we put it in the database, also, but since + that doesn't seem to be broken, I'm not going to fix it (at least + until someone reports it). Solution is to ask for the file in + smaller chunks. (Bug 8385) ........ + +2006-12-04 21:48 +0000 [r48237-48248] Jason Parker + + * apps/app_voicemail.c: Fix an issue which didn't allow + unavail/greet/busy/etc messages from being saved into ODBC (and + probably IMAP). + + * /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell | + 2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert + change from 8016 - this breaks other stuff... Needs further + review. Tip: When you've reported a bug about something and + somebody has put up a patch for it.. It's not a good idea to open + a completely new bug and say that something is broken because of + the patch in the other bug - PLEASE mention something in the bug + where the patch was actually created. ........ + + * /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell | + 2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an + issue where a message isn't saved correctly when using ODBC + storage and reviewing a message. Issue 8016 - patch by sokhapkin. + ........ + +2006-12-04 18:16 +0000 [r48234] Joshua Colp + + * /: Blocked revisions 48233 via svnmerge ........ r48233 | file | + 2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the + generic bridge tells us not to retry, and we have a frame to spit + out then break the bridge. Props to markit in #asterisk-bugs for + bringing this up. ........ + +2006-12-04 17:54 +0000 [r48228-48230] Jason Parker + + * configs/voicemail.conf.sample: Add documentation to + voicemail.conf.sample for ODBC storage. Issue 8499 - patch by + blitzrage. + + * doc/snmp.txt: Attempt to document some of the dependencies that + are needed for net-snmp Issue 8499 - initial patch by blitzrage. + +2006-12-03 06:34 +0000 [r48223] Russell Bryant + + * sounds/Makefile: When "fetch" is in use, instead of "wget", + --continue is not a valid option. (issue #8451) + +2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson + + * channels/chan_sip.c: - Removing one of two pieces of code to + handle 481 response on INVITE - Move handling of REFER response + to handle_response_refer() + + * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h, + configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax + transmission happens - Encapsulate RTP timers in the rtp + structure so we have one for video and one for audio The video + one is not used in 1.4, really. Will be used for RTP keepalives + when we can send something that video phones support in the RTP + stream. I now this is a big architectual change at this stage for + 1.4, but decided it was needed to avoid future bug reports. - + Document the RTP NAT keepalive option in sip.conf.sample Issue + 7679 in the bug tracker. Please test. + +2006-12-02 03:50 +0000 [r48195] Russell Bryant + + * include/asterisk/utils.h: Backport the comment containing the + warning regarding the limitations on the usage of this function. + It is thread safe, but not technically reentrant. + +2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming + + * apps/app_dial.c, /: Merged revisions 48192 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) + | 2 lines if Dial() is going to send music-on-hold to the calling + party, it has to send PROGRESS first to ensure that the reverse + audio path has been setup first (BE-106) ........ + +2006-12-01 23:16 +0000 [r48190] Russell Bryant + + * Makefile, configure, configure.ac, makeopts.in, sounds/Makefile: + FreeBSD 6.1 does not include wget by default. However, it has + fetch which will work just fine for our purposes of downloading + the sounds packages. So, check for both wget and fetch and the + configure script and use what was found to download them. If + neither one was found, and sound packages are selected that must + be downloaded, the install process will print out an informative + error message indicating the situation. Also, fix a couple places + where "make" was hard coded into some output messages by + replacing them with the $(MAKE) variable. (issue #8451, initial + patch by pabelanger, with additional modifications by me) + +2006-12-01 20:25 +0000 [r48184-48186] Jason Parker + + * configs/extensions.conf.sample, /: Merged revisions 48183 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 + lines Fix a small typo - issue 8848, reported by pabelanger + ........ + +2006-12-01 19:38 +0000 [r48179] Tilghman Lesher + + * main/cli.c: Double-unlock error (reported by blitzrage on IRC) + +2006-12-01 17:41 +0000 [r48177] Olle Johansson + + * channels/chan_sip.c, configs/sip.conf.sample: - Backport of the + "limitonpeers" patch from trunk, to fix a lot of issues with + queues and SIP device states - Remove support for T.38 early + media, since it's impossible. (Two patches in one - extra friday + evening offer due to being off line from svn today... :-) + +2006-11-30 21:18 +0000 [r48168] Joshua Colp + + * main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not + do a partial bridge for Google Talk since we need to handle STUN. + (issue #8448 reported by phsultan) + +2006-11-30 20:51 +0000 [r48166] Olle Johansson + + * /, channels/chan_sip.c: Issue 8319 - change noncecount before + using it. + +2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp + + * /: Blocked revisions 48161 via svnmerge ........ r48161 | file | + 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't + write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel + driver. (issue #8390 reported by hselasky) ........ + + * /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 + lines Only print out debug message if bridged channel is not + NULL. (issue #8412 reported by jubilex) ........ + + * /, res/res_features.c: Merged revisions 48154 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 + lines Do not listen for DTMF on the bridge that comes into + existence when ParkedCall is executed. This means native bridging + can now occur for this. (issue #8406 reported by kebl0155) + ........ + + * main/cdr.c, /: Merged revisions 48151 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 + lines Print certain CDR messages out at the NOTICE level versus + WARNING since they can occur when used with the CDR applications + and are perfectly fine. (issue #8367 reported by dartvader) + ........ + + * /: Blocked revisions 48146 via svnmerge ........ r48146 | file | + 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember + the pointer to the allocated block of memory so that we can free + it and not cause a memory leak. (issue #8449 reported by arkadia) + ........ + + * /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov + 2006) | 2 lines Document 'port' for SIP peers, came up because of + the current mailing list thread. (issue #8450 reported by + blitzrage) ........ + +2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson + + * doc/manager.txt: Explain status reports and make codefreeze more + happy :-) + + * /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by + GS 487 adapter without CSEQ on separate line in the REGISTER + request. Imported from 1.2. + +2006-11-29 21:05 +0000 [r48115] Joshua Colp + + * apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in + mm_login. (issue #8420 reported by slimey) + +2006-11-29 19:56 +0000 [r48113] Olle Johansson + + * configs/sip.conf.sample: Explain the use device status system + implemented in SIP for subscriptions, queues and manager a bit + better. Like in 1.2, you will get more detailed information if + you set a call limit for a device. When the call limit is + reached, the status system will report a device as busy. For + queues, setting a call limit per SIP device is propably a + requirement. In most cases, it will work much better if you only + use type=peer and not type=friend. We might decide to backport + the new setting from trunk to apply all call limits to the peer + part of a friend only. + +2006-11-29 16:50 +0000 [r48107] Joshua Colp + + * main/rtp.c, /: Merged revisions 48106 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 + lines If the frame was duplicated before writing out then we need + to free it. (issue #8429 reported by edguy3) ........ + +2006-11-29 08:03 +0000 [r48105] Olle Johansson + + * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma. + +2006-11-29 04:26 +0000 [r48101] Joshua Colp + + * apps/app_voicemail.c: Don't crash if the mailstream was not + created. + +2006-11-28 18:26 +0000 [r48095] Jason Parker + + * Makefile: Export several more variables in top level Makefile. + Inspired by issue 8438. + +2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp + + * channels/chan_phone.c, /: Merged revisions 48087 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov + 2006) | 2 lines According to the research I have done we never + needed to include compiler.h in the first place so let's not! + (issue #8430 reported by edguy3) ........ + + * apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 + lines Use the proper function to get the new message count + instead of always using the filesystem. (issue #8421 reported by + slimey) ........ + +2006-11-27 17:20 +0000 [r48049] Tilghman Lesher + + * /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 + Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381) + ........ + +2006-11-27 17:17 +0000 [r48046] Russell Bryant + + * main/manager.c: Remove a couple of unused variables (issue #8380, + casper) + +2006-11-27 15:32 +0000 [r48038] Joshua Colp + + * pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 + lines Do not reference the freed outgoing structure in the debug + message. (issue #8425 reported by arkadia) ........ + +2006-11-27 06:41 +0000 [r48031] Olle Johansson + + * channels/chan_sip.c: Change logging message + +2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy + + * funcs/func_cdr.c: might as well also document the raw values of + the flag vars + + * /, funcs/func_cdr.c: A little bit of func_cdr documentation + upgrade-- no bug# involved, although 8221 may have inspired it. + +2006-11-25 09:28 +0000 [r48002] Olle Johansson + + * /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4 + and future releases, you can disable subscription support totally + or per peer in sip.conf with allowsubscribe = yes | no + +2006-11-24 17:17 +0000 [r47992] Steve Murphy + + * main/translate.c: bug 8189 posted this fix for main/translate.c + for PLC + +2006-11-24 15:46 +0000 [r47989] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 + Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE. + beatufied some logs, changed some loglevels. changed the default + value of block_on_alarm ........ + +2006-11-23 11:01 +0000 [r47959] Olle Johansson + + * /, channels/chan_sip.c: Don't allocate unused variable. + +2006-11-22 21:47 +0000 [r47944] Joshua Colp + + * main/rtp.c: Video will never reach Packet2Packet bridging and can + do more harm then good. + +2006-11-21 17:32 +0000 [r47897] Joshua Colp + + * main/rtp.c: If we have the non standard G726-32 setting turned on + we want to return G726-32 to the SDP, not our AAL2 string. (issue + #8330 reported by voipgate) + +2006-11-21 15:20 +0000 [r47892] Olle Johansson + + * channels/chan_sip.c: Apparently Exosip sends a 101 after a 100 + provisional response. Let's not treat that as early media. + (discovered at the AVTF meeting in Paris). + +2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher + + * apps/app_voicemail.c: Oops, merge missed release of odbc object + + * apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006) + | 2 lines Failing to trap -1 error from mmap causes segfault + (Issue 8385) ........ + +2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp + + * main/frame.c, /: Merged revisions 47859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 + lines Don't forget to byte swap if we are exiting the smoother + feed early. (issue #8287 reported by arturs) ........ + + * /: Blocked revisions 47855 via svnmerge ........ r47855 | file | + 2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines Free + history items at the end of use of the temporary SIP pvt + structure. (issue #8383 reported by benh) ........ + + * main/rtp.c: Only remove/destroy the RTCP I/O item if it exists. + + * .cleancount, apps/app_dial.c, apps/app_directed_pickup.c, + include/asterisk/channel.h: Use a separate variable in the + channel structure to store the context that the channel was + dialed from. (issue #8382 reported by jiddings) + +2006-11-20 11:45 +0000 [r47843-47845] Olle Johansson + + * configs/sip.conf.sample: Explain properly how videosupport works. + Committ from Asterisk Video Task Force meeting in Paris! + + * /, channels/chan_sip.c: Make sure we destroy scheduled items and + not use them ever again after destruction (rizzo) + +2006-11-18 17:59 +0000 [r47823] Luigi Rizzo + + * channels/chan_sip.c: fix bug 7450 - Parsing fails if From header + contains angle brackets (the bug was only in a corner case where + the < was right after the opening quote, and the fix is trivial). + +2006-11-16 23:19 +0000 [r47781-47782] Jason Parker + + * apps/app_db.c, apps/app_dial.c: Fix a couple of typos. Initially + pointed out by mrobinson. + + * /: Blocked revisions 47780 via svnmerge ........ r47780 | qwell | + 2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a + couple of typos in applications.. Initially spotted by mrobinson. + ........ + +2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming + + * /, doc/billing.txt: update documentation regarding IAX2 transfers + and CDRs Merged revisions 47776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006) + | 2 lines update clearly wrong documentation regarding cdr_custom + ........ + +2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp + + * channels/chan_sip.c: Compare technology using the pointers + instead of a straight comparison based on name. (issue #8228 + reported by dean bath) + + * /: Blocked revisions 47761 via svnmerge ........ r47761 | file | + 2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines Look for + the header file specifically in all cases, not just the existence + of the directory. (issue #8358 reported by mrness) ........ + +2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming + + * configure, configure.ac: check for pre-1.4 versions of Zaptel and + abort the configure script if found with an appropriate error + message + +2006-11-16 19:24 +0000 [r47755] Olle Johansson + + * channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD + notification optional, in order to avoid a lot of extra database + lookups for all those realtime users out there. + +2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp + + * channels/chan_local.c, /: Merged revisions 47750 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov + 2006) | 2 lines Because of the way chan_local is written we + should be extra careful and make sure our callback functions have + a tech_pvt. (issue #8275 reported by mflorell) ........ + + * apps/app_meetme.c: Don't unreference the SLA object if there is + no SLA object in the devicestate callback. (issue #8354 reported + by loloski) + +2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson + + * /, channels/chan_sip.c: Don't fixup if there's nothing to fixup + + * UPGRADE.txt: Warn users about change in canreinvite + + * channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never + authenticated (according to the RFC) - Update docs on + canreinvite. "nonat" is the recommended setting for most users + with phones behind a NAT. + +2006-11-15 22:31 +0000 [r47712] Joshua Colp + + * channels/chan_local.c, /: Merged revisions 47711 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov + 2006) | 2 lines Make sure that the pvt structure exists before + trying to do fixup on Local channels. (issue #7937 reported by + mada123, fix by alamantia with mods by me) ........ + +2006-11-15 21:56 +0000 [r47709] Tilghman Lesher + + * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL + +2006-11-15 21:33 +0000 [r47707] Joshua Colp + + * main/channel.c: We need to ensure timelimit stuff is included as + well so warnings get played. (issue #8050 reported by KNK) + +2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming + + * main/file.c: don't try to call fclose() if fopen() failed + +2006-11-15 20:31 +0000 [r47698] Olle Johansson + + * channels/chan_sip.c: - Improve SIP history - Never send reply to + ACK (again...) + +2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming + + * apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006) + | 4 lines ensure that message duration is included in email + notifications for forwarded messages (BE-96, fix by me after + corydon used his clue-bat on me) ensure that duration in the + message metadata is updated if prepending is done during + forwarding (related to BE-96) remove prototype for API call that + does not exist ........ + + * main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 + Nov 2006) | 2 lines clear the category's variable tail pointer as + well when variables are detached from it ........ r47688 | + kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 + lines when appending a list of variable to a category, ensure the + tail pointer points to the last variable in the list ........ + r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) + | 2 lines when re-writing the config file, don't repeat the path + if it hasn't changed ........ + + * main/config.c, /: Merged revisions 47682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006) + | 2 lines ouch... don't use printf, use ast_log/ast_verbose + ........ + +2006-11-15 17:46 +0000 [r47672] Luigi Rizzo + + * main/cli.c: fix longest match search in find_cli. Trunk already + fixed. 1.2 not affected (well, i have no idea, the code is + totally different there). + +2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson + + * /, channels/chan_sip.c: Send error message when we can't allocate + SIP dialog, possibly due to limitation of file descriptors. + (imported from 1.2) + +2006-11-15 04:45 +0000 [r47645] Joshua Colp + + * main/rtp.c: If NAT detection is turned on or already detected + then say NAT is active when setting the remote RTP peer when + doing early bridging. (issue #8365 reported by marcelbarbulescu) + +2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming + + * main/term.c: more formatting cleanup, and avoid running off the + end of the string + +2006-11-15 00:14 +0000 [r47639] Joshua Colp + + * main/rtp.c: Turn notice about unknown RTCP packet type into a + debug message instead. + +2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming + + * channels/misdn/isdn_lib.c: silence compiler warning on 64-bit + platforms (this variable is an 'int' anyway, comparing it to + 'signed long' is not useful) + +2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp + + * apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2 + lines Update copyright information in the ADSI logo blob. + ........ + + * channels/chan_sip.c: Only keep the video RTP structure around if + 1. Video support is enabled and 2. A video codec is enabled on + the dialog + + * funcs/func_uri.c: Small documentation clarification for + URIENCODE. (issue #8294 reported by salaud) + +2006-11-14 18:54 +0000 [r47621] Tilghman Lesher + + * apps/app_voicemail.c: Conversion of res_odbc API to include ast_ + prefix did not completely transition app_voicemail when + ODBC_STORAGE is used (reported on IRC by caio1982, not in + bugtracker) + +2006-11-14 16:45 +0000 [r47617] Joshua Colp + + * apps/app_amd.c: Use LOG_DEBUG to print out the indication that + app_amd is using default settings instead of using LOG_NOTICE. + This stops needless logging of this information under normal + circumstances. (issue #8361 reported by Seb7) + +2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson + + * channels/chan_sip.c: Update documentation to fit the + implementation... + + * /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in + retransmission system if it's an OPTION packet from peerpoke + +2006-11-13 21:28 +0000 [r47584] Joshua Colp + + * /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2 + lines Initialize global pointers for connection and result to + NULL. (issue #8356 reported by james) ........ + +2006-11-13 20:20 +0000 [r47581] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) + | 2 lines Having more than 255 old messages caused corruption in + the new/old count ........ + +2006-11-13 19:15 +0000 [r47576] Steve Murphy + + * main/config.c: This solves bug 8342, whereby a crash occurs under + certain circumstances while reading a config file with comments-- + a call to CB_ADD shouldn't happen if withcomments is zero + +2006-11-13 19:11 +0000 [r47573] Tilghman Lesher + + * main/cli.c, channels/chan_sip.c: Re-enable old deprecated + commands + +2006-11-13 19:10 +0000 [r47572] Olle Johansson + + * /, channels/chan_sip.c: - Don't reply to INVITE already replied + to when we get BYE - Declare errmsg as int. Oops. + +2006-11-13 18:18 +0000 [r47564] Steve Murphy + + * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing + the messed if, but we all forgot to update the regressions. Until + now. + +2006-11-13 17:13 +0000 [r47553] Steve Murphy + + * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being + found... just confuses users + +2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp + + * /, apps/app_sms.c: Merged revisions 47549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2 + lines When sending an SMS with a user data header properly set + the UDH flag in the first byte. (issue #8347 reported by + hoffmeis) ........ + + * main/cli.c: Free full command string upon unregistering of CLI + command. Backported from revision 47536 from rizzo. + +2006-11-13 16:00 +0000 [r47540] Olle Johansson + + * channels/chan_sip.c: Only produce error message about sip history + once + +2006-11-13 05:48 +0000 [r47527] Russell Bryant + + * configure, acinclude.m4: AC_PROG_SED is included in autoconf + 2.60, but apparently it is not included in 2.59. So, to maintain + compatability with 2.59 since it is a small change, copy this + macro into acinclude.m4 and rename it to AST_PROG_SED. (issue + #8345) + +2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher + + * res/res_odbc.c, /: Merged revisions 47525 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006) + | 2 lines If the execute fails a second time, make sure that we + don't pass back a stale handle ........ + + * channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006) + | 2 lines Don't play dialtone if the seizing the channel fails + (Bug 7754) ........ + +2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson + + * channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks + DEA!!!) + + * channels/chan_sip.c: Part of issue 8078 - parse even if udptl is + UDPTL in sdp... + + * channels/chan_sip.c: - Don't destroy SIP dialog because of a + failed T.38 re-invite. Wait for a bye. Final response to a + re-invite does not mean that the session dies, only that the + re-invite fails. - Keep RTP active during processing of T.38 + re-invite. If the re-invite fails, RTP needs to remain as before + the re-invite. Issue 8338 - darren1713. Please test. + + * channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp + -Add some comments to t.38 code + +2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) | + 4 lines Only do the check to determine whether the channel + calling this function is an IAX2 channel when getting the IP + address using the special argument, CURRENTCHANNEL. (issue #8341, + jcovert) ........ + + * Makefile: Add the target "menuconfig" as an alias for the + "menuselect" target. This is just a favor to users so that if you + accidentally type "make menuconfig" instead of "make menuselect", + it still works. (inspired by a comment on IRC from wangster + calling me an "especially devious asterisk developer" for having + it be menuselect instead of menuconfig. :) ) + + * main/term.c: Tweak the formatting of this new function to better + conform to coding guidelines. + +2006-11-11 02:04 +0000 [r47490] Matt O'Gorman + + * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo + safe output! + +2006-11-10 22:23 +0000 [r47480] Matt Frederickson + + * channels/chan_zap.c: Make sure we don't use 32 bits when we only + need one bit. + +2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson + + * channels/chan_sip.c: ...and make sure that the dialog is + destroyed, even if we don't get any answer on the bye... This is + the channel that remains dead after the SIP transfer + + * channels/chan_sip.c: Add debug output while trying to trace bug + in bug report + + * channels/chan_sip.c: Make sure we destroy dialog... + + * /, channels/chan_sip.c: Small cleanup of handle_request_invite() + - imported from 1.2 with changes + +2006-11-10 19:47 +0000 [r47462] Matt Frederickson + + * channels/chan_zap.c: Fix for #7321. Be able to explicitly hide + callerid name for switches that bork on it. + +2006-11-10 18:56 +0000 [r47454] Olle Johansson + + * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart + SDP (alphaque) + +2006-11-10 17:13 +0000 [r47444] Luigi Rizzo + + * build_tools/prep_moduledeps: grep -m is not available on BSD, so + use head -1 instead + +2006-11-10 16:53 +0000 [r47437] Joshua Colp + + * apps/app_chanspy.c: Only split up extension and context if a + value exists. (issue #8332 reported by loloski) + +2006-11-10 16:51 +0000 [r47436] Tilghman Lesher + + * channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c, + channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_iax2.c: Discussion of these CLI changes resulted in + more consistency (Bug 8236) + +2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming + + * apps/app_queue.c: if adding a queue member is LOG_NOTICE, then + removing them should be LOG_NOTICE, not LOG_DEBUG + + * apps/app_queue.c: reflect addition/removal of dynamic queue + members in queue_log, so that people using dialplan replacement + for AgentCallbackLogin can still track login/logout (issue #7736, + reported/patched by whoiswes but this commit was written by me + and covers all three paths for AQM/RQM) + +2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson + + * channels/chan_sip.c: Rip out half implementation of 491 response + support, since it wasn't implemented properly and caused memory + leaks in the case of us getting 491's, which Asterisk actually + sends... Since it is a bit too complicated to fix this, I'll rip + it out of 1.4 and put it on the to-do-list for future releases. + Now, we handle this as congestion, which it really is. Issue + #8331 + + * channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD. + Thanks fenlander! + +2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp + + * channels/chan_h323.c: Fix building of chan_h323 by completeing + some structure definitions. (issue #8327 reported by Mithraen) + + * apps/app_voicemail.c: Do conversion in a more easier to read and + working way for \r, \n, and \t. (issue #8324 reported by + johnlange) + +2006-11-09 21:26 +0000 [r47391] Russell Bryant + + * apps/app_voicemail.c, channels/chan_zap.c, + build_tools/prep_moduledeps: Work around an issue that caused + menuselect to display a bogus description for app_voicemail and + chan_zap. These modules use some preprocessor directives to + determine what it will report to Asterisk as its description. + However, the way we extract this information from the source + files for menuselect is not smart enough to figure this out. + (issue #8326, #8328) + +2006-11-09 16:53 +0000 [r47380] Joshua Colp + + * channels/chan_phone.c, /: Merged revisions 47379 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov + 2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and + higher as, well, it's apparently going to be removed. This should + make all you FC6 fans happy as your Asterisk will now build + without any mods. ........ + +2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant + + * main/cli.c: fix tab completion for "core debug channel" and "core + no debug channel" + + * main/cli.c: Fix "core show channel". Also, fix tab completion for + both "core show channel" and "core show channels". + + * main/cli.c: Fix "core debug channel ". I guess someone + needs to go through and audit every CLI command that changed + number of arguments ... + + * main/asterisk.c: revert the previous change, which actually + modified the deprecated command, "show profile". Now, actually + apply the change to "core show profile". + + * main/asterisk.c: Fix argument parsing for the "core show profile" + CLI command (fixed by rizzo in his branch, team/rizzo/astobj2) + + * main/cli.c: Fix another CLI command, "core show uptime" ... + (issue #8323, reported by johnlange, fixed by myself) + + * main/asterisk.c: fix "core show version" to reflect the new + number of arguments for this CLI command (issue #8316, kshumard) + +2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy + + * main/channel.c: This update fixes 7531 + + * channels/chan_skinny.c: Committed in behalf of 8190. + +2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming + + * main/frame.c: the battle over CLI command formats has broken + stuff... + + * channels/chan_sip.c: add simple fix for SDP to report proper + sample rate for G.722 media sessions + +2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant + + * utils/streamplayer.c: I occasionally get email from users that + are trying to figure out what this does, or due to some + misunderstanding as to what it is supposed to do, can't get it to + work. So, I have added some text here to hopefully explain what + this application does and does not do. + + * channels/chan_gtalk.c: Make this module build again + + * configure, configure.ac, acinclude.m4: Copy the macros from + libtool.m4 to our own acinclude.m4 such that libtool is no longer + required to be installed to be able to generated the configure + script. + +2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson + + * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo) + +2006-11-07 23:46 +0000 [r47303] Steve Murphy + + * channels/chan_oss.c, main/channel.c, channels/chan_phone.c, + channels/chan_misdn.c, channels/chan_skinny.c, + channels/chan_features.c, channels/chan_h323.c, + channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c, + include/asterisk/stringfields.h, apps/app_voicemail.c, + main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c, + channels/chan_zap.c, channels/chan_sip.c, res/res_features.c, + channels/chan_agent.c, main/utils.c, include/asterisk/channel.h, + channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to + solve the problem in bug 7506. It's a lot of rework to solve a + fairly small problem... such is life. + +2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp + + * channels/chan_local.c: Make MOH work as it did before in + chan_local, without this then it can go funky when transfers and + MOH are involved. (issue #7671 reported by jmls) + +2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming + + * configs/musiconhold.conf.sample: clean up sample config, and make + native file playback the more obvious default choice + +2006-11-07 18:38 +0000 [r47275] Matt O'Gorman + + * apps/app_voicemail.c: large overhaul to voicemail imap support. + Allows support for more imap servers, also a better + implementation of several parts of the original work. patch + provided by 8033 with major upgrades. + +2006-11-07 17:30 +0000 [r47268] Olle Johansson + + * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of + continue. + +2006-11-07 13:13 +0000 [r47250] Olle Johansson + + * /, channels/chan_sip.c: Fixing the attack shield so it doesn't + produce attacks... Issue 8265 - never reply to an ACK + +2006-11-07 01:25 +0000 [r47239] Russell Bryant + + * /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 + Nov 2006) | 5 lines If random order is enabled for files mode + music on hold, set a random initial position, instead of always + starting at the first file, and doing the random operation only + when switching to the next file. (bug reported by John Lange on + the asterisk-dev mailing list) ........ + +2006-11-04 18:32 +0000 [r47199] Olle Johansson + + * channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and + transfer from "john" Thank you! + +2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant + + * main/cli.c: Fix another bug in "core set debug" ... + + * main/asterisk.c, main/cli.c: Really fix the "core set debug" and + "core set verbose" CLI commands. + + * main/cli.c: fix the "atleast" option to the "core set verbose" + and "core set debug" CLI commands + +2006-11-03 23:17 +0000 [r47176] Steve Murphy + + * channels/chan_sip.c: This fix introduced via bug 8233 + +2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo + + * bootstrap.sh: align bootstrap.sh with the version in trunk (needs + to be blocked as it is already in trunk) + + * configure.ac: add proper environment vars to detect modules on + freebsd. (already applied to trunk so it needs to be blocked + there) + +2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher + + * main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c, + channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More + changes making the CLI more consistent with "category verb + arguments" (continuation of issue 8236) + + * main/config.c, main/cli.c, main/channel.c, main/manager.c, + channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c, + main/http.c, main/file.c, main/logger.c, main/image.c, + res/res_indications.c, main/asterisk.c, res/res_odbc.c, + channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, + channels/chan_local.c, main/frame.c, channels/chan_sip.c, + res/res_features.c, channels/chan_agent.c, res/res_crypto.c, + res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c: + Reverse change of "show" to "list" and make several other + commands more consistent with "category verb arguments" + +2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson + + * channels/chan_sip.c: Move check for codec translation to + sip_call() instead of in add_sdp. No one bothers with the result + of add_sdp anyway... Yet... + + * channels/chan_sip.c: Disable code for T38 over TCP and RTP since + there's no trace of actual functionality for it :-) + +2006-11-02 17:49 +0000 [r46965] Russell Bryant + + * /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02 + Nov 2006) | 3 lines ignore files in a music on hold directory + that begin with '.' (issue #8249, cboie) ........ + +2006-11-02 17:17 +0000 [r46963] Nadi Sarrar + + * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix + +2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming + + * channels/chan_sip.c: don't send INVITE when we have determined + that we can't offer any audio formats due to lack of transcoding + support (or incorrect configuration) + +2006-11-02 16:06 +0000 [r46930] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 + lines Repeat after me oej: I will at least make sure my code + compiles before I commit it. ........ + +2006-11-02 15:24 +0000 [r46901] Olle Johansson + + * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2) + +2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant + + * /, main/callerid.c: Add the missing call to free described in + issue #8268. Also, add a bunch of missing calls to free in + callerid_feed_jp(). + + * main/say.c: fix saying one hundred and two hundred in hebrew + (issue #7810, eldadran) + + * Makefile, configure, codecs/gsm/Makefile, configure.ac, + build_tools/strip_nonapi, makeopts.in: Fixes for + cross-compilation on mips (issue #8058, ywalther, with some + modifications) + + * aclocal.m4, build_tools/menuselect-deps.in, configure, + build_tools/embed_modules.xml, configure.ac: Add a check in the + configure script to determine whether ld is GNU ld or not. This + is needed because module embedding only works for gnu ld. GNU ld + is now listed as a dependency for all of the module embedding + options in menuselect. (issue #8143) + +2006-11-01 20:35 +0000 [r46822] Matt O'Gorman + + * channels/chan_gtalk.c: bind address support from bug 8164 + +2006-11-01 19:49 +0000 [r46802] Steve Murphy + + * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to + accept longer strings or mass confusion and a lot of lost time is + the result + +2006-11-01 18:39 +0000 [r46780] Joshua Colp + + * main/Makefile: Force poll() emulation for Darwin to always be on. + It's too broken to consider being used. This resolves the console + issue OSX users have been seeing. I would have liked to autoconf + this but I haven't been able to come up with a test case that + works. Que sera. + +2006-11-01 18:26 +0000 [r46778] Russell Bryant + + * res/res_monitor.c, /: Merged revisions 46776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | + 9 lines soxmix and Asterisk expect different file extensions for + certain formats. This was already handled for the wav49 format. + However, it was not handled for ulaw and alaw. I fixed this in + such a way that using the alternate extensions for ulaw and alaw + will only happen if we know we're calling soxmix, and not a + custom script defined using the MONITOR_EXEC variable. The wav49 + processing was left alone so that external scripts will see no + behavior change. (issue #7550, reported by mnicholson, proposed + patch by junky, committed fix is a bit different) ........ + +2006-11-01 18:21 +0000 [r46775] Joshua Colp + + * channels/chan_iax2.c: It's another round of chan_iax2 fixes! + Should hopefully fix the deadlock issues people have been + reporting. IAXtel now has qualify turned on for 800 peers and it + is handling it fine. + +2006-11-01 17:48 +0000 [r46760] Steve Murphy + + * main/config.c: Cleanups suggested by Russell. + +2006-11-01 16:39 +0000 [r46744] Russell Bryant + + * channels/chan_zap.c: Prevent an infinite loop when config + processing gets to a jitterbuffer option + +2006-10-31 22:02 +0000 [r46716] Jason Parker + + * main/translate.c: Fix "core show translation" output. Issue + #8243, patch by Damin. + +2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming + + * include/asterisk/translate.h, main/translate.c: add an API so + that translators can activate/deactivate themselves when needed + + * include/asterisk/translate.h, main/translate.c: revert changes + that were the wrong way to address this... proper fix coming + + * main/translate.c: let's set the seen flag early enough to + actually make a difference... + + * include/asterisk/translate.h, main/translate.c: don't re-do setup + operations for translators that can dynamically register + themselves + +2006-10-31 15:49 +0000 [r46663] Tilghman Lesher + + * /: Blocked revisions 46662 via svnmerge ........ r46662 | + tilghman | 2006-10-31 09:46:04 -0600 (Tue, 31 Oct 2006) | 3 lines + Move thread-unsafe initializer to the module loading code; add + the corresponding function to the module unload to fix a memory + leak. ........ + +2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson + + * main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue + #8089 - Fix the ENUM support (picking one record by number). + Thanks otmar! + + * /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport + when we're supposed to support ;rport. Issue #7473. + + * /, channels/chan_sip.c: If peer fails ACL check, fail peer at + REGISTER + + * channels/chan_sip.c: Fix T38 too. Thanks, tgrman ! + +2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant + + * contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the + boot process to ensure it starts after stuff like MySQL (issue + #8253, Alric) + + * /, main/utils.c: Merged revisions 46560 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) | + 3 lines When handling the case where the hostname is just an IPV4 + numeric address, be sure to set the address type. (issue #8247, + alexr) ........ + + * /, res/res_agi.c: Merged revisions 46557 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) | + 3 lines fix some copy/paste bugs in the checking of arguments for + the "control stream file" AGI command (issue #8255, mnicholson) + ........ + + * main/translate.c: Add a small tweak to the code that checks to + see whether destination formats are translatable based on the + source format. If we have already determined that there is no + translation path in one direction, don't bother checking the + other direction. + +2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming + + * main/translate.c: when unregistering a translator, don't rebuild + the translation matrix unless needed when filtering formats out + of an offer, ensure we check for translation ability in both + directions + + * include/asterisk/linkedlists.h: ensure that items removed from a + list are always unlinked from the list (next pointer set to NULL) + +2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp + + * configure, configure.ac: Don't explicitly link in crypt as it is + not used on some platforms. + + * channels/chan_iax2.c: We need to lock the pvt structure during + retransmission as another worker thread may be doing something as + well. + +2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson + + * main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h, + include/asterisk/doxyref.h, channels/chan_sip.c, + main/ast_expr2f.c, include/asterisk/module.h, + formats/format_ogg_vorbis.c, main/app.c, + include/asterisk/channel.h, include/asterisk/lock.h, + include/asterisk/frame.h: Issue #8246 - Doxygen fixes from + kshumard. An extra big thankyou is given to everyone that + contributes to doxygen! THANK YOU! + + * main/rtp.c, /: Bind RTCP to the same IP as RTP + + * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302 + redirects (imported from 1.2) + + * /, channels/chan_sip.c: Issue #7608 - Notifications sent with + wrong content-type (imported from 1.2, modified) + + * channels/chan_sip.c, CHANGES: Backport of patch for #7828 that + was reported for trunk, but obviously exists in 1.4 too. + + * channels/chan_sip.c: Restoring the old logic, since working + around it and fixing it seemed too complicated. - The + SIP_OUTGOING flag indicates the direction of the last transaction + in the dialog. - The initreq stores the last request in the + dialog, the request that opened the latest transaction. Please + now retry all the 1.4 bug reports with mixed to/from headers, + tags etc in ACK, BYE, CANCEL. Thanks! + + * channels/chan_sip.c: Accepting a message twice may be + misinterpreted... + + * channels/chan_sip.c: - 183 is not reliable message... - Error + should not have SDP + +2006-10-28 16:37 +0000 [r46377] Joshua Colp + + * utils/Makefile: Don't build muted on OpenBSD, it is not + supported. + +2006-10-27 19:03 +0000 [r46370] Russell Bryant + + * channels/chan_zap.c: move the copy of the default settings to the + global settings back out of process_zap, so that they aren't + overwritten when process_zap is called multiple times + +2006-10-27 18:29 +0000 [r46367] Olle Johansson + + * contrib/asterisk-ng-doxygen: Put some doxygen pressure on + Christian :-) + +2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant + + * main/asterisk.c, res/res_agi.c, apps/app_externalivr.c, + res/res_musiconhold.c: We should always be using _exit() after a + fork() or vfork() instead of exit(). This is because exit() does + some extra cleanup which in some implementations of vfork(), for + example, can actually modify the state of the parent process, + causing very weird bugs or crashes. (issue #7971, Nick Gavrikov) + + * /: Blocked revisions 46361 via svnmerge ........ r46361 | russell + | 2006-10-27 12:36:07 -0500 (Fri, 27 Oct 2006) | 5 lines We + should always be using _exit() after a fork() or vfork() instead + of exit(). This is because exit() does some extra cleanup which + in some implementations of vfork(), for example, can actually + modify the state of the parent process, causing very weird bugs + or crashes. (issue #7971, Nick Gavrikov) ........ + + * channels/chan_zap.c: Instead of iterating all of the options once + to look for jitterbuffer options, and then again for everything + else, move the processing of jitterbuffer options into the main + loop so that there are no erroneous messages about ignoring + unknown options. (issue #8226) + +2006-10-27 10:03 +0000 [r46351-46353] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 46350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | + 1 line fixed a bug which caused chan_misdn to try to allocate 2 + times the same channel on high load, which then caused + instability of mISDN. removed a useless function from isdn_lib.c + ........ + + * channels/misdn_config.c: fixed not compile issue, which was just + introduced + + * channels/misdn_config.c, channels/chan_misdn.c, /, + channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: + Merged revisions 46176 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | + 1 line added nttimeout option to configure wether we disconnect + calls on NT timeouts or not during an overlapdial session + ........ + +2006-10-26 17:57 +0000 [r46335-46340] Jason Parker + + * /, contrib/scripts/astgenkey.8: Merged revisions 46337 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2 + lines oops - somebody forgot to change this - long ago, probably. + ........ + + * CHANGES: grammar check + +2006-10-26 16:38 +0000 [r46331] Olle Johansson + + * CHANGES: Corrections to changes (Multiparking is not included) + +2006-10-26 16:31 +0000 [r46329] Russell Bryant + + * main/translate.c: - If the source has no audio or no video + portion, do not call powerof() to get the format index. - Don't + run through the audio and video loops if there is no audio or + video portion of the source If 0 is passed to powerof, it will + return -1. This value of -1 was then being used as an array index + in these loops, which caused a crash on some systems. Other than + this issue, this code works as we expected it to. If a format is + not in the source, and we have to translation path to it, it is + not offered in the list of acceptable destination formats. (fixes + issue #8231) + +2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming + + * CHANGES: update to reflect G.722 addition + +2006-10-26 04:18 +0000 [r46298] Russell Bryant + + * doc/backtrace.txt: update backtrace documentation to reflect + changes in 1.4 (issue #8230, kshumard) + +2006-10-26 01:37 +0000 [r46287] Mark Spencer + + * main/config.c, main/manager.c: Fix config comment code + preservation code (thanks murf!) + +2006-10-25 20:14 +0000 [r46276] Olle Johansson + + * channels/chan_sip.c: Old todo note - Don't add Contact header on + BYE and Cancel + +2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant + + * configure.ac: fix error output when checking for openh323 to + refer to openh323 instead of pwlib (issue #8222, misaksen) + +2006-10-25 19:16 +0000 [r46252] Olle Johansson + + * channels/chan_sip.c: Somewhat ugly code to try to fix issue + #7608. Since the problem was not very well defined, the fix is a + bit fuzzy too... Thanks to Luigi for accidentally spotting the + possible problem! + +2006-10-25 19:08 +0000 [r46249] Russell Bryant + + * apps/app_queue.c: update warning message to include "agi" option + (issue #8225, jmls) + +2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming + + * sounds/Makefile: use 1.4.3 extra sounds with corrected silence + files + + * sounds/sounds.xml, sounds/Makefile: add support for prebuilt + G.722 prompts and music on hold files + +2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson + + * channels/chan_sip.c: show settings doesn't produce a list of + similar objects, it should stay a "show" + +2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming + + * main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c, + channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c, + pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c, + main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c, + cdr/cdr_custom.c, channels/chan_mgcp.c, + apps/app_parkandannounce.c, apps/app_voicemail.c, + channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c, + res/res_adsi.c, main/utils.c, apps/app_ices.c, + pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c, + apps/app_getcpeid.c: apparently developers are still not aware + that they should be use ast_copy_string instead of strncpy... fix + up many more users, and fix some bugs in the process + +2006-10-25 04:58 +0000 [r46165] Tilghman Lesher + + * main/pbx.c: WaitExten truncates decimals of times to wait, + instead of accepting them (Bug 8208) + +2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming + + * main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c, + channels/chan_h323.c, channels/chan_iax2.c, + include/asterisk/frame.h: add passthrough and file format support + for G.722 16KHz audio (issue #5084, original patch by andrew, + updated by mithraen) + + * channels/chan_sip.c, main/translate.c: code zone experiment: + don't offer formats in the outbound INVITE that aren't either + passthrough or translatable + + * main/translate.c: if multiple translators are registered for the + same source/dest combination, ensure that the lowest-cost one is + always inserted earlier in the list + +2006-10-24 20:30 +0000 [r46142] Mark Spencer + + * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628, + #8147) + +2006-10-24 19:29 +0000 [r46130] Joshua Colp + + * channels/chan_iax2.c: We need to initialize our scheduler pthread + condition... yes. + +2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo + + * main/http.c: merge 45152 don't leak descriptors in http.c + + * channels/chan_sip.c: merge 45966 refer_to_domain potentially + containing options + + * channels/chan_sip.c: merge 46026 improper checks on get_header() + return values + + * channels/chan_sip.c: merge 46045 prevent NULL args to + ast_strdupa() in chan_sip.c + +2006-10-24 05:23 +0000 [r46093] Russell Bryant + + * Makefile: Restore the ability to remove the firmware directory + without causing the installation to fail (issue #8111) + +2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming + + * main/translate.c: ensure that the translation matrix is properly + lock-protected every place it is used + + * include/asterisk/translate.h, main/translate.c: add an API call + to allow channel drivers to determine which media formats are + compatible (passthrough or transcode) with the format an existing + channel is already using + + * doc/imapstorage.txt: simplify and correct voicemail IMAP storage + build instructions + +2006-10-24 03:01 +0000 [r46078] Tilghman Lesher + + * main/channel.c: Pass through a frame if we don't know what it is, + rather than trying to pass a NULL, which will segfault a channel + driver (Bug 8149) + +2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant + + * utils/muted.c, utils/ael_main.c: In muted.c, check the return + value of strdup. In ael_main.c, check the return value of calloc. + (issue #8157) In passing fix a few minor bugs in ael_main.c. The + last argument to strncpy() was a hard-coded 100, where it should + have been 99. I changed this to use sizeof() - 1. + + * apps/app_meetme.c: Fix the descriptions of some of the + MeetMeAdmin options (issue #8098, mflorell) + + * res/res_jabber.c: don't crash when an incoming message has no + "from" (issue #8205, jmls) + +2006-10-23 00:27 +0000 [r45928] Joshua Colp + + * /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2 + lines Don't leak memory mmmk? ........ + +2006-10-22 21:44 +0000 [r45916] Christian Richter + + * channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21 + Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and + couldn't be initialized it would cause a segfault after 'reload'. + Reported by Drew/Matt thx. ........ + +2006-10-21 18:49 +0000 [r45818] Russell Bryant + + * res/res_monitor.c: Add a couple missing unregistrations of + manager actions and remove duplicate unregistrations of + applications. (issue #8194, jmls) + +2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp + + * main/loader.c: Don't use promotion on Darwin because it doesn't + seem to work quite right in all cases, this should solve the + unresolved symbol issue people have been seeing. + + * Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get + installed in the proper location (reported on asterisk-dev + mailing list) + +2006-10-20 07:44 +0000 [r45741] Olle Johansson + + * channels/chan_sip.c: Let's understand SIP: - REFER can create + dialog, Asterisk does not support it yet - NOTIFY can create + dialog in Asterisk's implementation (voicemail) even though we + don't support the server side of it. In this case, the standard + is a side issue ;-) - Added extened functionality for unsupported + methods (PING, PUBLISH) so we don't create PVT's for those + either. Russellb needs to judge what to do with this in 1.2, but + I think the current implementation n 1.2 is a bug since we're + sending bad replies to NOTIFY and REFER outside of dialogs + +2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp + + * res/res_jabber.c: Let's remember to unregister JabberStatus too + (issue #8184 reported by jmls) + + * /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct + 2006) | 2 lines Respect language selection when seeing if the + file exists (issue #8178 reported by mnicholson) ........ + + * channels/chan_sip.c: If the jitterbuffer is forced on then we + can't partially bridge (reported by wangster on #asterisk-dev) + +2006-10-19 00:59 +0000 [r45622] Russell Bryant + + * channels/chan_sip.c: Don't leak the actual thread-specific + sip_pvt struct + +2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming + + * channels/chan_sip.c: don't leak memory when a chan_sip thread is + destroyed that has a thread-local temp_pvt allocated + +2006-10-18 21:03 +0000 [r45595] Joshua Colp + + * main/asterisk.c: Don't modify things if we are using vfork as + this is very bad and may cause unexpected behavior (issue #7970 + reported by Nick Gavrikov) + +2006-10-18 11:54 +0000 [r45517] Olle Johansson + + * channels/chan_sip.c: remove duplicate declarations + +2006-10-18 04:09 +0000 [r45464] Luigi Rizzo + + * main/http.c: merge from trunk: move ast_variables_destroy() to a + better place in handle_uri() to avoid leaking memory on non + existing files. + +2006-10-18 03:02 +0000 [r45452] Joshua Colp + + * main/rtp.c: Don't segfault if you're using a channel driver that + doesn't turn RTCP on + +2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant + + * main/channel.c: Don't attempt to access private data members of + the pthread_mutex_t object, because this does not work on all + linux systems. Instead, just access the reentrancy field in the + ast_mutex_info struct when DEBUG_THREADS is enabled. If + DEBUG_CHANNEL_LOCKS is enabled, the developer probably has + DEBUG_THREADS on as well. (issue #8139, me) + + * configs/sip_notify.conf.sample: update entry to reboot a snom + phone (issue #7850, pnlarsson) + +2006-10-17 Kevin P. Fleming + + * Asterisk 1.4.0-beta3 released. + +2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming + + * include/asterisk/stringfields.h, main/ast_expr2.c, + main/channel.c, channels/chan_sip.c, channels/chan_iax2.c: + optimize the 'quick response' code a bit more... no more malloc() + or memset() for each response expand stringfields API a bit to + allow reusing the stringfield pool on a structure when needed, + and remove some unnecessary code when the structure was being + freed + +2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp + + * channels/chan_sip.c: Don't create a "real" pvt structure for + requests that shouldn't be able to create one. Instead use a + temporary pvt and fill it with enough information so we can send + a reply. + +2006-10-17 17:39 +0000 [r45329] Olle Johansson + + * configs/sip.conf.sample: Adding information about Marks + direct-RTP hack to the docs... + +2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming + + * LICENSE: provide licensing language for IAXy firmware file + +2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp + + * apps/app_dial.c, apps/app_directed_pickup.c: Backport of new + directed pickup (BE-85). + +2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson + + * CREDITS: Adding Inotel to credits for SIP transfers. Thanks for + your support! + + * channels/chan_sip.c: Don't destroy dialog for unexpected REFER + response... + +2006-10-14 04:38 +0000 [r45143] Steve Murphy + + * funcs/func_rand.c: update the doc string for both AEL and + extensions.conf users. + +2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming + + * main/acl.c don't drop the entire permit/deny list when an attempt + is made to add an invalid entry (BE-92) + +2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp + + * res/res_speech.c: Clear the quiet flag too since we are + restarting a recognition again (reported on -dev by Stephan + Edelman) + + * res/res_speech.c: Check return value from engine in case of + failure (ie: out of licenses) (reported on -dev mailing list) + +2006-10-13 20:52 +0000 [r45103] Steve Murphy + + * pbx/ael/ael-test/ref.ael-vtest17 (added), + pbx/ael/ael-test/ael-vtest17/extensions.ael (added), + pbx/ael/ael-test/ael-vtest17 (added), + pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in + this release via these changes + +2006-10-13 19:19 +0000 [r45088] Christian Richter + + * channels/chan_misdn.c: avoiding warning, fixing potential bug + +2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp + + * codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c, + codecs/lpc10/decode.c, codecs/lpc10/dcbias.c, + codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c, + codecs/lpc10/difmag.c, codecs/lpc10/hp100.c, + codecs/lpc10/synths.c, codecs/lpc10/preemp.c, + codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c, + codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c, + codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c, + codecs/lpc10/lpcini.c, codecs/lpc10/random.c, + codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c, + codecs/lpc10/placea.c, codecs/lpc10/tbdm.c, + codecs/lpc10/analys.c, codecs/lpc10/onset.c, + codecs/lpc10/energy.c, codecs/lpc10/deemp.c, + codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c, + codecs/lpc10/median.c, codecs/lpc10/encode.c, + codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c, + codecs/lpc10/invert.c: And file said... let the compiler warnings + STOP! + + * apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136 + reported by mnicholson) + + * apps/app_playback.c: Move say.conf existence check to do_say + function since it is called from multiple places (issue #8144 + reported by kshumard) + +2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming + + * channels/chan_iax2.c: when sending a call to a peer, use the proper socket if + we have multiple bindings (reported on asterisk-dev) + +2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp + + * channels/chan_sip.c: Complete merging in RPID screen changes + (issue #8101 reported by hristo, patch by oej in revision 44757) + + * main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add + the background refresh item back into the scheduler if enabled + since it is deleted during reload. (issue #8142 reported by + p_lindheimer) + +2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/utils.c: use a configure script test for PMTU discovery + control instead of just assuming it's available on Linux + +2006-10-13 14:45 +0000 [r44994-45026] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some + echocandisable issues when bridged. this caused a kernel panic + sometimes.. also some minor formatting fixes + + * channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause + got a wrong isdn cause at RELEASE_COMPLETE + +2006-10-12 22:07 +0000 [r44992] Luigi Rizzo + + * channels/chan_sip.c: merge formatting and minor code + simplifications from trunk + +2006-10-12 20:34 +0000 [r44982] Matt O'Gorman + + * channels/chan_gtalk.c: fix for bug 7764. + +2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming + + * channels/chan_sip.c: we can only send one 'a=ptime' attribute per + media session, not one for each format + + * main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c, + main/utils.c: ensure that IAX2 and SIP sockets allow UDP + fragmentation when running on Linux (thanks to Brian Candler on + the asterisk-dev list for the tip) + +2006-10-12 16:56 +0000 [r44945] Russell Bryant + + * main/manager.c: fix a silly typo in a comment that I saw while + reading the commit list + +2006-10-12 16:08 +0000 [r44942] Joshua Colp + + * Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue + #8135 reported by ssokol) + +2006-10-12 12:55 +0000 [r44921] Nadi Sarrar + + * main/manager.c: append_event must be called while holding the + session lock + +2006-10-12 10:24 +0000 [r44911] Russell Bryant + + * res/res_jabber.c: change some debug output to use LOG_DEBUG + instead of verbose output + +2006-10-11 16:57 +0000 [r44888] Jason Parker + + * main/db1-ast/Makefile: These are already set by the parent + Makefile.. There is no need to have this here (it doesn't + actually work anyways). + +2006-10-11 09:18 +0000 [r44854] Christian Richter + + * channels/misdn/isdn_lib.c: removed warning because of missing + prototype declaration + +2006-10-10 19:23 +0000 [r44830] Olle Johansson + + * channels/chan_sip.c: Do not set default/global values in the + variable declaration, set it in reload_config() + +2006-10-10 17:21 +0000 [r44819] Joshua Colp + + * channels/chan_sip.c: Move some stuff around so that a NOTIFY + dialog won't hang around until the end of the world under certain + circumstances + +2006-10-10 16:44 +0000 [r44809] Paul Cadach + + * main/channel.c, funcs/func_channel.c, include/asterisk/channel.h: + CHANNEL() function sometime mix parameter and value + +2006-10-10 16:42 +0000 [r44808] Tilghman Lesher + + * funcs/func_logic.c: Lost of a bit of logic when this was + simplified between 1.2 and 1.4 (Bug 8117) + +2006-10-10 16:30 +0000 [r44806] Joshua Colp + + * channels/chan_sip.c: Bail out if we have no refer structure and + we get a refer response + +2006-10-10 16:21 +0000 [r44805] Luigi Rizzo + + * channels/chan_sip.c: more merge from trunk (comments and change a + static function name) + +2006-10-10 15:23 +0000 [r44788] Joshua Colp + + * channels/chan_sip.c: Only set DTMF information if an RTP + structure exists + +2006-10-10 13:50 +0000 [r44786] Christian Richter + + * channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added + support of dynamically enabling hdlc on bchannels + +2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo + + * channels/chan_sip.c: whitespace changes related to previous + commit + + * channels/chan_sip.c: merge a few code simplifications that have + gone into trunk during last week, to reduce differences between + the two branches and make porting fixes easier. + +2006-10-09 16:12 +0000 [r44764] Jason Parker + + * channels/chan_skinny.c: Fix a problem where phones that go + "missing" never got unregistered. Issue #8067, reported by pj, + patch by Anthony LaMantia (with minor whitespace modifications) + +2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp + + * channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid + the deadlock + + * channels/chan_iax2.c: Properly avoid a collision with iax2_hangup + (issue #8115 reported by vazir) + +2006-10-08 14:14 +0000 [r44746] Luigi Rizzo + + * channels/chan_sip.c: do not dereference p if we + know it is NULL + +2006-10-07 14:39 +0000 [r44684] Paul Cadach + + * channels/h323/ast_h323.cxx, channels/chan_h323.c, + channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate + caller's transfer capability too + +2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo + + * channels/chan_sip.c: put common code in a + function to avoid repetitions. + + * channels/chan_sip.c: remove hardwired usage of 5060, use + DEFAULT_SIP_PORT instead + + * channels/chan_sip.c: option_debug checking + before printing to debug channel. + + * channels/chan_sip.c: backport simplifications on sip_register, + usage of ast_set2_flag(), and fixes to the handling of failed + module loading. + + * channels/chan_sip.c: improve and document function + get_in_brackets(), introducing a helper function + find_closing_quote() of more general use. + +2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming + + * include/asterisk/linkedlists.h: ensure that mutex locks inside + list heads are initialized properly on platforms that require + constructor initialization (issue #8029, patch from timrobbins) + + * CHANGES: remove Jingle as per mog + +2006-10-06 21:08 +0000 [r44628] Joshua Colp + + * main/rtp.c: Remove the seqno check for RFC2833, the handler is + smart enough to not need it. + +2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming + + * CHANGES: various cleanups + +2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp + + * main/rtp.c: When the sequence number rolls over then reset the + recorded sequence number for DTMF (issue #8106 reported by + bungalow) + + * main/file.c: Even more frames to treat as though the remote side + disappeared (issue #8097 reported by eldadran) + +2006-10-06 15:59 +0000 [r44567] Luigi Rizzo + + * main/manager.c, main/http.c: make sure sockets are blocking when + they should be blocking. + +2006-10-06 12:53 +0000 [r44559-44563] Christian Richter + + * channels/chan_misdn.c: fixed segfault which happens during + hold/transfer action + + * channels/chan_misdn.c: if INFORMATION Message come with keypad + instead of called party number, we just use the keypad as called + party number. + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, + channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: + added the option 'reject_cause' to make it possible to set + the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, + which is automatically rejected because chan_misdn does not + support that kind of callwaiting. Therefore chan_misdn supports + now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc + now gets the info if the requested channel is incoming or + outgoing to make the 3. channel possible + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c: fixed the hold/retrieve/transfer issues, + removed a useless bc field, added setting of frame.delivery fields, + some minor code cleanups + +2006-10-05 19:57 +0000 [r44502] Joshua Colp + + * main/file.c: Treat busy control frames as hangup in the file streaming + core (issue #8097 reported by eldadran) + +2006-10-05 18:21 +0000 [r44488] Steve Murphy + + * pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang. + Many thanks to Doug! + +2006-10-05 18:01 +0000 [r44486] Joshua Colp + + * channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite + hanging by a thread if the other side is already setup with T.38 + +2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming + + * main/app.c: don't segfault when an argument without a close + parenthesis is found stop parsing as soon as that situation + occurs + +2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy + + * CHANGES: I put the accumulated changes from the commit logs and + inspection, into CHANGES. Hope everyone approves! + + * configs/muted.conf.sample, utils/muted.c: Hang on a minute, the + install process sticks muted.conf in /etc/asterisk, so that's + where muted should look for it, right? + +2006-10-05 02:40 +0000 [r44450] Joshua Colp + + * channels/chan_sip.c: Don't totally bail out if T.38 was + negotiated + +2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming + + * channels/chan_sip.c: fix Polycom presence notification again + +2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo + + * utils/Makefile: as far as i can tell astman only uses newt... + + * Makefile: put linker flags in ASTLDFLAGS where they belong + +2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming + + * channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE + requests add workaround for new Polycom firmware SUBSCRIBE + requests (bug is known to exist in 2.0.1 firmware) + + * include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually + work + +2006-10-04 19:57 +0000 [r44380] Steve Murphy + + * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c, + pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12, + pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, + pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, + pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, + pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8, + pbx/ael/ael-test/ael-test16/extensions.ael (added), + pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y, + pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, + pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, + pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the + problems reported in bug 8090 + +2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming + + * channels/chan_oss.c, main/cdr.c, channels/chan_phone.c, + main/manager.c, pbx/pbx_spool.c, res/res_smdi.c, + channels/chan_skinny.c, channels/chan_h323.c, main/http.c, + channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c, + main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c, + include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c, + channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c, + main/devicestate.c, main/utils.c, res/res_musiconhold.c, + channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update + thread creation code a bit reduce standard thread stack size + slightly to allow the pthreads library to allocate the stack+data + and not overflow a power-of-2 allocation in the kernel and waste + memory/address space add a new stack size for 'background' + threads (those that don't handle PBX calls) when LOW_MEMORY is + defined + +2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy + + * configs/muted.conf.sample: I've been meaning to add some + explanation about muted... here it is + + * configs/manager.conf.sample: CLI reverbification update to this + config file + + * apps/app_macro.c: In response to bug 7776, a Warning has been + added to the doc string for Macro(). + +2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming + + * main/asterisk.c, main/loader.c, main/term.c, Makefile, + include/asterisk.h: ensure that local include files are always + used avoid a duplicate function name (term_init()) + +2006-10-03 22:35 +0000 [r44312] Matt O'Gorman + + * channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing + client without resource. + +2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming + + * apps/app_queue.c: fix a logic error in my previous fix to the queue + reload code + +2006-10-03 18:42 +0000 [r44286] Paul Cadach + + * channels/h323/ast_h323.cxx: Change default presentation indicator + to "user provided not screened" if octet 3a missed in + CallingPartyNumber IE + +2006-10-03 18:35 +0000 [r44284] Joshua Colp + + * channels/chan_sip.c: Use VideoSupport instead so it is considered + a valid XML attribute name. (issue #8075 reported by renemendoza) + +2006-10-03 18:30 +0000 [r44283] Paul Cadach + + * channels/h323/ast_h323.cxx: Fix preparation of type and + presentation of calling number + +2006-10-03 00:01 +0000 [r44240] Matt O'Gorman + + * doc/jingle.txt, channels/chan_jingle.c (removed), + include/asterisk/jabber.h, configs/jingle.conf.sample (removed), + res/res_jabber.c: updated res_jabber for even better component + support, soon will be jep-0100 compliant. also removed + chan_jingle and infromed info from jingle.txt, chan_gtalk still + works and should be used in this version. + +2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp + + * channels/chan_sip.c: Change the fd on the I/O context in case it + changed during the reload, which is indeed possible. (issue #7943 + reported by eclubb) + + * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN + instead of hardcoding the path for the error message (issue #7942 + reported by eclubb) + +2006-10-02 18:52 +0000 [r44186] Paul Cadach + + * configs/users.conf.sample, pbx/pbx_config.c: Missed part of + userconf functionality for chan_h323 + +2006-10-02 17:25 +0000 [r44169] Joshua Colp + + * main/io.c: Shrink when current_ioc is unused. It is set to -1 when + unused, not 0. (issue #7941 reported by eclubb) + +2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach + + * doc/realtime.txt: Typo fix + + * channels/chan_h323.c: Optimization of oh323_indicate(): less + locks - less problems, plus single exit point + +2006-10-02 02:38 +0000 [r44146] Mark Spencer + + * channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when + you're not talking about a channel :) + +2006-10-01 19:32 +0000 [r44135] Paul Cadach + + * channels/chan_h323.c: Do not simulate any audio tones if we got + PROGRESS message + +2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant + + * Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to + be empty. The cause is that since ASTDATADIR is explicitly + exported using "export ASTDATADIR" at the top of the Makefile, + make no longer considers the variable "undefined", so the + Makefile can't use ?= to set ASTDATADIR if not yet set. (issue + #8063, reported by akohlsmith, fixed by me) + + * configs/queues.conf.sample: Fix the name of the "eventmemberstatus" + option in the sample queues.conf (issue #8065, adamg) + +2006-10-01 15:01 +0000 [r44109] Luigi Rizzo + + * channels/chan_sip.c: sync with trunk - move variable declarations + to the beginning of a block. + +2006-09-30 19:20 +0000 [r44090] Paul Cadach + + * main/rtp.c: Allow one-way RTP streams (device->Asterisk) + +2006-09-30 16:28 +0000 [r44080] Luigi Rizzo + + * codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent + build problems: - with AST_DEVMODE, building codecs/lpc10 fails + because of lots of warnings, and the configure step in editline + fails as well. Fix this by removing the -Werror in these steps. - + on FreeBSD (but probably on other platforms as well), the final + link of asterisk fails because AST_LIBS was not exported to the + subdirs Makefiles. Add a proper fix in the top-level Makefile (a + possible alternative way is to add "export AST_LIBS" near the + beginning of the file). With this fix, i believe that some of the + platform-specific conditionals in main/Makefile are redundant + (because they should be already dealt with in the top level + Makefile) but i don't have a platform to check. Merging to head + will happen in a moment. + +2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach + + * channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment + of previous fix: Issue #7928 - Don't send both 404 and 503. Fix + by phsultan with a small fix by me, myself or I. Thanks, + Philippe! (This was caused by my changes to the transaction + handling) + + * channels/chan_sip.c: Found some buggy SIP clients (phones Planet + VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which + sends ACK not on OK message only (when remote party answers) but + on RINGING message too, so when we send 200 OK message, we get + unidentified ACK message (because INVITE acknowledged on RINGING + message already), so 200 OK retransmits within its retransmission + interval then call gets dropped. If someone else knows how to + provide workaround for such cases, please, fix it in correct way. + Thanks to ssh from #asteriskru for provide access to his box to + study and fix this case. + +2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming + + * agi, utils: ignore temporary files made by the Makefiles during a + build + + * codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile, + codecs/Makefile, utils/Makefile, configure, + build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac, + Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile, + pbx/Makefile, res/Makefile, channels/Makefile: fix a few build + system bugs, and convert Makefiles to be compatible with GNU make + 3.80 + +2006-09-29 22:35 +0000 [r44053] Jason Parker + + * main/asterisk.c, main/cli.c: Fix a bug with the removal of + 'atleast' argument to 'core verbose' and 'core debug'. Add that + argument back in. + +2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach + + * channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more + carefully when no CallingNumber IE available + + * channels/h323/ast_h323.cxx: Fake display name by called number on + incoming calls (until passing connected number/connected name is + not implemented) + + * channels/h323/ast_h323.cxx: Ported code refers to H.450 - add + includes + + * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly + pass TON/PRESENTATION information - original + H323Connection::SendSignalSetup() destroys Q.931 fields. + +2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming + + * main/Makefile: yet another place where we were not using the + correct CFLAGS by default + + * main/Makefile: missed one conversion to ASTCFLAGS + +2006-09-29 18:30 +0000 [r44009] Paul Cadach + + * channels/h323/ast_h323.cxx, channels/chan_h323.c, + channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass + TON/PRESENTATION information too + +2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming + + * main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile, + main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, + Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse + CFLAGS and LDFLAGS for build of Asterisk components, because they + are also then used for non-Asterisk components (like menuselect); + use our own variables instead + + * configure, configure.ac: support --without-curl in configure + script + + * Makefile.rules: another cross-compile fix + + * Makefile: a couple more environment settings that can't leak into + the menuselect build + + * main/cli.c: proper fix for ast_group_t change + + * include/asterisk/lock.h: eliminate compiler warning when + DEBUG_CHANNEL_LOCKS is enabled and users of this header file + don't also include channel.h + +2006-09-28 20:11 +0000 [r43944] Jason Parker + + * apps/app_queue.c: Fix incorrect argument order for member names, + on persisted members. Issue 8047, patch by jmls. + +2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp + + * apps/app_playback.c, res/res_monitor.c, + include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c, + channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c, + main/udptl.c, main/frame.c, funcs/func_timeout.c, + channels/chan_sip.c, apps/app_festival.c, + channels/iax2-provision.c, apps/app_alarmreceiver.c, + res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c: + Put in missing \ns on the end of ast_logs (issue #7936 reported + by wojtekka) + +2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming + + * apps/app_queue.c: fix buggy (and overly complex) loop used during reload + of app_queue for static member list updating + +2006-09-28 17:34 +0000 [r43918] Paul Cadach + + * channels/h323/ast_h323.cxx: Extend call establishment timeout + +2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp + + * channels/chan_iax2.c: Make sure the pvt exists before accessing + it again as it may have gone away (issue #7562 reported by Seb7 + and issue #7939 reported by sorg) + + * main/cli.c: Warning be gone! + +2006-09-28 16:41 +0000 [r43899] BJ Weschke + + * apps/app_queue.c: app_queue is comparing the device names incorrectly + while checking their statuses. It's internal list of interfaces + includes the dial string, while the argument passed to this + function does not have the dial string (/n for a local channel). + This causes it to ignore the device state changes because it + thinks it belongs to none of its members. (#8040 reported and + patch by tim_ringenbach) + +2006-09-28 16:17 +0000 [r43893] Joshua Colp + + * apps/app_meetme.c: Stop the stream after waitstream returns so that our + formats get restored. (issue #7370 reported by kryptolus) + +2006-09-28 15:56 +0000 [r43877] Paul Cadach + + * channels/h323/ast_h323.cxx: Fix compiler warning + +2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke + + * apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 - + tim_ringenbach reported and patched) + + * apps/app_queue.c: Autopause not working for queue members. (#8042 + - jmls reported and patch) + +2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach + + * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force + remote side to start media on outgoing PROGRESS message + + * include/asterisk/compiler.h: Put attribute tag at correct place + +2006-09-28 11:03 +0000 [r43852] Christian Richter + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, + channels/chan_misdn.c: fixed a bug which led to chan_list zombies, + when the call could not be properly established in misdn_call. + also removed the ACK_HDLC stuff which is not really needed. + +2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach + + * channels/h323/ast_h323.cxx: Do not open transmit channel until + TCS is received + + * main/file.c: Don't warn on HOLD/UNHOLD control frames + + * main/file.c: Don't treat unknown control frames as voice + +2006-09-27 20:21 +0000 [r43816] Tilghman Lesher + + * apps/app_voicemail.c: Avoid inability to lock directory log message by + creating the directory ahead of time. (Issue 7631) + +2006-09-27 19:44 +0000 [r43801-43803] Jason Parker + + * apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS + not being set under certain circumstances. Fix a minor issue, to + make it use the filenames that were parsed, instead of the entire + argument string. Fix Background() to return -1 like Playback(), + if no args are specified. + +2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp + + * main/rtp.c: Compensate for out of order packets better if RFC2833 + compensation is turned on. + + * channels/chan_iax2.c: Get rid of two functions from a time now + past (we THINK these are from pre-recursive lock time) that may + be contributing to two open issues on the bug tracker (7562/7939) + and that has the potential to just make bad things happen if the + timing is right. + +2006-09-27 16:55 +0000 [r43779] Russell Bryant + + * main/channel.c,res/res_features.c: Fix a problem that occurred if + a user entered a digit + that matched a bridge feature that was configured using multiple + digits, and the digit that was pressed timed out in the feature + digit timeout period. For example, if blind transfer is + configured as '##', and a user presses just '#'. In this + situation, the call would lock up and no longer pass any frames. + (issue #7977 reported by festr, and issue #7982 reported by + michaels and valuable input provided by mneuhauser and kuj. Fixed + by me, with testing help and peer review from Joshua Colp). There + are a couple of issues involved in this fix: 1) When + ast_generic_bridge determines that there has been a timeout, it + returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets + this result, it calls ast_generic_bridge over again with the same + timestamp for the next event. This results in an endless loop of + nothing until the call is terminated. This is resolved by simply + changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it + sees a timeout. 2) I also changed ast_channel_bridge such that if + in the process of calculating the time until the next event, it + knows a timeout has already occured, to immediately return + AST_BRIDGE_COMPLETE instead of attempting to bridge the channels + anyway. 3) In the process of testing the previous two changes, I + ran into a problem in res_features where ast_channel_bridge would + return because it determined that there was a timeout. However, + ast_bridge_call in res_features would then determine by its own + calculation that there was still 1 ms before the timeout really + occurs. It would then proceed, and since the bridge broke out and + did *not* return a frame, it interpreted this as the call was + over and hung up the channels. The reason for this was because + ast_bridge_call in res_features and ast_channel_bridge in + channel.c were using different times for their calculations. + channel.c uses the start_time on the bridge config, which is the + time that the feature digit was recieved. However, res_features + had another time, 'start', which was set right before calling + ast_channel_bridge. 'start' will always be slightly after + start_time in the bridge config, and sometimes enough to round up + to one ms. This is fixed by making ast_bridge_call use the same + time as ast_channel_bridge for the timeout calculation. ........ + +2006-09-27 16:24 +0000 [r43775] Christian Richter + + * channels/chan_misdn.c, channels/Makefile: removed the chan_misdn + versioning, since Asterisk has it's own + +2006-09-27 16:23 +0000 [r43774] Joshua Colp + + * channels/chan_sip.c: Make rfc2833compensate a global option. + +2006-09-27 04:35 +0000 [r43756] Russell Bryant + + * apps/app_voicemail.c: Backport revision 43754 from the trunk, + which removes an unused buffer from mm_login to close bug 8038, + as well as addresses some formatting and coding guidelines issues + in passing. Originally, I did not commit this to 1.4 since it is + not necessarily fixing a bug. However, since the IMAP storage + code is brand new, I decided it would be better to make the + change here as well, in case someone has to work on this code to + address issues in the very near future. I don't want to make + unnecessary merge problems going to the trunk. + +2006-09-27 02:32 +0000 [r43739] Steve Murphy + + * configs/extensions.ael.sample: This change to extensions.ael was + to fix bug 8031; the install scripts are causing it to be copied + to /etc/asterisk/extensions.ael, and because it is a fairly + direct conversion of the original extensions.conf, the macro and + context names clash with the existing extensions.conf. So, I put + an ael- in front of all macros and contexts, and checked every + goto and macro call. Also, this file compiles under aelparse. + +2006-09-26 20:56 +0000 [r43710] Russell Bryant + + * main/asterisk.c: Back in revision 4798, this message was changed from + using ast_cli() to directly calling write(). During this change, + checking if this was a remote console was removed. This caused + this message about using "exit" or "quit" to exit an Asterisk + console to come up in times where it did not make sense. This + change restores the check to see if this is a remote console + before printing the message. (fixes BE-65) + +2006-09-26 20:47 +0000 [r43707] Joshua Colp + + * .cleancount, main/cli.c, channels/chan_sip.c, + include/asterisk/channel.h: Use proper type to represent the group variable + (issue #8025 reported by makoto) + +2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant + + * channels/chan_sip.c: Add missing newline character in the warning + message about deprecated TOS values in configuration. + + * apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain + mailbox definitions, don't introduce a length limit on the + definition by using a 256 byte temporary storage buffer. Instead, + make the temporary buffer just as big as it needs to be to hold + the entire mailbox definition. (fixes BE-68) + +2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp + + * channels/chan_local.c: Strip options off the argument passed for + devicestate in chan_local. (issue #8034 reported by pcardozo) + + * apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight + overhaul of the whisper support. 1. We need to duplicate the + frame from ast_translate 2. We need to ensure we always have + signed linear coming in for signed linear combining. 3. We need + to ensure we are always feeding signed linear out. 4. Properly + store and restore write format when beeping on the channel we are + whispering on. 5. Properly discontinue the stream on the channel + for the beep. (issue #8019 reported by timkelly1980) + +2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming + + * sounds/Makefile: update to use 1.4.3 core sounds, with corrected + beep/beeperr/tt-monkeys files + +2006-09-26 18:08 +0000 [r43650-43674] Jason Parker + + * doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by + Dan Austin. Maximum values were incorrect, which is why this is + being put in 1.4 + + * channels/chan_skinny.c: Add proper codec support to chan_skinny. + Works with at least ulaw, alaw, and g729a. This is technically a + "new feature", but there are justifications for it. I found a bug + with the recent rtp packetization changes, which caused the media + setup to fail under certain circumstances, particularly when + using allow=all, or having no allow= statements (globally or on + the device). I could have either removed the rtp packetization + features, or I could add proper codec support (which, without, I + think most people would consider to be a bug anyways). + +2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher + + * apps/app_voicemail.c: Should have moved these lines up in the + merge, instead of removing them + + * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1) + delete=yes was ignored 2) maxmessages was ignored + +2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach + + * channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h, + channels/h323/cisco-h225.asn: Fix ASN1 description of + non-standard Cisco extensions + + * channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport + changes of trunk: 1) r43540: Avoid possible deadlock on channel + destruction 2) r43590: Disable fastStart if requested by remote + side + +2006-09-25 15:23 +0000 [r43616] Jason Parker + + * sounds/Makefile: One more fix for sounds installation - this time + for portability. Reported to asterisk-dev mailing list. + +2006-09-25 14:52 +0000 [r43605] Steve Murphy + + * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from + crashing if trying to play an OGG moh file. + +2006-09-25 06:15 +0000 [r43582] Paul Cadach + + * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h, + channels/chan_h323.c: Merged revisions 43472,43495 from trunk + +2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant + + * channels/iax2-provision.c: Fix a CLI command registration issue + where an erroneous message claiming that "iax2 show provisioning" + was already registered. This was because this command was + registering itself as both the command, as well as the command it + is deprecating. (issue #8022, reported by bjweeks, fixed by + myself) + + * channels/chan_iax2.c:Check to see if the channel that is activating the + IAXPEER function is actually an IAX2 channel before proceeding to + process it to avoid crashing. (issue #8017, reported by admott, + fixed by myself) + +2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming + + * Makefile: don't output the 'build complete' message when the + target being run is already going to do an installation + +2006-09-22 22:12 +0000 [r43518] Jason Parker + + * channels/chan_skinny.c: Allow chan_skinny.so to be unloaded + properly. Remove reload support, since it doesn't + actually...work. + +2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy + + * pbx/pbx_ael.c: This commits a change to return + MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all + goes well for bug 8004 + + * pbx/pbx_ael.c: If the extensions.ael file not found, or + unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004. + +2006-09-22 17:25 +0000 [r43492] Jason Parker + + * main/cli.c: Make sure we explicitly set the CLI command to not be + deprecated, if it isn't. + +2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming + + * sounds/Makefile: use rebuilt extra sounds + + * main/channel.c: all the Linux systems I have don't use + '__m_count' for this field, so I don't know where this came + from... + +2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant + + * include/asterisk/threadstorage.h: backport the compatability fix + to use attribute_malloc instaed of __attribute__ ((malloc)) + + * channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN + could not be configured (issue #8006, Mithraen) + + * main/frame.c: Suppress a compiler warning about the use of a + potentially uninitialized variable. It couldn't actually happen, + though. + +2006-09-22 03:01 +0000 [r43469] Jason Parker + + * channels/chan_skinny.c: First shot at unload_module in + chan_skinny.. More to come. + +2006-09-21 23:50 +0000 [r43466] Matt O'Gorman + + * include/asterisk/jabber.h, channels/chan_gtalk.c, + res/res_jabber.c: updates for better compontent support + +2006-09-21 23:24 +0000 [r43464] Tilghman Lesher + + * res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we + actually documented how the new features in res_odbc actually + work. (Oops) + +2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp + + * channels/chan_oss.c: Some more clean up in the load function for + chan_oss (issue #8002 reported by Mithraen with minor mods by + moi) + + * channels/chan_mgcp.c: Clean up chan_mgcp's module load function + (issue #8001 reported by Mithraen with mods by moi) + +2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming + + * main/Makefile, build_tools/strip_nonapi (added): add another + attempt to strip non-API symbols from the final binary... script + will need to be extended to work on non-Linux systems + +2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher + + * apps/app_url.c: Fix documentation to reflect how Url() really + works + + * cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates + +2006-09-21 Kevin P. Fleming + + * Asterisk 1.4.0-beta2 released. + +2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming + + * main/Makefile: remove this change... it requires binutils 2.17 + +2006-09-20 23:19 +0000 [r43396] Jason Parker + + * build_tools/make_version: fix minor typo in the way version is + handled + +2006-09-20 Kevin P. Fleming + + * Asterisk 1.4.0-beta1 released. -- cgit v1.2.3