From b3dac212ac04cab1dfcfdfa30dba53989d59fcda Mon Sep 17 00:00:00 2001 From: lmadsen Date: Tue, 10 Aug 2010 17:51:30 +0000 Subject: Importing files for 1.8.0-beta3 release. git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0-beta3@281571 f38db490-d61c-443f-a65b-d21fe96a405b --- .lastclean | 3 + .version | 1 + ChangeLog | 22649 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 22653 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..c364cf642 --- /dev/null +++ b/.lastclean @@ -0,0 +1,3 @@ +38 + + diff --git a/.version b/.version new file mode 100644 index 000000000..431eaf9f0 --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.8.0-beta3 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..249898ea7 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,22649 @@ +2010-08-10 Leif Madsen + + * Asterisk 1.8.0-beta3 Released. + +2010-08-10 17:48 +0000 [r281529-281568] Russell Bryant + + * apps/app_dial.c, /: Merged revisions 281567 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281567 | russell | 2010-08-10 12:47:13 -0500 + (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) + | 8 lines Reset visible indication after answer. (closes issue + #17641) Reported by: klaus3000 Patches: + ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by + klaus3000 (license 65) Tested by: schmidts ........ + ................ + + * channels/chan_sip.c: Ensure that the proper external address is + used for the RTP destination. (closes issue #17044) Reported by: + ebroad Tested by: ebroad Review: + https://reviewboard.asterisk.org/r/566/ + + * main/cli.c: Resolve a problem with channel name tab completion. + Hitting tab without typing any part of a channel name resulted in + no results. This now results in getting a full list of active + channels, just as it did in previous versions of Asterisk. + Review: https://reviewboard.asterisk.org/r/818/ + +2010-08-10 07:26 +0000 [r281497] TransNexus OSP Development + + * apps/app_osplookup.c: Fixed the issue caused by EXTEN including + user parameters. + +2010-08-09 23:04 +0000 [r281466] Jeff Peeler + + * channels/chan_local.c: Add some more stuff to copy from 281429. + +2010-08-09 20:47 +0000 [r281432] David Vossel + + * /, channels/chan_sip.c: Merged revisions 281430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010) + | 13 lines fixes SIP peers memory leak We zeroed out the peer's + addr before it was removed from the peers_by_ip container. This + made it impossible to be removed from the container as the addr + is the key used by the container to find the peer. (closes issue + #17774) Reported by: kkm Patches: + 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888) + 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888) + ........ + +2010-08-09 20:43 +0000 [r281429] Jeff Peeler + + * channels/chan_local.c, /: Merged revisions 281391 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500 + (Mon, 09 Aug 2010) | 20 lines Merged revisions 281390 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) + | 13 lines Prevent loss of Caller ID information set on local + channel after masquerade. Caller ID set on the channel before a + masquerade occurs when using a local channel would cause the + information to be lost. The problem was that the information was + set on a channel destined to be hung up. The somewhat confusing + fix is to detect if any Caller ID has been set on the channel and + if so preswap the Caller ID data so that basically the masquerade + puts the data back. (closes issue #17138) Reported by: kobaz + Review: https://reviewboard.asterisk.org/r/847/ ........ + ................ + +2010-08-09 14:49 +0000 [r281358] Matthew Nicholson + + * res/res_fax.c: Validate minrate, maxrate, and modem settings + before attempting a fax session. FAX-224 + +2010-08-09 14:31 +0000 [r281356] + + * configs/sip.conf.sample: Added comment about IPv4-mapped IPv6 + addresses and the output of netstat. + +2010-08-09 12:51 +0000 [r281294-281325] Russell Bryant + + * configs/cdr.conf.sample: Add a couple of default values to the + documentation of cdr.conf. + + * configs/cdr.conf.sample: Reorder some options in cdr.conf.sample. + Put all of the options that affect the contents of CDRs together, + instead of having the batch mode options in the middle of them. + +2010-08-06 18:57 +0000 [r281085] Tilghman Lesher + + * main/utils.c: Fix alignment of stringfields on the SPARC + architecture (closes issue #17789) Reported by: Ian Mason + Patches: 20100806__issue17789__2.diff.txt uploaded by tilghman + (license 14) Tested by: Ian_Mason + +2010-08-05 13:16 +0000 [r281052] Russell Bryant + + * main/cdr.c, /: Merged revisions 281051 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010) + | 9 lines Cleanup default option value handling for cdr.conf + [general]. The default values would differ depending on whether + or not cdr.conf exists. That is no longer the case. Apply a + default value to the unanswered option. Define all default values + as named constants. ........ + +2010-08-05 07:46 +0000 [r280984] Tilghman Lesher + + * include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280983 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500 + (Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) + | 8 lines Change context lock back to a mutex, because + functionality depends upon the lock being recursive. (closes + issue #17643) Reported by: zerohalo Patches: + 20100726__issue17643.diff.txt uploaded by tilghman (license 14) + Tested by: zerohalo ........ ................ + +2010-08-04 15:11 +0000 [r280909] Matthew Nicholson + + * res/res_fax.c: Initialize FAXOPT() status variables in sendfax + and receivefax instead of when the details structure is created. + +2010-08-04 14:04 +0000 [r280809-280879] Tilghman Lesher + + * channels/chan_mgcp.c: Check cur value before attempting a deref. + (closes issue #17775) Reported by: svinson Patches: + 20100804__issue17775.diff.txt uploaded by tilghman (license 14) + Tested by: svinson (closes issue #17743) Reported by: tgruenberg + Patches: 20100804__issue17775.diff.txt uploaded by tilghman + (license 14) Tested by: tgruenberg + + * CHANGES, funcs/func_strings.c: Sneak FIELDNUM() into 1.8. Returns + a 1-based index into a list of a specified item. Matches up with + FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth + Patches: svn-279754.diff uploaded by gareth (license 208) Tested + by: gareth, tilghman Review: + https://reviewboard.asterisk.org/r/810/ + +2010-08-03 19:54 +0000 [r280777-280778] + + * channels/chan_sip.c: Fixed IPv6-related SIP parsing bugs. (closes + issue #17663) Reported by: oej Patches: diff uploaded by + sperreault (license 252) diff2 uploaded by sperreault (license + 252) get_domain.diff uploaded by sperreault (license 252) + + * configs/sip.conf.sample: Better documentation related to IPv6. + (closes issue #17737) Reported by: oej Patches: doc.diff uploaded + by sperreault (license 252) Tested by: mmichelson + +2010-08-03 18:48 +0000 [r280742] Russell Bryant + + * addons/Makefile, addons/mp3 (removed), + contrib/scripts/get_mp3_source.sh (added): Remove the MP3 decoder + source code and replace it with a small shell script. Review: + https://reviewboard.asterisk.org/r/836/ + +2010-08-03 18:42 +0000 [r280624-280740] Tilghman Lesher + + * doc/asterisk.sgml, /, doc/asterisk.8, doc/Makefile (added): + Merged revisions 280739 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010) + | 2 lines Document -B and -W flags and regenerate manpage from + sgml ........ + + * apps/app_voicemail.c, /: Merged revisions 280671 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02 + Aug 2010) | 2 lines Allow the pipe, but also allow the comma + ........ + + * main/Makefile: Make this a little more deterministic... we want + the latest value, not just a 1 somewhere. + + * main/Makefile: Apparently, the values in makeopts are sometimes + 1:1 and sometimes 1. Compensate for this. + +2010-07-29 21:07 +0000 [r280557] Matthew Nicholson + + * res/res_fax.c: Fix regression introduced in r1664. Give the fax + stack time to shutdown and populate the FAXOPT output variables. + FAX-222 + +2010-07-29 20:43 +0000 [r280552] David Vossel + + * /, channels/chan_sip.c: Merged revisions 280551 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010) + | 11 lines fixes wrong SRV query for TLS connection (closes issue + #17612) Reported by: marcelloceschia Patches: + chan-sip_srvQuery.patch uploaded by marcelloceschia (license + 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) + chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia + (license 1079) Tested by: marcelloceschia, st, pabelanger + ........ + +2010-07-29 20:35 +0000 [r280549] Russell Bryant + + * configs/ccss.conf.sample: Add header to ccss.conf to appease oej. + (closes issue #17755) Reported by: oej + +2010-07-29 19:47 +0000 [r280519] Sean Bright + + * channels/sig_pri.c: Fix compilation error in chan_dahdi (strdupa + -> ast_strdupa). (closes issue #17751) Reported by: b11d Patches: + strdupa_oops.diff uploaded by malcolmd (license 924) + +2010-07-29 19:13 +0000 [r280450] David Vossel + + * main/channel.c, /: Merged revisions 280449 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500 + (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) + | 12 lines fixes issue with translator frame not getting freed A + translator frame even if it local storage so the translation path + can be freed. This issue prevented g729 licenses from being freed + up. (closes issue #17630) Reported by: manvirr Patches: + encoder_fix.diff uploaded by dvossel (license 671) Tested by: + manvirr, dvossel ........ ................ + +2010-07-29 18:37 +0000 [r280414-280446] Paul Belanger + + * tests/test_utils.c: Remove res_crypto dependency. + + * tests/test_utils.c: crypto_loaded_test depends on res_crypto, + else test will fail. + +2010-07-29 16:25 +0000 [r280391] Russell Bryant + + * main/rtp_engine.c: Don't blow up if get_codec() was not provided + in the RTP glue. + +2010-07-29 16:07 +0000 [r280346] Jean Galarneau + + * /, apps/app_meetme.c: Merged revisions 280345 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280345 | jeang | 2010-07-29 11:01:35 -0500 + (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | + 2 lines Fix a dsp structure leak occuring when a local channel is + put into a meetme conference, then masquaraded away. ABE-2422 + ........ ................ + +2010-07-29 15:57 +0000 [r280307-280343] Matthew Nicholson + + * channels/chan_usbradio.c: Use PRIx64 instead of PRId64 in format + string. related to r280302 + + * main/channel.c, channels/chan_local.c, /: Merged revisions 280306 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul + 2010) | 2 lines Implement support for ast_channel_queryoption on + local channels. Currently only AST_OPTION_T38_STATE is supported. + ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........ + Additionally, pass AST_CONTROL_T38_PARAMETERS control frames + through generic bridges. This change appears to have been + unintentionally left out of rev 203699. + +2010-07-29 00:45 +0000 [r280302] Paul Belanger + + * channels/chan_usbradio.c: Use PRId64 with format_t + +2010-07-28 20:49 +0000 [r280269] Jeff Peeler + + * channels/sip/reqresp_parser.c: Give test category missing leading + slash + +2010-07-28 20:12 +0000 [r280235] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 280229 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28 + Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7 + called_nai and calling_nai config options. ........ + +2010-07-28 20:03 +0000 [r280233] Jason Parker + + * sounds/Makefile, /: Merged revisions 280231 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) | + 6 lines Work around some silly behavior on BSD. A non-zero exit + from a subshell should make the build fail. (closes issue #17621) + ........ + +2010-07-28 19:34 +0000 [r280225] Terry Wilson + + * res/res_rtp_asterisk.c: Do rtp/rtcp debugging when it is turned + on w/o filtering + +2010-07-28 18:24 +0000 [r280195] Jason Parker + + * sounds/Makefile, /: Merged revisions 280193 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) | + 9 lines Remove unnecessary subshells. Attempt to make + checksumming work. Also improves readability. (issue #17621) + Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/ + ........ + +2010-07-28 16:52 +0000 [r280161] Sean Bright + + * apps/app_queue.c, /: Merged revisions 280160 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul + 2010) | 8 lines Plug a reference leak in app_queue when adding + members dynamically. (closes issue #17738) Reported by: + bobwienholt Patches: issue17738.patch uploaded by bobwienholt + (license 950) Tested by: bobwienholt, seanbright ........ + +2010-07-28 13:52 +0000 [r280090] Leif Madsen + + * contrib/scripts/live_ast, /: Merged revisions 280089 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280089 | lmadsen | 2010-07-28 08:51:16 -0500 + (Wed, 28 Jul 2010) | 9 lines Merged revisions 280088 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 + Jul 2010) | 1 line Update help text to be less confusing. + ........ ................ + +2010-07-28 13:01 +0000 [r280058] Russell Bryant + + * res/res_crypto.c: s/init keys/keys init/ + +2010-07-28 01:37 +0000 [r280023] Paul Belanger + + * channels/chan_usbradio.c: Resolve compiler warning about + formatting (closes issue #17732) Reported by: pabelanger + +2010-07-27 22:30 +0000 [r280019-280020] Sean Bright + + * main/editline/el.h, main/term.c, main/cli.c, + main/editline/parse.c, main/editline/tokenizer.c, + main/editline/config.sub, main/editline/parse.h, + main/editline/tokenizer.h, configure, main/editline/histedit.h, + main/editline/sig.c, main/editline/PLATFORMS, + main/editline/sig.h, main/editline/key.c, main/editline/editrc.5, + main/editline/np/fgetln.c, main/editline/key.h, + main/editline/TEST/test.c, main/Makefile, + main/editline/configure, main/editline/Makefile.in, configure.ac, + main/editline/configure.in, main/editline/readline/readline.h, + main/editline/README, main/editline/editline.3, + main/editline/vi.c, main/editline/sys.h, main/editline/emacs.c, + main/asterisk.c, main/editline/install-sh, main/editline/term.c, + main/editline/config.guess, main/editline/read.c, + main/editline/term.h, main/editline/map.c, + main/editline/np/strlcpy.c, main/editline (added), + main/editline/config.h.in, main/editline/read.h, + main/editline/tty.c, main/editline/np/unvis.c, + main/editline/prompt.c, main/editline/map.h, main/editline/tty.h, + main/editline/chared.c, main/editline/prompt.h, + main/editline/np/strlcat.c, main/editline/chared.h, + main/editline/np, main/editline/TEST, main/editline/refresh.c, + main/editline/history.c, main/editline/readline, + include/asterisk/term.h, main/editline/refresh.h, + main/editline/search.c, main/editline/hist.c, + main/editline/search.h, main/editline/hist.h, + main/editline/np/vis.c, build_tools/menuselect-deps.in, main, + main/editline/readline.c, main/editline/np/vis.h, + main/editline/INSTALL, makeopts.in, main/editline/CHANGES, + main/editline/common.c, main/xmldoc.c, main/editline/makelist.in, + include/asterisk/autoconfig.h.in, main/editline/el.c: Revert + r280019 for now - This was poorly executed. + + * include/asterisk/term.h, makeopts.in, main/asterisk.c, + main/xmldoc.c, main/cli.c, main/term.c, main/editline (removed), + build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, + main: Add ability to use system libedit and update bundled + libedit. The version of libedit that is bundled with asterisk is + old and has some bugs. This patch updates the bundled version of + libedit within asterisk, and also updates asterisk to use the + system libedit instead if one is available (and pkg-config is + available). This review integrates several patches from other + users specifically kkm and tzafrir. (closes issue #15929) + Reported by: kkm Patches: 015929-astcli-editrc-trunk.240324.diff + uploaded by kkm (license 888) (issue #16858) Reported by: + jw-asterisk (closes issue #17039) Reported by: tzafrir Patches: + 0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir + (license 46) Review: https://reviewboard.asterisk.org/r/807/ + +2010-07-27 21:16 +0000 [r279953] Russell Bryant + + * res/ais, main/db1-ast/mpool, Makefile.rules, res/snmp, cdr, + formats, codecs/gsm/src, funcs, bridges, codecs/lpc10, + main/db1-ast/btree, configure, main/editline, codecs/g722, main, + main/db1-ast/recno, channels/sip, makeopts.in, pbx, res, res/ael, + channels, main/stdtime, main/editline/np, codecs, utils, + main/db1-ast/hash, cel, apps, configure.ac, main/db1-ast/db: Add + --enable-coverage option to configure script. This option enables + the proper compiler flags for tracking code coverage, which is + useful along side automated testing. + +2010-07-27 20:57 +0000 [r279949] David Vossel + + * main/audiohook.c, main/channel.c, /, + include/asterisk/audiohook.h: Merged revisions 279946 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500 + (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) + | 19 lines remove empty audiohook write list on channel If a + channel has an audiohook write list created on it, that list + stays on the channel until the channel is destroyed. There is no + reason to keep that list on the channel if it becomes empty. If + it is empty that just means we are doing needless translating for + every ast_read and ast_write. This patch removes the audiohook + list from the channel once it is detected to be empty on either a + read or write. If a audiohook is added back to the channel after + this list is destroyed, the list just gets recreated as if it + never existed to begin with. (closes issue #17630) Reported by: + manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ + ................ + +2010-07-27 19:50 +0000 [r279916] Russell Bryant + + * channels/sig_pri.c, channels/chan_dahdi.c: Fix inband DTMF + detection on outgoing ISDN calls. This is a regression from the + sig_pri split from chan_dahdi. When a call is first initiated, + the inband DTMF detector is not enabled if it's an outgoing ISDN + call. However, it needs to be turned on once the media path + starts up. This handling was put back in the open_media() + callback of chan_dahdi. In sig_pri, open_media() calls were added + to a few places where it was needed, including handling of + PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING. + Thanks to rmudgett for helping me with the patch! + +2010-07-27 18:54 +0000 [r279887] Mark Michelson + + * channels/chan_sip.c: Fix parsing error in sip_sipredirect(). The + code was written in a way that did a bad job of parsing the port + out of a URI. Specifically, it would do badly when dealing with + an IPv6 address. In this particular scenario, there was no value + from parsing the port out, so I just removed that logic. And + while I was messing around in the function, I changed some + variable names to be more descriptive. (closes issue #17661) + Reported by: oej Patches: 17661.diff uploaded by mmichelson + (license 60) + +2010-07-27 16:40 +0000 [r279850] Jason Parker + + * sounds/Makefile, /: Merged revisions 279849 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul 2010) | + 1 line Simply sounds/Makefile some more. ........ + +2010-07-27 16:09 +0000 [r279817] David Vossel + + * main/netsock2.c, channels/chan_sip.c: fix sip transaction match + with authentication, fix confusing log message when using + getaddrinfo + +2010-07-27 16:06 +0000 [r279815] Russell Bryant + + * channels/chan_dahdi.c: Support "channels" in addition to + "channel" in chan_dahdi.conf. Review: + https://reviewboard.asterisk.org/r/804 + +2010-07-27 15:15 +0000 [r279785] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 279784 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul + 2010) | 14 lines Fix bad behavior of dynamic_exclude_static + option in sip.conf. We were attempting to create a contactdeny + rule based on the peer's IP address before the peer's IP address + had been set. By moving the processing further down in the + function, we can ensure stuff works as we expect for it to. + (closes issue #17717) Reported by: mmichelson Patches: + 17717.patch uploaded by mmichelson (license 60) Tested by: + DennisD ........ + +2010-07-27 02:57 +0000 [r279726-279755] Paul Belanger + + * channels/chan_dahdi.c: If dringXcontext is null, fallback to + default context value. (closes issue #17693) Reported by: + iasgoscouk Patches: issue17693.patch uploaded by pabelanger + (license 224) Tested by: iasgoscouk Review: + https://reviewboard.asterisk.org/r/803/ + + * main/http.c: Use ast_sockaddr_setnull() when http is not enabled. + Otherwise, ast_tcptls_server_start() will still start http. + (closes issue #17708) Reported by: pabelanger Patches: http.patch + uploaded by pabelanger (license 224) + +2010-07-26 Leif Madsen + + * Asterisk 1.8.0-beta2 Released. + +2010-07-26 23:29 +0000 [r279689] Paul Belanger + + * UPGRADE.txt, CHANGES: Updated documentation for FAX logger level. + +2010-07-26 23:03 +0000 [r279658] Jason Parker + + * sounds/Makefile (added), /, sounds/Makefile.380 (removed), + configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381 + (removed), configure.ac: Merged revisions 279657 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul + 2010) | 5 lines Really fix sounds Makefile (and make it + readableish). There was a rather large syntax error that should + have caused ALL versions of GNU make to fail. I don't know how it + worked. ........ + +2010-07-26 21:53 +0000 [r279636] Russell Bryant + + * main/channel.c: Ignore a control subclass of -1 in + ast_waitfordigit_full(). + +2010-07-26 21:20 +0000 [r279599-279619] Tilghman Lesher + + * /, configure, configure.ac: Merged revisions 279609 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26 + Jul 2010) | 2 lines Dunno why this worked on my machine, but it + works better this way. ........ + + * res/res_config_ldap.c, /: Merged revisions 279597 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26 + Jul 2010) | 13 lines Apply all patches in: + https://issues.asterisk.org/view.php?id=13573 (closes issue + #13573) Reported by: navkumar Patches: + res_config_ldap-category.diff uploaded by navkumar (license 580) + res_config_ldap.patch uploaded by bencer (license 961) + res_config_ldap uploaded by bencer (license 961) Tested by: + suretec ........ + + * /: Reverting property remove + +2010-07-26 20:58 +0000 [r279598] Gavin Henry + + * /: Merged revisions 279597 via svnmerge from + https://origsvn.digium.com/svn/asterisk/1.6.2 + ----------------------------------------------------------------------- + r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) | + 13 lines Apply all patches in: + https://issues.asterisk.org/view.php?id=13573 [^] (closes issue + 0013573) Reported by: navkumar Patches: + res_config_ldap-category.diff uploaded by navkumar (license 580) + res_config_ldap.patch uploaded by bencer (license 961) + res_config_ldap uploaded by bencer (license 961) Tested by: + suretec + ------------------------------------------------------------------------ + +2010-07-26 19:59 +0000 [r279568] David Vossel + + * channels/sip/include/sip.h, + channels/sip/include/reqresp_parser.h, channels/chan_sip.c, + channels/sip/reqresp_parser.c: transaction matching using top + most Via header This patch modifies the way chan_sip.c does + transaction to dialog matching. Asterisk now stores information + in the top most Via header of the initial incoming request and + compares that against other Requests that have the same call-id. + This results in Asterisk being able to detect a forked call in + which it has received multiple legs of the fork. I completely + stripped out the previous matching code and made the comparisons + a little more explicit and easier to understand. My comments in + the code should offer all the details involving this patch. This + patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to + find multiple dialogs with the same call-id. Since the callback + function was returning (CMP_MATCH | CMP_STOP) only the first item + found was being returned. I fixed this by making a new callback + function for finding multiple dialogs that only returns + (CMP_MATCH) on a match allowing for multiple items to be + returned. Review: https://reviewboard.asterisk.org/r/776/ + +2010-07-26 19:51 +0000 [r279566] Paul Belanger + + * UPGRADE.txt, CHANGES, configs/logger.conf.sample: Add + documentation for FAX logger level. (closes issue #17715) + Reported by: vrban Patches: 17715.patch uploaded by pabelanger + (license 224) Tested by: vrban + +2010-07-26 19:18 +0000 [r279562] Tilghman Lesher + + * sounds/Makefile (removed), /, sounds/Makefile.380 (added), + configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381 + (added), configure.ac: Merged revisions 279561 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010) + | 2 lines Use a special Makefile for noobs who still have GNU + Make 3.80. ........ + +2010-07-26 16:04 +0000 [r279504] Mark Michelson + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + channels/sip/reqresp_parser.c: Allow for systems without locale + support to be usable. A recent change to SIP URI comparison code + added a locale-specific string comparison to the mix, and certain + systems do not support such functions. This fix allows for those + systems to still use Asterisk 1.8 (closes issue #17697) Reported + by: pprindeville Patches: asterisk-trunk-bugid17697.patch + uploaded by pprindeville (license 347) Tested by: mmichelson + +2010-07-26 15:43 +0000 [r279502] Sean Bright + + * autoconf/ast_ext_lib.m4, /: Merged revisions 279501 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon, + 26 Jul 2010) | 5 lines Expand the correct value within + AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth + ........ + +2010-07-26 03:27 +0000 [r279472] Tilghman Lesher + + * formats/format_sln16.c, formats/format_wav_gsm.c, + formats/format_siren7.c, formats/format_ilbc.c, + formats/format_vox.c, formats/format_pcm.c, + formats/format_h263.c, formats/format_g723.c, + formats/format_h264.c, formats/format_g726.c, + formats/format_jpeg.c, formats/format_siren14.c, + formats/format_gsm.c, formats/format_g719.c, + formats/format_g729.c, formats/format_sln.c, + formats/format_wav.c, formats/format_ogg_vorbis.c: Formats need + to load before apps, because some apps call + ast_format_str_reduce() at load time. + +2010-07-25 21:26 +0000 [r279442] Paul Belanger + + * tests/test_func_file.c: Add trailing backslash to silence warning + message. + +2010-07-25 18:21 +0000 [r279390-279410] Tilghman Lesher + + * cdr/cdr_odbc.c: Don't re-register CDR module on reload. (closes + issue #17304) Reported by: jnemeth Patches: + 20100507__issue17304.diff.txt uploaded by tilghman (license 14) + Tested by: jnemeth + + * main/logger.c: Don't assume qlog is open. (closes issue #17704) + Reported by: vrban Patches: issue17704.patch uploaded by + pabelanger (license 224) Tested by: vrban + +2010-07-24 23:58 +0000 [r279348] Bradley Latus + + * doc/asterisk.8: Minor update to man page + +2010-07-24 20:47 +0000 [r279273-279314] Paul Belanger + + * Makefile: Remove duplicate -c flag when using $(INSTALL) (closes + issue #17695) Reported by: pabelanger Patches: Makefile.diff + uploaded by pabelanger (license 224) + + * include/asterisk/netsock2.h: Check if ast_sockaddr is NULL then + return. (closes issue #17677) Reported by: outcast Patches: + issue0017677.patch uploaded by pabelanger (license 224) Tested + by: elguero + + * main/manager.c: Default sin_family to AF_INET for TCP / TLS + Bindaddress. Otherwise, 'manager show settings' will generate + errors if manager is not enabled. + +2010-07-23 22:20 +0000 [r279227] Richard Mudgett + + * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279207 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500 + (Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) + | 7 lines SIP promiscuous redirect could fail to dial the + redirect. The ast_channel was created with one variable to + ast_request() but the call to ast_call() that initiates the + outgoing call was using a different variable. The two variables + are not equivalent if the call_forward string included a channel + technology specifier. e.g., SIP/200 ........ ................ + +2010-07-12 Leif Madsen + + * Asterisk 1.8.0-beta1 Released. + +2010-07-23 18:56 +0000 [r279113] Tilghman Lesher + + * res/res_odbc.c: Silly 64-bit compilers (who uses 64-bit anyway?) + +2010-07-23 18:23 +0000 [r279056-279094] Russell Bryant + + * /: fix up properties on 1.8 branch + + * / (added): Create a branch for Asterisk 1.8. + + ___ _ _ _ _ ___ + / _ \ ___| |_ ___ _ __(_)___| | __ / | ( _ ) + | |_| / __| __/ _ \ '__| / __| |/ / | | / _ \ + | _ \__ \ || __/ | | \__ \ < | || (_) | + |_| |_|___/\__\___|_| |_|___/_|\_\ |_(_)___/ + +2010-07-23 17:05 +0000 [r278982-278985] Tilghman Lesher + + * autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged + revisions 278984 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010) + | 5 lines Establish a maximum version for openh323 (i.e. not + opal), because chan_h323 will fail to load, even if it links. + (issue #17679) Reported by: am ........ + + * /, main/asterisk.c: Merged revisions 278981 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010) + | 8 lines Avoid race with consolethread on shutdown (on parallel + processors). (closes issue #17080) Reported by: sybasesql + Patches: 20100721__issue17080.diff.txt uploaded by tilghman + (license 14) Tested by: sybasesql ........ + +2010-07-23 16:33 +0000 [r278980] Mark Michelson + + * channels/chan_sip.c, channels/sip/reqresp_parser.c, + channels/sip/include/reqresp_parser.h: SIP URI comparison fixes. + This initially was created to work around the issue of using a + string comparison instead of a binary comparison for IP + addresses. It evolved a bit when test cases were created and it + was discovered that comparison of URI parameters was not working + exactly as it should. sip_uri_cmp() and its helpers have been + moved to reqresp_parser.c and a new test has been added. (closes + issue #17662) Reported by: oej Review: + https://reviewboard.asterisk.org/r/792 + +2010-07-23 16:19 +0000 [r278957] Tilghman Lesher + + * include/asterisk/res_odbc.h, res/res_config_odbc.c, + configs/extconfig.conf.sample, CHANGES, main/config.c, + res/res_odbc.c, configs/res_odbc.conf.sample: Merge the realtime + failover branch + +2010-07-23 16:07 +0000 [r278947] Tzafrir Cohen + + * doc/asterisk.8: Some left-over hyphen-minus fixes in the man page + +2010-07-23 15:57 +0000 [r278944-278945] Russell Bryant + + * channels/chan_sip.c: ... just kidding. Enable SIP by default. :-) + + * channels/chan_sip.c: Disable SIP support by default for Asterisk + 1.8. + +2010-07-23 15:52 +0000 [r278943] Mark Michelson + + * addons/chan_ooh323.c: Well, who knew chan_ooh323 used udptl? I + sure didn't! + +2010-07-23 15:41 +0000 [r278942] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Rename sig_pri_pri to sig_pri_span. More descriptive of concept. + +2010-07-23 15:16 +0000 [r278908] Mark Michelson + + * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h, + channels/sip/include/sip.h: Allow IPv6 addresses for UDPTL + streams. Review: https://reviewboard.asterisk.org/r/795 + +2010-07-23 13:37 +0000 [r278875] Olle Johansson + + * res/res_config_ldap.c: Minor corrections to the LDAP realtime + driver Review: https://reviewboard.asterisk.org/r/798/ Thanks + Mark for a quick review! + +2010-07-23 13:26 +0000 [r278873] Paul Belanger + + * Makefile, agi/Makefile, sounds/Makefile: Portability updates for + Makefiles. When possible, use $(INSTALL). This allows us to use + the functionality within install for setting directory / file + permissions, a requirement for unprivileged installation. Also + move any directory we plan to create within the installdirs + macro. Plus various other formatting issues. (issue #17436) + Reported by: pabelanger Patches: non-root.patch.v8 uploaded by + pabelanger (license 224) Tested by: pabelanger Review: + https://reviewboard.asterisk.org/r/654/ + +2010-07-23 11:01 +0000 [r278809-278841] Alec L Davis + + * channels/chan_dahdi.c, channels/sig_analog.c: missed FXS kewl + start polarityswitch when finally on hook. (issue #17318) + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + channels/sig_analog.c, channels/sig_analog.h: Support FXS module + Polarity Reversal on remote party Answer and Hangup FXS lines + normally connect to a telephone. However, when FXS lines are + routed to an external PBX or Key System to act as "external" or + "CO" lines, it is extremely difficult, if not impossible for the + external PBX to know when the call has been disconnected without + receiving a polarity reversal on the line. Now using + answeronpolarityswitch and hanguponpolarityswitch keywords that + previously were used only for FXO ports, now applies like + functionality for an FXS port, but from the connected equipment's + point of view. (closes issue #17318) Reported by: armeniki + Patches: fxs_linepolarity.diff5.txt uploaded by alecdavis + (license 585) Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/797/ + +2010-07-22 21:16 +0000 [r278777] Richard Mudgett + + * channels/chan_dahdi.c: DNID not cleared when channel hang up + (Affects PRI and SS7) The "dahdi show channels" CLI command still + reports the DNID of the previous call even if the call is already + hang up. The "dahdi show channels" command of older releases + clear the DNID once the channel is hang up. Regression from the + sig_analog/sig_pri extraction from chan_dahdi. (closes issue + #17623) Reported by: klaus3000 Patches: issue17623.patch uploaded + by rmudgett (license 664) Tested by: rmudgett + +2010-07-22 19:45 +0000 [r278708] Jeff Peeler + + * main/xmldoc.c: Add method for finding XML doc files for systems + that don't support GLOB_BRACE. In particular, Solaris and perhaps + others do not support the above mentioned GNU extension. In this + case the paths are simply expanded without the braces and the + calls to glob are made separately. Note: I could not explain + memory allocation failures that were being reported from within + libxml itself when making calls to glob without using + GLOB_NOCHECK. This is the only reason why that flag is being + used. (closes issue #15402) Reported by: snuffy Patches: + bug_xmlpatt-v3.diff uploaded by snuffy (license 35), modified by + me + +2010-07-22 14:58 +0000 [r278620] Mark Michelson + + * main/channel.c, /: Merged revisions 278618 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul + 2010) | 13 lines Allow PLC to function properly when channels use + SLIN for audio. If a channel involved in a bridge was using SLIN + audio, then translation paths were not guaranteed to be set up + properly since in all likelihood the number of translation steps + was only 1. This patch enforces the transcode_via_slin behavior + if transcode_via_slin or generic_plc is enabled and one of the + formats to make compatible is SLIN. AST-352 ........ + +2010-07-22 14:56 +0000 [r278619] David Vossel + + * channels/chan_sip.c: update sip subscription debug message to a + warning message If the Expire header of a SUBSCRIBE is less that + our expiremin, a log warning will be displayed. + +2010-07-22 05:29 +0000 [r278579] Tilghman Lesher + + * include/asterisk/doxyref.h: Add the full current set of CDR + drivers + +2010-07-21 19:16 +0000 [r278539] David Vossel + + * tests/test_func_file.c: make func_file unit test's category + consistent with other tests + +2010-07-21 19:11 +0000 [r278538] Terry Wilson + + * channels/iax2-parser.h, include/asterisk/crypto.h, + main/aescrypt.c (removed), include/asterisk/aes_internal.h + (removed), funcs/func_aes.c, res/res_crypto.c, main/aestab.c + (removed), main/aesopt.h (removed), include/asterisk/aes.h + (removed), main/aeskey.c (removed), pbx/pbx_dundi.c, + channels/chan_iax2.c, res/res_crypto.exports.in, + pbx/dundi-parser.h: Remove built-in AES code and use optional_api + instead Review: https://reviewboard.asterisk.org/r/793/ + +2010-07-21 18:52 +0000 [r278536] David Vossel + + * channels/chan_sip.c: send "423 Interval too small" Response to + Subscribe with Expires less that min allowed [RFC3265]3.1.6.1.... + The notifier MAY also check that the duration in the "Expires" + header is not too small. If and only if the expiration interval + is greater than zero AND smaller than one hour AND less than a + notifier- configured minimum, the notifier MAY return a "423 + Interval too small" error which contains a "Min-Expires" header + field. The "Min- Expires" header field is described in SIP [1]. + +2010-07-21 17:44 +0000 [r278501] Tzafrir Cohen + + * channels/chan_dahdi.c, channels/sig_analog.c: Fix invalid test + for rxisoffhook in FXO channels This fixes some cases of no + outgoing calls on FXO before an incoming call. Remove an + unnecessary testing of an "off-hook" bit from DAHDI for FXO + (KS/GS) channels.In some cases the bit would not be initialized + properly before the first inbound call and thus prevent an + outgoing call. If those tests are actually required by anybody, + they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c + . (closes issue #14577) Reported by: jkroon Patches: + asterisk_chan_dahdi_hookstate_fix_trunk_new.diff uploaded by + frawd (license 610) Tested by: frawd Review: + https://reviewboard.asterisk.org/r/699/ + +2010-07-21 16:15 +0000 [r278465] Russell Bryant + + * res/res_timing_pthread.c: Use poll() instead of select() in + res_timing_pthread to avoid stack corruption. This code did not + properly check FD_SETSIZE to ensure that it did not try to + select() on fds that were too large. Switching to poll() removes + the limitation on the maximum fd value. (closes issue #15915) + Reported by: keiron (closes issue #17187) Reported by: Eddie + Edwards (closes issue #16494) Reported by: Hubguru (closes issue + #15731) Reported by: flop (closes issue #12917) Reported by: + falves11 (closes issue #14920) Reported by: vrban (closes issue + #17199) Reported by: aleksey2000 (closes issue #15406) Reported + by: kowalma (closes issue #17438) Reported by: dcabot (closes + issue #17325) Reported by: glwgoes (closes issue #17118) Reported + by: erikje possibly other issues, too ... + +2010-07-21 15:56 +0000 [r278463] Tilghman Lesher + + * apps/app_meetme.c: Ensure realtime conferences are treated the + same as static conferences when trying to find an empty one. + Also, parse the useropts properly, when retrieving from realtime, + and add them to the existing flags. (closes issue #17502) + Reported by: kenji Patches: 20100720__issue17502.diff.txt + uploaded by tilghman (license 14) Tested by: kenji + +2010-07-21 15:54 +0000 [r278426-278462] Matthew Nicholson + + * res/res_fax_spandsp.c: Properly show the current page being + transfered for 'fax show session' + + * channels/chan_sip.c: Properly set the port number for UDPTL media + sessions. + + * res/res_fax.c: Don't print failure status when the remote end + hangs up, it may not be an actual failure. + +2010-07-21 13:02 +0000 [r278425] Russell Bryant + + * main/features.c, UPGRADE.txt, configs/features.conf.sample: + Update documentation for 'comebacktoorigin' in featuers.conf. The + documentation for this option did not match the code. Fix that + along with some minor cleanups to the code along the way. + Document a slight change in behavior (to something that was + previously undocumented) in UPGRADE.txt. + +2010-07-21 06:45 +0000 [r278393] Tilghman Lesher + + * channels/chan_iax2.c: Change order so that it more closely + matches the related SIP command. (closes issue #17648) Reported + by: GMLudo Review: https://reviewboard.asterisk.org/r/789/ + +2010-07-21 03:53 +0000 [r278361] Jeff Peeler + + * channels/chan_dahdi.c: include stat.h for everybody, needed for + device2chan + +2010-07-20 23:23 +0000 [r278275-278307] Tilghman Lesher + + * res/res_config_pgsql.c, main/logger.c, CHANGES, + contrib/realtime/mysql/queue_log.sql (added), + configs/logger.conf.sample: Separate queue_log arguments into + separate fields, and allow the text file to be used, even when + realtime is used. (closes issue #17082) Reported by: coolmig + Patches: 20100720__issue17082.diff.txt uploaded by tilghman + (license 14) Tested by: coolmig + + * /, apps/app_voicemail.c: Merged revisions 278261 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 + Jul 2010) | 7 lines Delete IMAP messages in reverse order, to + ensure reordering after each expunge does not cause deletion of + the wrong message. (closes issue #16350) Reported by: noahisaac + Patches: 20100623__issue16350.diff.txt uploaded by tilghman + (license 14) ........ + +2010-07-20 22:38 +0000 [r278274] Richard Mudgett + + * channels/sig_pri.c: Reference correct struct member for unlikely + event PRI_EVENT_CONFIG_ERR. + +2010-07-20 22:26 +0000 [r278272] Tilghman Lesher + + * main/autoservice.c, /, main/features.c, + include/asterisk/channel.h: Merged revisions 278167 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 + Jul 2010) | 4 lines Do not queue up DTMF frames while a call is + on hold. (Fixes ABE-2110) ........ + +2010-07-20 21:41 +0000 [r278234] David Vossel + + * channels/chan_sip.c: fixes sip CANCEL race condition If Asterisk + sends a 4xx error and the other side sends a CANCEl before + receiving the 4xx and responding with the ACK, Asterisk will + process the CANCEL and send a 487 Request Terminated as a new + final response to the INVITE. Since we are issuing a new final + response to the INVITE, the old one must be pretend_acked else it + will keep retransmitting. + +2010-07-20 21:01 +0000 [r278168] Matthew Nicholson + + * res/res_fax.c: This commit contains several changes to the way + output channel variables are handled. FAX output channel + variables will now match the values reported by FAXOPT() and + should be set in all failure and success cases. This commit also + contains a few modifications to the way FAXOPT() variables are + populated in a few spots and fixes for some reference count leaks + of the session details structure in some failure cases. Also + found and fixed more cases where FAXOPT(status) may not have + gotten set. FAX-214 FAX-203 + +2010-07-20 19:35 +0000 [r278132] Tilghman Lesher + + * cel/cel_pgsql.c, cdr/cdr_sqlite3_custom.c, channels/chan_local.c, + res/res_timing_dahdi.c, cdr/cdr_adaptive_odbc.c, + res/res_calendar_caldav.c, formats/format_sln16.c, + formats/format_wav_gsm.c, channels/chan_iax2.c, main/config.c, + main/loader.c, res/res_rtp_multicast.c, channels/chan_dahdi.c, + res/res_smdi.c, channels/chan_skinny.c, + include/asterisk/module.h, formats/format_pcm.c, + channels/chan_alsa.c, formats/format_h263.c, res/res_curl.c, + cdr/cdr_odbc.c, formats/format_jpeg.c, res/res_speech.c, + formats/format_gsm.c, cdr/cdr_manager.c, formats/format_g719.c, + res/res_calendar_exchange.c, cel/cel_tds.c, formats/format_wav.c, + channels/chan_bridge.c, channels/chan_agent.c, + formats/format_ogg_vorbis.c, res/res_monitor.c, + res/res_calendar_ews.c, res/res_config_curl.c, + channels/chan_misdn.c, funcs/func_curl.c, + res/res_timing_kqueue.c, formats/format_g726.c, main/asterisk.c, + res/res_odbc.c, cel/cel_adaptive_odbc.c, res/res_calendar.c, + cel/cel_radius.c, channels/chan_multicast_rtp.c, + apps/app_meetme.c, formats/format_sln.c, res/res_musiconhold.c, + channels/chan_gtalk.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c, + res/res_jabber.c, res/res_config_sqlite.c, + formats/format_siren7.c, cdr/cdr_csv.c, formats/format_ilbc.c, + res/res_config_odbc.c, cel/cel_manager.c, cel/cel_custom.c, + cdr/cdr_sqlite.c, res/res_agi.c, res/res_timing_timerfd.c, + apps/app_confbridge.c, formats/format_h264.c, + res/res_config_ldap.c, addons/chan_mobile.c, + formats/format_siren14.c, cdr/cdr_custom.c, channels/chan_mgcp.c, + res/res_rtp_asterisk.c, res/res_config_pgsql.c, + res/res_calendar_icalendar.c, channels/chan_sip.c, + cdr/cdr_syslog.c, res/res_fax.c, res/res_crypto.c, + res/res_adsi.c, include/asterisk/config.h, pbx/pbx_lua.c, + channels/chan_console.c, apps/app_queue.c, cdr/cdr_tds.c, + res/res_srtp.c, channels/chan_jingle.c, formats/format_vox.c, + res/res_timing_pthread.c, channels/chan_h323.c, + cel/cel_sqlite3_custom.c, formats/format_g723.c, + funcs/func_devstate.c, formats/format_g729.c, + addons/res_config_mysql.c: Add load priority order, such that + preload becomes unnecessary in most cases + +2010-07-20 18:11 +0000 [r278051-278096] Russell Bryant + + * contrib/scripts/install_prereq: Add a package to install_prereq. + + * channels/chan_local.c: Only call ast_channel_cc_params_init() if + allocating a channel succeeds. + +2010-07-20 16:50 +0000 [r278024] Tilghman Lesher + + * main/manager.c, /: Merged revisions 278023 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) + | 7 lines Off-by-one error (closes issue #16506) Reported by: + nik600 Patches: 20100629__issue16506.diff.txt uploaded by + tilghman (license 14) ........ + +2010-07-19 21:07 +0000 [r277945] Jean Galarneau + + * /, main/features.c: Merged revisions 277906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) | + 7 lines Avoid trying to pickup a parked extension before the park + operation is completed. A crash could occur if the extension is + picked up while the parking extension is being announced. Testing + pu->notquiteyet while searching for a parked extension resolves + this crash. (ABE-2418) ........ + +2010-07-19 17:16 +0000 [r277872-277873] Mark Michelson + + * channels/chan_sip.c, configs/sip.conf.sample, + channels/sip/include/sip.h: Fix port setting of external address + in SIP. There are two changes here: 1. Since the externip setting + can now have a port attached to it, calling it "externip" is + misleading. The option is now documented and parsed as + "externaddr." This also extends to the "matchexterniplocally" + setting. It is now documented and parsed as + "matchexternaddrlocally." The old names for the options may still + be used, but they are no longer used in the sip.conf.sample file. + 2. If no port is set for the externaddr, and UDP is the transport + to be used, then we will set the port of the externaddr to that + of the udpbindaddr. This was how things worked prior to the IPv6 + merge, so this is a regression fix. (closes issue #17665) + Reported by: mmichelson Patches: 17665.diff#2 uploaded by + pprindeville (license 347) Tested by: pprindeville + + * tests/test_acl.c: Remove the fe80:1234::1234 test case from + test_acl.c The ACL test was failing on Mac OS X because it would + convert the above invalid link-local address into fe80::1234 + while reporting no error from getaddrinfo(). Linux does not do + this. + +2010-07-19 14:39 +0000 [r277837] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h: Fix regression with distinctive ring + detection. The issue here is that passing an array to a function + prohibits the ARRAY_LEN macro from returning the real size. To + avoid this the size is now defined and use of ARRAY_LEN is + avoided. (closes issue #15718) Reported by: alecdavis Patches: + bug15718.patch uploaded by jpeeler (license 325) + +2010-07-19 14:17 +0000 [r277814] Mark Michelson + + * include/asterisk/acl.h, main/netsock2.c, main/manager.c, + channels/chan_sip.c, channels/chan_skinny.c, tests/test_acl.c, + main/acl.c, include/asterisk/netsock2.h, configs/sip.conf.sample, + channels/chan_iax2.c: Make ACLs IPv6-capable. ACLs can now be + configured to match IPv6 networks. This is only relevant for ACLs + in chan_sip for now since other channel drivers do not support + IPv6 addressing. However, once those channel drivers are + outfitted to support IPv6 addressing, the ACLs will already be + ready for IPv6 support. https://reviewboard.asterisk.org/r/791 + +2010-07-17 17:42 +0000 [r277773-277775] Tilghman Lesher + + * /, autoconf/ast_func_fork.m4, configure, + include/asterisk/autoconfig.h.in: Merged revisions 277738 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) + | 5 lines Remove uclibc cross-compile triplet, as uclibc has a + working fork()... it's only uclinux that does not. (closes issue + #17616) Reported by: pprindeville ........ + + * res/res_config_pgsql.c, res/res_config_odbc.c, /, + include/asterisk/config.h, main/config.c, + addons/res_config_mysql.c: Merged revisions 277568 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 + Jul 2010) | 8 lines Since we split values at the semicolon, we + should store values with a semicolon as an encoded value. (closes + issue #17369) Reported by: gkservice Patches: + 20100625__issue17369.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ + +2010-07-17 13:10 +0000 [r277703] Russell Bryant + + * Makefile, configure, include/asterisk/autoconfig.h.in, + configure.ac, makeopts.in: Allow xmllint to be used for XML docs + validation. xmllint seems to be more commonly available since it + comes with libxml2. + +2010-07-17 00:03 +0000 [r277667] Bradley Latus + + * res/res_fax.c: Update res_fax.c to be a good xml citizen. (closes + issues #17667) Reported by: snuffy + +2010-07-16 23:23 +0000 [r277657] Tim Ringenbach + + * main/features.c: Merged revisions 277625 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul + 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on + attended transfer. ast_bridge_call() clears + AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer, + ast_bridge_call() is called for a second bridge on the same + channel, and it clears that flag, which still needs to get set + for when the original ast_bridge_call() gets control back and + checks it. Review: https://reviewboard.asterisk.org/r/741 + ........ + +2010-07-16 21:24 +0000 [r277530] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 277497 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul + 2010) | 4 lines Default to no udptl error correction so that + error correction will be disabled in the event that the remote + end indicates that they do not support the error correction mode + we requested. FAX-128 ........ + +2010-07-16 21:16 +0000 [r277488] Jeff Peeler + + * apps/app_queue.c: Fix reporting estimated queue hold time. Just + say the number of seconds (after minutes) rather than doing some + incorrect calculation with respect to minutes. (closes issue + #17498) Reported by: corruptor Patches: holdesecs_bug.diff + uploaded by corruptor (license 253) + +2010-07-16 20:35 +0000 [r277484] Tilghman Lesher + + * include/asterisk/sched.h, main/sched.c: Finally, a method that + really fixes the assertions in chan_iax2.c related to cancelling + lagid. No, replacing usleep(1) with sched_yield() did not have an + effect. + +2010-07-16 20:27 +0000 [r277467] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 277419 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 + Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when + reloading dahdi module During a reload, the priexclusive and + outsignalling parameters are not read in from the config file as + intended. Unfortunately, they get set to defaults as a result. + This patch makes sure that they do not get set to defaults during + a reload. (closes issue #17441) Reported by: mtryfoss Patches: + issue17441_v1.4.patch uploaded by rmudgett (license 664) + issue17441_v1.6.2.patch uploaded by rmudgett (license 664) + issue17441_trunk.patch uploaded by rmudgett (license 664) Tested + by: rmudgett ........ + +2010-07-16 20:25 +0000 [r277452] Tilghman Lesher + + * res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql + (added): Add documentation for MOH realtime fields + +2010-07-16 19:32 +0000 [r277409] Matthew Nicholson + + * tests/test_devicestate.c: updated devicestate test for device + state changes + +2010-07-16 19:22 +0000 [r277366] Jeff Peeler + + * apps/app_queue.c: Add missing handling for ringing state for use + with queue empty options. (closes issue #17471) Reported by: + jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056) + +2010-07-16 18:31 +0000 [r277331] Matthew Nicholson + + * main/pbx.c, /: Merged revisions 277327 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul + 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as + extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035) + Reported by: francesco_r Patches: pbx.c.patch uploaded by + viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar + ........ + +2010-07-16 18:14 +0000 [r277263] Tilghman Lesher + + * main/manager.c, /: Merged revisions 277261 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) + | 5 lines If variable gotten is not set, will segfault on + Solaris. (closes issue #17636) Reported by: bklang ........ + +2010-07-16 18:05 +0000 [r277250-277262] Matthew Nicholson + + * main/channel.c: Print f->subclass.integer instead of f->subclass. + (fix build breakage introduced in r277250) + + * main/channel.c, /: Merged revisions 277247 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul + 2010) | 4 lines For pass through DTMF tones, measure the actual + duration between the begin and end packets on the wire. If it is + detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf + emulation. AST-362 ........ + +2010-07-16 17:13 +0000 [r277183] Paul Belanger + + * /, apps/app_amd.c: Merged revisions 277182 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul + 2010) | 8 lines Total analysis time error with SIP and silence + suppression When using app_amd with SIP providers that have + silence suppression on, the iTotalTime count increases + exponentially. (closes issue #17656) Reported by: juls ........ + +2010-07-16 16:25 +0000 [r277175] Mark Michelson + + * channels/sip/reqresp_parser.c: Fix up some weird indentation + problems in reqresp_parser.c + +2010-07-16 15:20 +0000 [r277143] Sean Bright + + * main/translate.c: Avoid crashing when installing a duplicate + translation path with a lower cost. (closes issue #17092) + Reported by: moy Patches: translate.rev254273.patch uploaded by + moy (license 222) Tested by: moy + +2010-07-16 13:40 +0000 [r277103] Eliel C. Sardanons + + * CREDITS: Add Despegar.com (my main sponsor) to the CREDITS file. + +2010-07-16 13:32 +0000 [r276950-277102] Olle Johansson + + * main/dnsmgr.c, main/srv.c: Formatting changes + + * channels/chan_sip.c: Formatting fixes + + * configs/sip.conf.sample: Clarify syntax changes + + * CREDITS: Adding a few more to the list of CREDITS + + * channels/chan_sip.c: Formatting changes (guideline corrections) + Found a unused bag of curly brackets under my table. I always + wondered where they had gone. They where indeed needed in + chan_sip.c + + * CREDITS: Adding a few more credits + + * channels/chan_sip.c, doc/tex/channelvariables.tex, + configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Add + ability to configure the Max-Forwards header in the dialplan, as + well as in sip.conf configuration for the channel and for + devices. The Max-Forwards header is used to prevent loops in a + SIP network. Each intermediary, like SIP proxys and SBCs, + decrement this counter and detects when it reaches zero, at which + point the SIP request is nicely killed in a SIP-friendly way. + Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel + for the review and good advice. + + * CHANGES, apps/app_queue.c: Add a dialplan function to check if a + queue exists: QUEUE_EXISTS Review: + https://reviewboard.asterisk.org/r/777/ + +2010-07-16 06:04 +0000 [r276910-276911] Tilghman Lesher + + * res/res_jabber.c: And yet one more + + * res/res_jabber.c: "Item may be used uninitialized in this + function." + +2010-07-16 05:42 +0000 [r276909] Mark Michelson + + * channels/chan_sip.c: Fix reversed logic of if statement. Found + based on message from Philip Prindeville on the Asterisk + Developers mailing list. + +2010-07-16 05:38 +0000 [r276830-276908] Tilghman Lesher + + * configure, configure.ac: Detect the --dynamic-list flag a bit + better + + * configure, main/Makefile, configure.ac, makeopts.in: Fix build on + FreeBSD + + * tests/test_utils.c: Fix trunk build for Mac OS X 10.6 + + * contrib/realtime/mysql/iaxfriends.sql, + contrib/realtime/mysql/meetme.sql, + contrib/realtime/postgresql/realtime.sql, + contrib/realtime/mysql/sipfriends.sql: Allow ipaddress to contain + the maximum IPv6 address. Also, update meetme to the full list of + supported fields. + + * configure, autoconf/ast_gcc_attribute.m4: Quote AC_SUBST within + m4_ifval, so it does not get prematurely expanded. (closes issue + #17654) Reported by: pprindeville Patches: issue17654.diff + uploaded by qwell (license 4) Tested by: qwell, pprindeville + +2010-07-15 20:21 +0000 [r276788] Jeff Peeler + + * channels/chan_sip.c: Correct not setting the bindport before + attempting to open the socket. Related to changes from 276571, I + was accidentally testing with a port set in my configuration + causing me to miss this. Also moved the TCP handling as well to + occur before build_peer is called. + +2010-07-15 19:46 +0000 [r276731-276769] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, + include/asterisk/compat.h, configure.ac: Define LLONG_MAX on + systems that do not have it. (closes issue #17644) Reported by: + pprindeville + + * configure, main/Makefile, autoconf/ast_gcc_attribute.m4, + configure.ac, makeopts.in: Fix linking asterisk on CentOS 5, + which is using gcc 4.1.1. Gcc 4.1.2 has the real fix. Review: + https://reviewboard.asterisk.org/r/790/ + +2010-07-15 13:51 +0000 [r276653] Jeff Peeler + + * main/channel.c, /: Merged revisions 276652 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) + | 2 lines In a perfect world, the frame source would never be + NULL. In the meantime, don't crash when it is. ........ + +2010-07-15 12:21 +0000 [r276616] Russell Bryant + + * contrib/scripts/install_prereq: Add lua5.1 to the handy dandy + list of packages. + +2010-07-14 22:58 +0000 [r276571] Jeff Peeler + + * channels/chan_sip.c: Fix MWI notification transmission problems + over SIP. MWI updates were not being sent if no messages were + found in the event cache. This was corrected since a phone may + need to clear its MWI status configured previously from another + mailbox. Upon module or sip reload, MWI updates could not be sent + due to the sipsock socket not being set early enough in + reload_config. The code handling the descriptor assignment and + such has simply been moved before the call to build_peer. Issuing + a sip reload cleared the IP address of the peer, but skipped + checking the database for registration information. The database + is now checked both for sip reload and actually reloading the + module. If a transmission occurs before the do_monitor thread has + started, do not attempt to send a signal to it. (closes issue + #17398) Reported by: ip-rob + +2010-07-14 22:32 +0000 [r276570] Mark Michelson + + * res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c, + main/acl.c: Fix errors where incorrect address information was + printed. ast_sockaddr_stringiy_fmt (which is call by all + ast_sockaddr_stringify* functions) uses thread-local storage for + storing the string that it creates. In cases where + ast_sockaddr_stringify_fmt was being called twice within the same + statement, the result of one call would be overwritten by the + result of the other call. This usually was happening in + printf-like statements and was resulting in the same stringified + addressed being printed twice instead of two separate addresses. + I have fixed this by using ast_strdupa on the result of stringify + functions if they are used twice within the same statement. As + far as I could tell, there were no instances where a pointer to + the result of such a call were saved anywhere, so this is the + only situation I could see where this error could occur. + +2010-07-14 21:29 +0000 [r276531] Richard Mudgett + + * channels/chan_h323.c: Make compile again. + +2010-07-14 21:11 +0000 [r276490-276493] Tilghman Lesher + + * main/loader.c: Oops, merge reverted this fix. + + * include/asterisk/adsi.h, include/asterisk/agi.h, + include/asterisk/crypto.h, main/asterisk.dynamics, main/Makefile, + tests/test_utils.c, main/adsistub.c (removed), main/cryptostub.c + (removed), res/res_adsi.c, res/res_crypto.c, + res/res_crypto.exports.in (added), res/res_adsi.exports.in, + main/loader.c, include/asterisk/optional_api.h: Remove the old + stub files, preferring the optional_api method. (closes issue + #17475) Reported by: tilghman Review: + https://reviewboard.asterisk.org/r/695/ + +2010-07-14 20:15 +0000 [r276441] Kevin P. Fleming + + * main/loader.c: Don't try to call an embedded module's + backup_globals() function until after confirming it exists. + +2010-07-14 19:51 +0000 [r276439] David Vossel + + * channels/chan_sip.c: handle special case were "200 Ok" to pending + INVITE never receives ACK Unlike most responses, the 200 Ok to a + pending INVITE Request is acknowledged by an ACK Request. If the + ACK Request for this Response is not received the previous + behavior was to immediately destroy the dialog and hangup the + channel. Now in an effort to be more RFC compliant, instead of + immediately destroying the dialog during this special case, + termination is done with a BYE Request as the dialog is + technically confirmed when the 200 Ok is sent even if the ACK is + never received. The behavior of immediately hanging up the + channel remains. This only affects how dialog termination + proceeds for this one special case. RFC 3261 section 13.3.1.4 "If + the server retransmits the 2xx response for 64*T1 seconds without + receiving an ACK, the dialog is confirmed, but the session SHOULD + be terminated. This is accomplished with a BYE, as described in + Section 15." + +2010-07-14 16:58 +0000 [r276393] Richard Mudgett + + * channels/chan_vpb.cc, channels/chan_sip.c, + include/asterisk/channel.h, channels/sig_pri.c, + channels/chan_iax2.c, main/cel.c, channels/chan_oss.c, + main/channel.c, main/cdr.c, channels/chan_jingle.c, + channels/chan_usbradio.c, channels/chan_dahdi.c, + channels/chan_phone.c, channels/sig_analog.c, + channels/chan_misdn.c, channels/chan_skinny.c, + channels/chan_h323.c, res/snmp/agent.c, apps/app_amd.c, + funcs/func_callerid.c, channels/sig_ss7.c, channels/chan_mgcp.c: + Expand the caller ANI field to an ast_party_id Expand the ani + field in ast_party_caller and ast_party_connected_line to an + ast_party_id. This is an extension to the ast_callerid + restructuring patch in review: + https://reviewboard.asterisk.org/r/702/ Review: + https://reviewboard.asterisk.org/r/744/ + +2010-07-14 16:40 +0000 [r276392] David Vossel + + * channels/chan_sip.c: collapse debug code in retrans_pkt into + separate lines I've been working in this function a bunch lately, + and these huge debug strings are getting annoying. + +2010-07-14 16:39 +0000 [r276391] Richard Mudgett + + * res/snmp/agent.c: Make compile again. + +2010-07-14 16:36 +0000 [r276389] Jeff Peeler + + * channels/chan_sip.c: Do not skip sending MWI for a peer if an + address is defined. Really just a merge mistake from IPv6 + +2010-07-14 16:09 +0000 [r276349] Tim Ringenbach + + * cel/cel_pgsql.c, doc/tex/celdriver.tex, doc/tex/cdrdriver.tex: + Fix documentation for pgsql cel and cdr, and slightly improve + pgsql_cel. Change the documented pgsql schema to use "timestamp" + instead of "time", as the latter is only a time without a date. + Added some missing columns for cel's pgsql schema, and corrected + spelling on some others. Updated cel's uniqueid size to be the + same as the cdr. Added id column to cel's pgsql schema and + updated code to allow unknown columns to get their default value + instead of forcing 0 or empty string. Added microseconds to the + timestamp cel logs to pgsql. Review: + https://reviewboard.asterisk.org/r/734 + +2010-07-14 15:48 +0000 [r276347] Richard Mudgett + + * channels/chan_local.c, addons/chan_ooh323.c, + apps/app_alarmreceiver.c, channels/chan_iax2.c, main/cli.c, + channels/chan_dahdi.c, channels/sig_analog.c, + channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c, + channels/sig_analog.h, apps/app_amd.c, channels/sig_ss7.c, + apps/app_dial.c, main/pbx.c, apps/app_privacy.c, apps/app_fax.c, + channels/chan_agent.c, apps/app_disa.c, + include/asterisk/channel.h, apps/app_talkdetect.c, main/cel.c, + funcs/func_redirecting.c (removed), channels/chan_misdn.c, + apps/app_macro.c, apps/app_zapateller.c, apps/app_voicemail.c, + channels/chan_unistim.c, tests/test_substitution.c, + channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c, + apps/app_readexten.c, channels/chan_gtalk.c, apps/app_followme.c, + include/asterisk/callerid.h, main/cdr.c, main/channel.c, + channels/chan_phone.c, main/dial.c, apps/app_setcallerid.c, + apps/app_osplookup.c, main/manager.c, apps/app_minivm.c, + res/res_agi.c, main/app.c, apps/app_rpt.c, channels/chan_mgcp.c, + apps/app_parkandannounce.c, apps/app_while.c, + funcs/func_dialplan.c, channels/chan_sip.c, UPGRADE.txt, + channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c, + channels/chan_oss.c, channels/chan_usbradio.c, + channels/chan_jingle.c, funcs/func_blacklist.c, + apps/app_directed_pickup.c, main/file.c, + funcs/func_connectedline.c (removed), channels/chan_h323.c, + main/callerid.c, res/snmp/agent.c, apps/app_sms.c, + apps/app_stack.c, funcs/func_callerid.c: ast_callerid + restructuring The purpose of this patch is to eliminate struct + ast_callerid since it has turned into a miscellaneous collection + of various party information. Eliminate struct ast_callerid and + replace it with the following struct organization: struct + ast_party_name { char *str; int char_set; int presentation; + unsigned char valid; }; struct ast_party_number { char *str; int + plan; int presentation; unsigned char valid; }; struct + ast_party_subaddress { char *str; int type; unsigned char + odd_even_indicator; unsigned char valid; }; struct ast_party_id { + struct ast_party_name name; struct ast_party_number number; + struct ast_party_subaddress subaddress; char *tag; }; struct + ast_party_dialed { struct { char *str; int plan; } number; struct + ast_party_subaddress subaddress; int transit_network_select; }; + struct ast_party_caller { struct ast_party_id id; char *ani; int + ani2; }; The new organization adds some new information as well. + * The party name and number now have their own presentation value + that can be manipulated independently. ISDN supplies the + presentation value for the name and number at different times + with the possibility that they could be different. * The party + name and number now have a valid flag. Before this change the + name or number string could be empty if the presentation were + restricted. Most channel drivers assume that the name or number + is then simply not available instead of indicating that the name + or number was restricted. * The party name now has a character + set value. SIP and Q.SIG have the ability to indicate what + character set a name string is using so it could be presented + properly. * The dialed party now has a numbering plan value that + could be useful to have available. The various channel drivers + will need to be updated to support the new core features as + needed. They have simply been converted to supply current + functionality at this time. The following items of note were + either corrected or enhanced: * The CONNECTEDLINE() and + REDIRECTING() dialplan functions were consolidated into + func_callerid.c to share party id handling code. * CALLERPRES() + is now deprecated because the name and number have their own + presentation values. * Fixed app_alarmreceiver.c + write_metadata(). The workstring[] could contain garbage. It also + can only contain the caller id number so using + ast_callerid_parse() on it is silly. There was also a typo in the + CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() + on the channel's caller id number string. ast_callerid_parse() + alters the given buffer which in this case is the channel's + caller id number string. Then using ast_shrink_phone_number() + could alter it even more. * Fixed caller ID name and number + memory leak in chan_usbradio.c. * Fixed uninitialized char arrays + cid_num[] and cid_name[] in sig_analog.c. * Protected access to a + caller channel with lock in chan_sip.c. * Clarified intent of + code in app_meetme.c sla_ring_station() and dial_trunk(). Also + made save all caller ID data instead of just the name and number + strings. * Simplified cdr.c set_one_cid(). It hand coded the + ast_callerid_merge() function. * Corrected some weirdness with + app_privacy.c's use of caller presentation. Review: + https://reviewboard.asterisk.org/r/702/ + +2010-07-14 11:51 +0000 [r276268] Leif Madsen + + * /, configs/voicemail.conf.sample: Merged revisions 276267 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010) + | 1 line Update documentation for voicemail.conf externpass + option. ........ + +2010-07-13 22:18 +0000 [r276219] David Vossel + + * channels/chan_sip.c, channels/sip/include/sip.h: chan_sip: RFC + compliant retransmission timeout Retransmission of packets should + not be based on how many packets were sent, but instead on a + timeout period. Depending on whether or not the packet is for a + INVITE or NON-INVITE transaction, the number of packets sent + during the retransmission timeout period will be different, so + timing out based on the number of packets sent is not accurate. + This patch fixes this by removing the retransmit limit and only + stopping retransmission after a timeout period is reached. By + default this timeout period is 64*(Timer T1) for both INVITE and + non-INVITE transactions. For more information on sip timer values + refer to RFC3261 Appendix A. Review: + https://reviewboard.asterisk.org/r/749/ + +2010-07-13 21:42 +0000 [r276206] Terry Wilson + + * channels/sip/include/dialog.h, channels/chan_sip.c: Revert early + destruction of RTP sessions Some code improperly assumes that the + sessions are still there, so revert the change until I can find + all of them and fix them. + +2010-07-13 19:15 +0000 [r276124-276127] Russell Bryant + + * /: Recorded merge of revisions 276126 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r276126 | russell | 2010-07-13 14:14:54 -0500 (Tue, 13 Jul 2010) + | 2 lines Only reset a CDR that exists. ........ + + * /, main/features.c: Merged revisions 276123 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010) + | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr + instead of peer_cdr in the last commit). ........ + +2010-07-13 19:05 +0000 [r276114-276122] Tilghman Lesher + + * funcs/func_env.c: Oops, XML documentation fix. + + * funcs/func_env.c: It really cannot fail in the places below, but + the stupid compiler doesn't know that. + + * funcs/func_env.c: Weird compiler error on Bamboo. + + * funcs/func_env.c, CHANGES, tests/test_func_file.c (added): FILE() + now supports line-mode and writing (altering) files. (closes + issue #16461) Reported by: skyman Patches: + 20100622__issue16461.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman Review: + https://reviewboard.asterisk.org/r/737/ + +2010-07-13 17:37 +0000 [r276074] Jeff Peeler + + * /, apps/app_meetme.c: Merged revisions 275773 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) + | 12 lines Make user removals and traversals thread safe in + meetme. Race conditions present in meetme involving the user list + where a lack of locking has the potential for a user to be + removed during a traversal or as in the case of the reporter + after checking if the list is empty could cause a crash. Fixing + this was done by convering the userlist to an ao2 container. + (closes issue #17390) Reported by: Vince Review: + https://reviewboard.asterisk.org/r/746/ ........ + +2010-07-13 17:11 +0000 [r275998] Terry Wilson + + * channels/sip/include/dialog.h, channels/chan_sip.c: Destroy RTP + fds when we schedule final dialog destruction Since we are only + keeping the dialog around for retransmissions at this point and + there is no possibility that we are still handling RTP, go ahead + and destroy the RTP sessions. Keeping them alive for 32 past when + they are used is unnecessary and can lead to problems with having + too many open file descriptors, etc. + +2010-07-13 16:53 +0000 [r275995] Russell Bryant + + * /, main/features.c: Merged revisions 275994 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010) + | 14 lines Access peer->cdr directly instead of through a saved + off reference. At this point in the code, it is possible that + peer_cdr may be invalid. Specifically, in the blind transfer + code, CDRs are swapped between channels. So, peer_cdr is no + longer == peer->cdr. The scenario that exposed a crash in this + code was a blind transfer that hit the system call limit, causing + the transferee channel to get destroyed after the transfer + attempt failed. Even if it succeeds and this code doesn't crash, + this code was still trying to reset a CDR on a channel that was + now owned by a different thread, which is a BadThing(tm). + (ABE-2417) ........ + +2010-07-13 14:48 +0000 [r275910] Tilghman Lesher + + * contrib/scripts/realtime_pgsql.sql (removed), + contrib/scripts/iax-friends.sql (removed), /, + contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql + (removed), contrib/realtime (added), contrib/realtime/postgresql, + contrib/realtime/postgresql/realtime.sql, contrib/realtime/mysql, + contrib/realtime/oracle, contrib/scripts/sip-friends.sql + (removed), contrib/realtime/mysql/sipfriends.sql, + contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql + (removed), contrib/realtime/mysql/meetme.sql, + contrib/realtime/sqlserver: Merged revisions 275909 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13 + Jul 2010) | 2 lines Move SQL scripts into their own + database-specific directories. ........ + +2010-07-13 11:41 +0000 [r275863] Russell Bryant + + * configs/voicemail.conf.sample, + contrib/scripts/voicemailpwcheck.py (added): Add example script + for use with the externpasscheck voicemail.conf option. (closes + issue #17628) Reported by: lmadsen Tested by: russell, lmadsen + Review: https://reviewboard.asterisk.org/r/774/ + +2010-07-12 23:27 +0000 [r275816] Terry Wilson + + * channels/chan_sip.c: Don't try to ref authpeer when it isn't set + +2010-07-12 17:54 +0000 [r275725] Richard Mudgett + + * main/channel.c: Add which ITU spec specifies the numbering plan. + +2010-07-12 17:21 +0000 [r275682] Jeff Peeler + + * main/channel.c, /: Merged revisions 275665 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010) + | 11 lines Change ast_write to not stop generator when called + from ast_prod. For SIP channels configured with the + progressinband option on, the ringback was being immediately + stopped. This problem was due to ast_prod being moved for a + deadlock fix in 259858. Prodding the channel after setting up the + generator triggered the check in ast_write to stop the generator. + The fix here should write the frame the same as was done before + the call to ast_prod was moved. (closes issue #17372) Reported + by: tech_admin ........ + +2010-07-12 15:37 +0000 [r275626] Leif Madsen + + * cdr/cdr_pgsql.c: cdr_pgsql does not detect when a table is found. + This change adds an ERROR message to let you know when a failure + exists to get the columns from the pgsql database, which + typically means that the table does not exist. (closes issue + #17478) Reported by: kobaz Patches: cdr_pgsql.patch uploaded by + kobaz (license 834) Tested by: kobaz, russell, lmadsen + +2010-07-12 14:55 +0000 [r275587] Mark Michelson + + * main/netsock2.c: Allow netsock2.c to compile on systems that do + not define AI_NUMERICSERV. (closes issue #17617) Reported by: + pprindeville Patches: asterisk-trunk-bugid17617.patch uploaded by + pprindeville (license 347) + +2010-07-12 04:16 +0000 [r275551] TransNexus OSP Development + + * configs/osp.conf.sample, apps/app_osplookup.c: Added support for + indirect work mode. + +2010-07-10 20:49 +0000 [r275509] Eliel C. Sardanons + + * apps/app_meetme.c: When creating a conference for a unit test, it + is not mandatory to open a dahdi pseudo channel, so if we fail + doing it, continue creating the conference. + +2010-07-10 14:48 +0000 [r275424-275467] Russell Bryant + + * CHANGES: Make indentation consistent, move some queue features to + the queue section. + + * CREDITS, channels/chan_unistim.c, configs/unistim.conf.sample, + CHANGES: Add support for devices with less than 3 lines on the + LCD. (closes issue #17600) Reported by: minaguib Patches: + ast_unistim_height_v2.patch uploaded by minaguib (license 1078) + Tested by: minaguib + + * main/features.c, configs/features.conf.sample: Fix some issues + related to dynamic feature groups in features.conf. The bridge + handling code did not properly consider feature groups when + setting parameters that would affect whether or not a native + bridge would be attempted. If DYNAMIC_FEATURES only include a + feature group, a native bridge would occur that may prevent + features from working. Fix a bug in verbose output that would + show the key mapping as empty if it was using the default mapping + and not a custom mapping in the feature group. Add feature groups + to the output of "features show". Adjust the feature execution + logic to match that of the logic when executing a feature that + was not configured through a feature group. Update + features.conf.sample to show that an '=' is still required if + using the default key mapping from [applicationmap]. Finally, + clean up a little bit of formatting to better coform to coding + guidelines while in the area. (closes issue #17589) Reported by: + lmadsen Patches: issue_17589.rev4.txt uploaded by russell + (license 2) Tested by: russell, lmadsen + +2010-07-09 20:58 +0000 [r275385] Mark Michelson + + * channels/chan_sip.c: Fix error in parsing SIP registry strings + from ASTdb. It was essentially an off-by-one error. The easiest + way to fix this was to use the handy-dandy + AST_NONSTANDARD_RAW_ARGS macro to parse the pieces of the + registration string out. Tested and it works wonderfully. + +2010-07-09 20:01 +0000 [r275312] Tilghman Lesher + + * apps/app_meetme.c, channels/chan_iax2.c: Get more information + about the Bamboo test failures + +2010-07-09 19:58 +0000 [r275309-275310] Russell Bryant + + * main/features.c: Add missing ao2_iterator_destroy(). + + * apps/app_voicemail.c: Fix compile error. + +2010-07-09 19:46 +0000 [r275308] Mark Michelson + + * channels/chan_sip.c: Fix port parsing in check_via. If a Via + header contained an IPv6 address, we would not properly parse the + port. We would instead get the information after the first colon + in the address. (closes issue #17614) Reported by: oej Patches: + diff uploaded by sperreault (license 252) + +2010-07-09 19:32 +0000 [r275307] Paul Belanger + + * CHANGES, apps/app_voicemail.c: Include rdnis in msgXXXX.txt file. + (closes issue #17566) Reported by: outcast Patches: + voicemail-rdnis.patch uploaded by outcast (license 1071) Tested + by: outcast + +2010-07-09 19:29 +0000 [r275294] Mark Michelson + + * channels/chan_sip.c: Fix an issue where the port for p->ourip was + being set to 0. This should fix all the CDR tests that were not + passing. When they would originate a call, all fields in the + INVITE that contained the source port would have the port set to + 0. Most troubling of these was the Contact header. Tests are + passing locally now and should also pass on the bamboo build + agents. + +2010-07-09 19:21 +0000 [r275249] Paul Belanger + + * /, channels/chan_sip.c: Merged revisions 275241 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul + 2010) | 8 lines Fix logging message for stale nonce. (closes + issue #17582) Reported by: kenner Patches: chan_sip.c.diff + uploaded by kenner (license 1040) Tested by: lmadsen ........ + +2010-07-09 18:55 +0000 [r275227] Tilghman Lesher + + * apps/app_meetme.c, channels/chan_iax2.c: Weird, no output and + Bamboo still fails... + +2010-07-09 18:24 +0000 [r275186] Matthew Nicholson + + * /, main/loader.c: Merged revisions 275182 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul + 2010) | 2 lines give a better error message when attempting to + unload a module that is not loaded ........ + +2010-07-09 18:21 +0000 [r275172] Tilghman Lesher + + * apps/app_meetme.c, channels/chan_iax2.c: Add some diagnostic + feedback to our data tests + +2010-07-09 18:11 +0000 [r275147] Russell Bryant + + * configs/features.conf.sample: Move parking lot sample config out + from the middle of dynamic features sample config. + +2010-07-09 17:50 +0000 [r275144] Matthew Nicholson + + * /, main/loader.c: Merged revisions 275143 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul + 2010) | 2 lines don't unload modules that returned + AST_MODULE_LOAD_DECLINE when they were loaded ........ + +2010-07-09 17:00 +0000 [r275105] Tilghman Lesher + + * main/netsock2.c, tests/test_substitution.c, tests/test_heap.c, + apps/app_meetme.c, tests/test_gosub.c, funcs/func_strings.c, + tests/test_event.c, channels/sip/reqresp_parser.c, + channels/chan_iax2.c, tests/test_stringfields.c, + tests/test_time.c, tests/test_devicestate.c, tests/test_utils.c, + main/features.c, res/res_agi.c, include/asterisk/netsock2.h, + tests/test_astobj2.c, channels/chan_sip.c, + tests/test_ast_format_str_reduce.c, tests/test_app.c, + funcs/func_math.c, include/asterisk/channel.h, + tests/test_sched.c, tests/test_pbx.c, tests/test_strings.c, + main/data.c, tests/test_skel.c, tests/test_acl.c, + channels/sip/dialplan_functions.c, tests/test_aoc.c, main/test.c, + channels/sip/config_parser.c, res/res_timing_kqueue.c, + apps/app_voicemail.c: Kill some startup warnings and errors and + make some messages more helpful in tracking down the source. + +2010-07-09 16:39 +0000 [r275104] Mark Michelson + + * channels/chan_sip.c: Return logic of sip_debug_test_addr() to its + original functionality. + +2010-07-09 16:05 +0000 [r275028] Matthew Nicholson + + * apps/app_dial.c, /: Merged revisions 275027 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul + 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels + going into the pbx via the G option in app_dial (closes issue + #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff + uploaded by mnicholson (license 96) Tested by: jamicque, + mnicholson ........ + +2010-07-09 15:35 +0000 [r275022] Russell Bryant + + * include/asterisk/test.h, /, main/test.c: Merged revisions 275021 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) + | 4 lines Document that a leading and trailing slash is expected + for test categories. Also, emit a warning if a test is registered + without one of these. ........ + +2010-07-09 14:27 +0000 [r274984] Mark Michelson + + * channels/sip/reqresp_parser.c: Fix sip_uri_parse test comparison. + Part of the change with the IPv6 changes is to treat a host:port + as a single 'domain' entity. This test was not updated to have + the correct expectation after calling parse_uri(). + +2010-07-09 13:30 +0000 [r274909-274947] + + * channels/chan_sip.c: Copy the address into the peer structure + after we set the default port + + * main/netsock2.c: Sadly we can't dereference a pointer cast and + use it as an lvalue without getting this warning (at least with + gcc 4.4.4): netsock2.c:492: warning: dereferencing pointer + ‘({anonymous})’ does break strict-aliasing rules So we're back to + using memcpy()... + +2010-07-09 12:48 +0000 [r274907] Russell Bryant + + * include/asterisk/indications.h: Extend length limit on country + name in indications.conf. + +2010-07-09 11:06 +0000 [r274866] Olle Johansson + + * configs/cdr.conf.sample, cdr/cdr_csv.c: Make it possible to + disable individual cdr files per accountcode in cdr_csv Review: + https://reviewboard.asterisk.org/r/678/ + +2010-07-08 23:46 +0000 [r274827-274828] Richard Mudgett + + * channels/chan_jingle.c, channels/chan_h323.c, + channels/chan_gtalk.c: Fix calls of ast_sockaddr_from_sin() from + IPv6 integration. + + * addons/chan_ooh323.c: Fix compile of chan_ooh323.c from IPv6 + integration. + +2010-07-08 22:16 +0000 [r274783-274786] Mark Michelson + + * /: And the automerge property. + + * /: Delete properties I merged during v6-new merge. + + * channels/chan_unistim.c, include/asterisk/acl.h, main/netsock2.c + (added), channels/sip/include/dialog.h, + channels/chan_multicast_rtp.c, addons/chan_ooh323.c, + main/rtp_engine.c, /, channels/sip/reqresp_parser.c, + include/asterisk/tcptls.h, channels/chan_gtalk.c, + channels/chan_iax2.c, main/config.c, res/res_rtp_multicast.c, + main/manager.c, channels/chan_skinny.c, + channels/sip/include/globals.h, main/http.c, main/app.c, + include/asterisk/netsock2.h (added), apps/app_externalivr.c, + configs/sip.conf.sample, include/asterisk/rtp_engine.h, + channels/sip/include/sip.h, channels/chan_mgcp.c, + channels/sip/include/reqresp_parser.h, res/res_rtp_asterisk.c, + main/dnsmgr.c, channels/chan_sip.c, include/asterisk/config.h, + main/acl.c, CHANGES, channels/chan_jingle.c, main/tcptls.c, + channels/sip/dialplan_functions.c, channels/chan_h323.c, + include/asterisk/dnsmgr.h: Add IPv6 to Asterisk. This adds a + generic API for accommodating IPv6 and IPv4 addresses within + Asterisk. While many files have been updated to make use of the + API, chan_sip and the RTP code are the files which actually + support IPv6 addresses at the time of this commit. The way has + been paved for easier upgrading for other files in the near + future, though. Big thanks go to Simon Perrault, Marc Blanchet, + and Jean-Philippe Dionne for their hard work on this. (closes + issue #17565) Reported by: russell Patches: + asteriskv6-test-report.pdf uploaded by russell (license 2) + Review: https://reviewboard.asterisk.org/r/743 + +2010-07-08 22:05 +0000 [r274773-274782] Richard Mudgett + + * main/channel.c: Generate a correct AstData string for + ast_callerid.cid_ton + + * main/channel.c: Fix trunk compile. + +2010-07-08 14:48 +0000 [r274727] Eliel C. Sardanons + + * main/pbx.c, channels/chan_sip.c, apps/app_meetme.c, + include/asterisk/indications.h, channels/chan_agent.c, + include/asterisk/channel.h, include/asterisk/cdr.h, + include/asterisk/data.h, channels/chan_iax2.c, apps/app_queue.c, + main/indications.c, main/channel.c, main/cdr.c, + channels/chan_dahdi.c, main/data.c, res/res_odbc.c, + apps/app_voicemail.c: Implement AstData API data providers as + part of the GSOC 2010 project, midterm evaluation. Review: + https://reviewboard.asterisk.org/r/757/ + +2010-07-07 20:09 +0000 [r274686] David Vossel + + * channels/chan_sip.c: Fixes some ref count issues introduced by + r274539 + +2010-07-07 18:32 +0000 [r274595-274639] Richard Mudgett + + * channels/chan_dahdi.c: Add missing conditional around chan_dahdi + mfcr2_skip_category config parameter. + + * channels/chan_dahdi.c, /: Merged revisions 274579 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 + Jul 2010) | 1 line Close the DAHDI FD on error when processing + chan_dahdi toneduration config parameter. ........ + +2010-07-07 16:40 +0000 [r274540] Matthew Nicholson + + * res/res_fax.c: Set proper FAXOPT(status), FAXOPT(statusstr), and + FAXOPT(error) values where possible. Previously some failure + cases did not result in proper FAXOPT values. FAX-203 + +2010-07-07 16:21 +0000 [r274539] Mark Michelson + + * channels/chan_sip.c: Use the relatedpeer field of a sip_pvt + during INVITE processing. Review: + https://reviewboard.asterisk.org/r/629 + +2010-07-07 07:07 +0000 [r274492] TransNexus OSP Development + + * configs/osp.conf.sample, doc/osp.txt: Changed OSP TCP port from + 1080 to 5045. + +2010-07-07 06:32 +0000 [r274418-274491] Tilghman Lesher + + * CHANGES, apps/app_voicemail.c: Also run the externnotify script + when the pollmailboxes thread notices a change. + + * /, configs/say.conf.sample: Merged revisions 274417 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 + Jul 2010) | 8 lines Correct how 100, 200, 300, etc. is said. Also + add the crazy British numbers. (closes issue #16102) Reported by: + Delvar Patches: say.conf.fix.patch uploaded by Delvar (license + 908) (plus a few additional fixes and simplifications by me) + ........ + +2010-07-06 22:23 +0000 [r274316] Jeff Peeler + + * /, configs/sip.conf.sample: Merged revisions 274283 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 + Jul 2010) | 7 lines Correct sip.conf.sample comments for + prematuremedia option. (closes issue #17513) Reported by: festr + Patches: patch uploaded by festr (license 443) ........ + +2010-07-06 22:15 +0000 [r274284] Terry Wilson + + * /, channels/chan_sip.c, UPGRADE.txt: Merged revisions 274280 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) + | 9 lines Add option to not do a call forward on 482 Loop + Detected Asterisk has always set up a forwarded call when + receiving a 482 Loop Detected. This prevents handling the call + failure by just continuing on in the dialplan. Since this would + be a change in behavior, the new option to disable this behavior + is forwardloopdetected which defaults to 'yes'. Review: + https://reviewboard.asterisk.org/r/764/ ........ (no option for + trunk, just changing the behavior) + +2010-07-06 22:09 +0000 [r274281] Tilghman Lesher + + * channels/chan_dahdi.c: Status shows all non-CRC4 lines as + "yellow", even if "yellow" was not in the bitfield. + +2010-07-06 19:53 +0000 [r274243] Matthew Nicholson + + * res/res_fax.c: Properly detect and report invalid maxrate and + maxrate values in the FAXOPT dialplan function. Also make + fax_rate_str_to_int() return an unsigned int and return 0 instead + of -1 in the event of an error. FAX-202 + +2010-07-06 14:31 +0000 [r274164] Mark Michelson + + * res/res_rtp_asterisk.c, /: Merged revisions 274157 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, + 06 Jul 2010) | 16 lines Fix problem with RFC 2833 DTMF not being + accepted. A recent check was added to ensure that we did not + erroneously detect duplicate DTMF when we received packets out of + order. The problem was that the check did not account for the + fact that the seqno of an RTP stream will roll over back to 0 + after hitting 65535. Now, we have a secondary check that will + ensure that the seqno rolling over will not cause us to stop + accepting DTMF. (closes issue #17571) Reported by: mdeneen + Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license + 60) Tested by: richardf, maxochoa, JJCinAZ ........ + +2010-07-06 06:01 +0000 [r274053] Tilghman Lesher + + * main/pbx.c: Uh, yeah. + +2010-07-05 13:53 +0000 [r273886] Paul Belanger + + * /, main/config.c: Merged revisions 273884 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul + 2010) | 8 lines Remove extra line breaks from 'core show config + mappings' (closes issue #17583) Reported by: pabelanger Patches: + issue17583.patch uploaded by pabelanger (license 224) Tested by: + lmadsen ........ + +2010-07-03 02:36 +0000 [r273714-273830] Tilghman Lesher + + * channels/chan_local.c, /, channels/chan_agent.c, + channels/chan_h323.c, include/asterisk/lock.h: Merged revisions + 273793 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) + | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock + fails, to help catch potentially large software bugs. (closes + issue #17407) Reported by: pdf Patches: + 20100527__issue17407.diff.txt uploaded by tilghman (license 14) + Review: https://reviewboard.asterisk.org/r/751/ ........ + + * main/autoservice.c, /: Merged revisions 273717 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) + | 8 lines Autoservice loop optimization causes a busy loop, when + channels are serviced while in hangup. (closes issue #17564) + Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt + uploaded by tilghman (license 14) Tested by: ramonpeek ........ + + * apps/app_queue.c: The switch fallthrough could create some + errorneous situations, so best to force directly to the default + case. + +2010-07-02 15:57 +0000 [r273641] Tzafrir Cohen + + * channels/chan_dahdi.c, channels/chan_misdn.c, + channels/chan_sip.c, main/say.c, main/fixedjitterbuf.c, + res/res_agi.c, channels/chan_h323.c, main/utils.c, + channels/chan_iax2.c, addons/chan_mobile.c, apps/app_rpt.c, + channels/chan_mgcp.c, main/xmldoc.c, apps/app_voicemail.c, + apps/app_while.c: Fix various typos reported by Lintian (Also fix + the typos in the comments) + +2010-07-01 22:16 +0000 [r273566] Russell Bryant + + * /, main/datastore.c: Merged revisions 273565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) + | 7 lines Don't return a partially initialized datastore. If + memory allocation fails in ast_strdup(), don't return a partially + initialized datastore. Bad things may happen. (related to + ABE-2415) ........ + +2010-07-01 20:28 +0000 [r273522] Jeff Peeler + + * /, apps/app_meetme.c: Merged revisions 273474 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) + | 14 lines Allow admin user to join conference without using + admin mode and no user pin. Configuring the conference in + meetme.conf like the following: conf => 2345,,6666 did not prompt + for pin when used without admin mode. This meant that the + conference could not be joined as an admin even if the user knew + the correct pin. The original bug report was submitted claiming + that the blank user pin should deny entry into the conference. I + think a better way to handle this would be with a feature + enhancement that used the following syntax: conf => 2345,X,6666 - + where X denotes no acceptable pin allowed (closes issue #15704) + Reported by: modelnine ........ + +2010-07-01 19:34 +0000 [r273464] Matthew Nicholson + + * res/res_fax.c: Properly handle failures of fax->start_session() + FAX-177 + +2010-07-01 16:40 +0000 [r273427] David Vossel + + * channels/chan_sip.c, channels/sip/include/sip.h: correct handling + of get_destination return values A failure when calling the + get_destination can mean multiple things. If the extension is not + found, a 404 error is appropriate, but if the URI scheme is + incorrect, a 404 is not approperiate. This patch adds the + get_destination_result enum to differentiate between these and + other failure types. The only logical difference in this patch is + that we now send a "416 Unsupported URI scheme" response instead + of a "404" when the scheme is not recognized. This indicates to + the initiator of the INVITE to retry the request with a correct + URI. + +2010-07-01 15:12 +0000 [r273355] Jeff Peeler + + * /, apps/app_meetme.c: Merged revisions 273354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) + | 12 lines Ensure channel placed in meetme in ringing state is + properly hung up. An outgoing channel placed in meetme while + still ringing which was then hung up would not exit meetme and + the channel was not properly destroyed. Specifically checking for + this scenario by looking at the appropriate control frames + resolves the issue. (closes issue #15871) Reported by: Ivan + Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan + (license 229) ........ + +2010-07-01 14:37 +0000 [r273270-273352] Matthew Nicholson + + * main/manager.c: Fixed whitespace problems + + * main/manager.c: Altered my comment about TCP_NODELAY + + * addons/chan_mobile.c: Don't free written frames in chan_mobile's + mbl_write() function. (closes issue #16430) Reported by: azbest + Tested by: azbest + + * main/manager.c: Set TCP_NODELAY on manager TCP sockets to prevent + delays on outgoing packets. This regression was introduced in + r48338. AST-359 + +2010-06-30 17:28 +0000 [r273233] Paul Belanger + + * res/res_rtp_asterisk.c: Fix rt(c)p set debug ip taking wrong + argument Also clean up some coding errors. (closes issue #17469) + Reported by: wdoekes Patches: astsvn-rtp-set-debug-ip.patch + uploaded by wdoekes (license 717) Tested by: wdoekes, pabelanger + +2010-06-30 17:17 +0000 [r273197-273198] Richard Mudgett + + * include/asterisk/config.h: Remove unnecessary if test in + CV_DSTR() + + * include/asterisk/config.h: Misc doxygen cleanup in config.h + +2010-06-30 01:07 +0000 [r273054-273144] Tilghman Lesher + + * main/manager.c: Permission checking for the system application is + backwards. (closes issue #17550) Reported by: kenner Patches: + manager.c.diff uploaded by kenner (license 1040) Tested by: + kenner + + * main/config.c: Don't attempt to proceed if our internal parser + indicates an invalid file. (closes issue #17560) Reported by: + Nick_Lewis + + * /, channels/chan_sip.c: Merged revisions 273060 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) + | 10 lines Allow the "useragent" value to be restored into memory + from the realtime backend. This value is purely informational. It + does not alter configuration at all. (closes issue #16029) + Reported by: Guggemand Patches: realtime-useragent.patch uploaded + by Guggemand (license 897) Tested by: Guggemand ........ + + * /: Recorded merge of revisions 273057 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010) + | 4 lines _Really_ skip the channel... don't just retry for + another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........ + + * configure, include/asterisk/autoconfig.h.in, configure.ac: + Exclude libical for insufficient versions. + + * main/pbx.c: Send DialPlanComplete as a response, not as a + separate event. Otherwise, it goes to all manager sessions and + may exclude the current session, if the Events mask excludes it. + (closes issue #17504) Reported by: rrb3942 Patches: + showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested + by: rrb3942 + +2010-06-29 20:44 +0000 [r272981] David Vossel + + * channels/chan_sip.c: send a 400 Bad Request on malformed sip + request RFC 2361 section 24.4.1 send a 400 Bad Request if the + request can not be understood due to malformed syntax. Currently + we simply ignore a packet with a missing callid, to, from, or via + header. Instead of ignoring we now send the 400 Bad request. + +2010-06-28 21:50 +0000 [r272923-272926] Tilghman Lesher + + * /, main/asterisk.c: Merged revisions 272925 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) + | 8 lines Don't change ownership/group/permissions on run + directory, if it already exists. (closes issue #17076) Reported + by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by + tilghman (license 14) Tested by: stuarth ........ + + * /, main/config.c: Merged revisions 272921-272922 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 + Jun 2010) | 8 lines Change the way that we read include files, to + accommodate for changes in GCC 4.4. (closes issue #17472) + Reported by: seandarcy Patches: config2.patch uploaded by nivan + (license 1066) Tested by: nivan ........ r272922 | tilghman | + 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim + trailing blanks on #includes ........ + +2010-06-28 18:38 +0000 [r272880] David Vossel + + * channels/chan_sip.c, channels/sip/reqresp_parser.c, + channels/sip/include/sip.h, + channels/sip/include/reqresp_parser.h: rfc compliant sip option + parsing + new unit test RFC 3261 section 8.2.2.3 states that if + any unsupported options are found in the Require header field, a + "420 (Bad Extension)" response should be sent with an Unsupported + header field containing only the unsupported options. This is not + currently being done correctly. Right now, if Asterisk detects + any unsupported sip options in a Require header the entire list + of options are returned in the Unsupported header even if some of + those options are in fact supported. This patch fixes that by + building an unsupported options character buffer when parsing the + options that can be sent with the 420 response. A unit test + verifying this functionality has been created. Some code + refactoring was required. Review: + https://reviewboard.asterisk.org/r/680/ + +2010-06-28 17:33 +0000 [r272805] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 272804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun + 2010) | 5 lines Decode URI in contact header of 302 response. + ABE-2352 ........ + +2010-06-28 15:33 +0000 [r272684] Russell Bryant + + * doc/tex/chan-mobile.tex (added), doc/tex/celdriver.tex, + doc/tex/chan_mobile.tex (removed), doc/tex/cdrdriver.tex, + doc/tex/asterisk.tex, doc/tex/cel-doc.tex: Use the underscore + package so that underscores do not need to be escaped. + +2010-06-28 14:55 +0000 [r272652] David Vossel + + * channels/chan_sip.c: code guidelines cleanup for retrans_pkt() + function I am doing work in this function. I noticed a large + number of coding guidline fixes that needed to be made. Rather + than have those changes distract from my functional changes I + decided to separate these into a separate patch. + +2010-06-25 20:18 +0000 [r272568] Tilghman Lesher + + * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272562 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) + | 5 lines Make the structure of the table specified before match + the queries and results. (closes issue #17557) Reported by: cmaj + ........ + +2010-06-25 19:42 +0000 [r272558] Matthew Nicholson + + * res/res_fax.c, include/asterisk/res_fax.h: Implemement support + for handling multiple documents when sending. + +2010-06-25 19:39 +0000 [r272557] David Vossel + + * channels/chan_sip.c: chan_sip: more accurate retransmissions + RFC3261 states that Timer A should start at 500ms (T1) by + default. In chan_sip this value initially started at 1000ms and I + changed it to 500ms recently. After doing that I noticed in my + packet captures that it still occasionally retransmitted starting + at 1000ms instead of 500ms like I told it to. This occurs because + the scheduler runs in the do_monitor thread. If a new + retransmission is added while the do_monitor thread is sleeping + then it may not detect that retransmission for nearly 1000ms. To + fix this I just poke the do_monitor thread to wake up when a new + packet is sent reliably requiring retransmits. The thread then + detects the new scheduler entry and adjusts its sleep time to + account for it. Review: https://reviewboard.asterisk.org/r/747 + +2010-06-25 19:17 +0000 [r272533] Tilghman Lesher + + * sounds/Makefile: Symlink sounds files, to save disk space, when + multiple tarballs/checkouts are on the same system. + +2010-06-24 22:11 +0000 [r272447] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 272446 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) + | 10 lines ss_thread calls pri_grab without lock during overlap + dial Recent changes to chan_dahdi with relation to overlap + dialing call pri_grab without first obtaining a lock. (closes + issue #17414) Reported by: pdf Patches: bug17414.patch uploaded + by jpeeler (license 325) ........ + +2010-06-23 23:09 +0000 [r272370] Russell Bryant + + * channels/chan_iax2.c: Resolve some errors produced during module + unload of chan_iax2. The external test suite stops Asterisk using + the "core stop gracefully" command. The logs from the tests show + that there are a number of problems with Asterisk trying to + cleanly shut down. This patch addresses the following type of + error that comes from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: + lock.c:129 __ast_pthread_mutex_destroy: chan_iax2.c line 11371 + (iax2_process_thread_cleanup): Error destroying mutex + &thread->lock: Device or resource busy For an example in the + context of a build, see: + http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary + purpose of this patch is to change the thread pool shutdown + procedure to be more explicit to ensure that the thread exits + from a point where it is not holding a lock. While testing that, + I encountered various crashes due to the order of operations in + unload_module() being problematic. I reordered some things there, + as well. Review: https://reviewboard.asterisk.org/r/736/ + +2010-06-23 22:36 +0000 [r272368] Matthew Nicholson + + * /, apps/app_queue.c: Merged revisions 272367 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 This version + of the patch only adds AgentComplete for attended transfers. It + was already present for blind transfers. ........ r272367 | + mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 + lines Send AgentComplete manager events in the event of blind and + attended transfers. (closes issue #16819) Reported by: elbriga + Patches: app_queue.diff uploaded by elbriga (license 482) + ........ + +2010-06-23 21:53 +0000 [r272260-272332] Tilghman Lesher + + * res/res_musiconhold.c: If there is realtime configuration, it + does not get re-read on reload unless the config file also + changes. (closes issue #16982) Reported by: dmitri Patches: + res_musiconhold.patch uploaded by dmitri (license 1001) Tested + by: atis + + * res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael_lex.c, + res/ael/ael.flex: Ensure a NULL file while debugging cannot crash + AEL. (closes issue #17215) Reported by: vazir Patches: + 20100518__issue17215.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman + +2010-06-23 21:06 +0000 [r272257-272259] Paul Belanger + + * apps/app_meetme.c: Fix previous merge. ast_test_flag != + ast_test_flag64 + + * /, apps/app_meetme.c: Merged revisions 272255 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun + 2010) | 12 lines First caller into a dynamic conference now enter + pin once. If MeetMe is configured to use dynamic conference + numbers, then the first caller (which creates the conference) had + to enter the PIN number twice. (closes issue #15878) Reported by: + shawkris Patches: issue15878.patch uploaded by pabelanger + (license 224) Tested by: pabelanger ........ + +2010-06-23 20:59 +0000 [r272254-272256] Terry Wilson + + * configure, include/asterisk/autoconfig.h.in: Update configure + when changing autconf m4 files... + + * autoconf/ast_ext_tool_check.m4: Honor the --with-${library}=path + for AST_EXT_TOOL_CHECK (closes issue #16991) Reported by: + pprindeville Patches: with_netsnmp.patch.txt uploaded by twilson + (license 396) Tested by: twilson Review: + https://reviewboard.asterisk.org/r/739/ + +2010-06-23 20:35 +0000 [r272243-272252] Paul Belanger + + * main/manager.c: Correct manager variable 'EventList' case. + (closes issue #17520) Reported by: kobaz Patches: manager.patch + uploaded by kobaz (license 834) Tested by: lmadsen + + * configs/say.conf.sample: Add localization support for Spanish + (closes issue #17548) Reported by: cjacobsen Patches: + say.conf.sample.diff uploaded by cjacobsen (license 1029) + +2010-06-23 19:59 +0000 [r272218] Tim Ringenbach + + * channels/chan_local.c: Add new AMI command LocalOptimizeAway. + This command lets you request a "/n" local channel optimize + itself out of the way anyway. Review: + https://reviewboard.asterisk.org/r/732/ + +2010-06-23 18:45 +0000 [r272148-272150] Tilghman Lesher + + * channels/chan_mgcp.c: D'oh! Defaultenabled FTL. + + * /: Recorded merge of revisions 272147 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r272147 | tilghman | 2010-06-23 13:40:28 -0500 (Wed, 23 Jun 2010) + | 5 lines Backport part of revision 136715 to fix callerid in + voicemail text files (IMAP only). (closes issue #16945) Reported + by: mneuhauser ........ + +2010-06-23 18:39 +0000 [r272146] Terry Wilson + + * apps/app_meetme.c: Don't start the sla thread unless we realy + need it + +2010-06-23 18:25 +0000 [r272145] Tilghman Lesher + + * channels/chan_mgcp.c: Load all lines from realtime, not just the + first one. (closes issue #17144) Reported by: nahuelgreco + Patches: 20100513__issue17144__trunk.diff.txt uploaded by + tilghman (license 14) Tested by: tilghman + +2010-06-23 17:21 +0000 [r272109] Terry Wilson + + * apps/app_meetme.c: Make sure reload updates SLA config Even if + there are no stations or trunks defined, we need to start the sla + thread to make sure we get the reload event. Also, when doing a + reload we need to remove the existing trunks and stations or they + end up hanging around. (closes issue #16818) Reported by: mbonin + Patches: sla_reload.patch uploaded by twilson (license 396) + Tested by: twilson + +2010-06-23 17:08 +0000 [r272090] Mark Michelson + + * channels/chan_sip.c: Add extra protection for reinvite glare + scenario. Testing proved that if Asterisk sent a connected line + reinvite, and the endpoint to which the reinvite were being sent + sent a reinvite, Asterisk would not properly respond with a 491 + response. The reason is that on connected line reinvites, we set + the dialog's invitestate to INV_CALLING to prevent Asterisk from + sending a rapid flurry of connected line reinvites. For other + reinvites we do not do this. Because of the current invitestate, + when Asterisk received the reinvite, we interpreted this as a + spiraled INVITE, and thus did not behave properly. The fix for + this is to not enter the loop detection or spiral logic in + handle_request_invite if the channel state is currently up. This + way, no mid-call reinvites will be misinterpreted, no matter what + the nature of the reinvite may have been. + +2010-06-22 23:20 +0000 [r272052] Russell Bryant + + * channels/chan_dahdi.c: Don't try to lock/unlock an uninitialized + lock on a dahdi_pri. This small changes prevents + destroy_all_channels() from accessing a lock on an unused + dahdi_pri struct, resolving a ton of ERRORs that get spewed out + when shutting Asterisk down gracefully. + +2010-06-22 22:11 +0000 [r271905-272014] David Vossel + + * pbx/pbx_config.c: fixes issue with 'dialplan remove extension + blah' segfaulting with tab completion (closes issue #17440) + Reported by: kobaz + + * channels/chan_sip.c: ignore CANCEL request after having already + received final response to INVITE RFC 3261 section 9 states that + a CANCEL has no effect on a request to a UAS that has already + given a final response. This patch checks to make sure there is a + pending invite before allowing a CANCEL request to be processed, + otherwise it responds to the CANCEL with a "481 Call/Transaction + Does Not Exist". Review: https://reviewboard.asterisk.org/r/697/ + + * main/manager.c: minor fixes for white/black event filters This + fixes a ref count leak in event filters and checks for a filter + container allocation failure during session creation. + +2010-06-22 17:35 +0000 [r271903] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 271902 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun + 2010) | 8 lines Decrease the module ref count in sip_hangup when + SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the + ref count correct. (closes issue #16815) Reported by: rain + Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) + (modified) Tested by: rain ........ + +2010-06-22 16:29 +0000 [r271868] Jeff Peeler + + * main/manager.c, configs/manager.conf.sample, CHANGES: Add regular + expression filtering for manager events. This patch as documented + in the sample config allows one to optionally apply white, black, + or both types of filtering to manager events. The new + 'eventfilter' option is set per user. (closes issue #14861) + Reported by: fnordian Patches: eventfilter3.patch uploaded by + fnordian (license 110), modified by me Review: + https://reviewboard.asterisk.org/r/673/ + +2010-06-22 16:28 +0000 [r271833-271867] Russell Bryant + + * res/ais/clm.c, res/ais/evt.c: Resolve some errors that occur on a + graceful shutdown. Don't Finalize() if Initialize() did not + succeed. This resulted in an error about trying to Finalize() an + invalid handle. Also trim some trailing whitespace while in the + area. + + * res/res_fax.c: Change the method of retrieving the Asterisk + version string. Using this method makes it so res_fax doesn't + have to be rebuilt on every svn update. + +2010-06-22 15:46 +0000 [r271831] David Vossel + + * main/features.c: fixes attended transfer behavior when both + transferee and transferer hung up If both the transferer and + transferee of a attended transfer hangup before the new channel + picks up, the new channel should be hung up as well as it has no + endpoint to talk to. This mirrors the expected behavior used in + 1.4. (closes issue #17444) Reported by: corruptor + +2010-06-22 15:08 +0000 [r271690-271764] Matthew Nicholson + + * CHANGES: Updated the CHANGES file documenting the addition of a + configurable port in the dundi config file. + + * configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions + 271761 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun + 2010) | 9 lines Allow users to specify a port for dundi peers. + (closes issue #17056) Reported by: klaus3000 Patches: + dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65) + Tested by: klaus3000 ........ + + * /, channels/chan_sip.c, include/asterisk/strings.h, + channels/sip/include/sip.h: Merged revisions 271689 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, + 22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to + automatically calculate the Content-Length. This is done by + storing packet content in a buffer until it is actually time to + send the packet, at which time the size of the packet is + calculated. This change was made to ensure that the + Content-Length is always correct. (closes issue #17326) Reported + by: kenner Tested by: mnicholson, kenner Review: + https://reviewboard.asterisk.org/r/693/ ........ This change also + adds an ast_str_copy_string() function (similar to + ast_copy_string), that copies one ast_str into another, properly + handling embedded nulls. + +2010-06-21 22:41 +0000 [r271657] Tilghman Lesher + + * build_tools/menuselect-deps.in, configure, configure.ac, + res/res_timing_kqueue.c: Conflict kqueue on OS X, since it + doesn't work there yet, anyway. + +2010-06-21 21:58 +0000 [r271625] David Vossel + + * codecs/codec_speex.c, codecs/ex_speex.h, + contrib/editors/asterisk.vim: add speex 16khz sample frame so + codec cost can be calculated (closes issue #17534) Reported by: + fabled Patches: speex-wb-sample.diff uploaded by fabled (license + 448) + +2010-06-21 20:46 +0000 [r271554] Jeff Peeler + + * res/ael/pval.c, /: Merged revisions 271552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) + | 7 lines Do not use sizeof to calculate size of a heap allocated + character array. Change left out from 271399. (closes issue + #16053) Reported by: diLLec ........ + +2010-06-21 20:46 +0000 [r271551-271553] David Vossel + + * channels/chan_sip.c, channels/sip/reqresp_parser.c: fixes crash + when From header URI is missing "sip:" (closes issue #17437) + Reported by: klaus3000 Patches: sip_crash uploaded by dvossel + (license 671) Tested by: klaus3000 + + * res/res_rtp_asterisk.c: fixes logic error introduced by slin16 + sip support + +2010-06-21 05:10 +0000 [r271520] Tilghman Lesher + + * apps/app_saycounted.c (added), CHANGES: Add new application for + declining counting words in multiple languages. (closes issue + #16869) Reported by: chappell Patches: app_say_counted-20100317.c + uploaded by chappell (license 8) Tested by: chappell + +2010-06-18 21:32 +0000 [r271483] Jeff Peeler + + * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged + revisions 271399 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) + | 11 lines Fix crash when parsing some heavily nested statements + in AEL on reload. Due to the recursion used when compiling AEL in + gen_prios, all the stack space was being consumed when parsing + some AEL that contained nesting 13 levels deep. Changing a few + large buffers to be heap allocated fixed the crash, although I + did not test how many more levels can now be safely used. (closes + issue #16053) Reported by: diLLec Tested by: jpeeler ........ + +2010-06-18 18:59 +0000 [r271341] David Vossel + + * main/file.c: file.c was truncating audio file formats to the + lower 32bits. + +2010-06-18 18:36 +0000 [r271336] Jeff Peeler + + * /: Recorded merge of revisions 271335 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010) + | 13 lines Eliminate deadlock potential in dahdi_fixup(). (This + is a backport of 269307, committed to trunk by rmudgett.) Calling + dahdi_indicate() when the channel private lock is already held + can cause a deadlock if the PRI lock is needed because + dahdi_indicate() will also get the channel private lock. The + pri_grab() function assumes that the channel private lock is held + once to avoid deadlock. (closes issue #17261) Reported by: aragon + ........ + +2010-06-17 21:23 +0000 [r271231-271300] David Vossel + + * channels/sip/reqresp_parser.c: fixes some coding guideline issue + + * channels/sip/include/dialog.h, channels/chan_sip.c, + channels/sip/include/sip.h: retransmit response to BYE requests + until timer J expires According to RFC 3261 section 17.2.2, which + describes non-INVITE server transaction, when a dialog enters the + Completed state it must destroy the dialog after Timer J (T1*64) + fires. For a BYE transaction Asterisk terminates the dialog + immediately during sip_hangup() when it should be waiting T1*64 + ms. This results in some odd behavior. For instance if Asterisk + receives a BYE and transmits a 200ok in response, if the endpoint + never receives the 200ok it will retransmit the BYE to which + Asterisk responds with a "481 Call leg/transaction does not + exist" because the dialog is already gone. To resolve this I made + a function called sip_scheddestroy_final(). This differs slightly + from sip_schedestroy() in that it enables a flag that will + prevent the destruction from ever being rescheduled or canceled + afterwards. It also prevents the pvt's needdestroy flag from + being set which triggers the destruction of the dialog within the + do_monitor thread(). By using this function we are guaranteed + destruction will not occur until the scheduled time. This allows + Asterisk to respond to any possible retransmits for a dialog + after we process the initial BYE request for T1*64 ms. Other + changes: I removed two instances where sip_cancel_destroy is used + right before calling sip_scheddestroy. sip_scheddestroy always + calls sip_cancel_destroy before scheduling the new destruction so + it is completely unnecessary. Review: + https://reviewboard.asterisk.org/r/694/ + + * res/res_rtp_asterisk.c, main/rtp_engine.c, CHANGES: adds support + for slin16 in sip (closes issue #16153) Reported by: kfister + Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license + 912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested + by: kfister, malcolmd + + * main/channel.c, res/res_rtp_asterisk.c, main/frame.c, + main/rtp_engine.c, codecs/codec_speex.c, CHANGES, + include/asterisk/frame.h: adds speex 16khz audio support (closes + issue #17501) Reported by: fabled Patches: + asterisk-trunk-speex-wideband-v2.patch uploaded by fabled + (license 448) Tested by: malcolmd, fabled, dvossel + +2010-06-17 15:34 +0000 [r271192] Jeff Peeler + + * channels/sig_analog.c: Change expected operation from error to + debug message + +2010-06-17 00:30 +0000 [r271089] Paul Belanger + + * apps/app_meetme.c: option w[(secs)] incorrectly capitalized in + xmldoc (closes issue #17516) Reported by: karlfife + +2010-06-16 22:37 +0000 [r271056] David Vossel + + * channels/sip/reqresp_parser.c: addition of more parse_uri test + cases + +2010-06-16 21:17 +0000 [r270987] Paul Belanger + + * /, configs/extensions.conf.sample: Merged revisions 270979 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun + 2010) | 4 lines Fixed typo in macro-page Reported to + #asterisk-dev by a student of jsmith. ........ + +2010-06-16 21:12 +0000 [r270981-270983] Jason Parker + + * channels/chan_agent.c: Fix the actual place that was pointed out, + for previous commit. + + * /, channels/chan_agent.c: Merged revisions 270980 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun + 2010) | 4 lines Need to lock the agent chan before access its + internal bits. Pointed out by russellb on asterisk-dev mailing + list. ........ + +2010-06-16 20:34 +0000 [r270974] Matthew Nicholson + + * main/dnsmgr.c, main/acl.c: Set sin_family to AF_INET when doing + lookups, also reset sin_port the first time the ip address + changes. (closes issue #17496) Reported by: ManChicken (closes + issue #15827) Reported by: DennisD Patches: dnsmgr_15827.patch + uploaded by chappell (license 8) Tested by: DennisD, gentlec, + damage, wimpy + +2010-06-16 19:03 +0000 [r270940] David Vossel + + * main/channel.c, res/res_rtp_asterisk.c, main/frame.c, + main/rtp_engine.c, channels/chan_sip.c, CHANGES, + channels/chan_iax2.c, include/asterisk/frame.h, + formats/format_g719.c (added): addition of G.719 pass-through + support (closes issue #16293) Reported by: malcolmd Patches: + g719.passthrough.patch.7 uploaded by malcolmd (license 924) + format_g719.c uploaded by malcolmd (license 924) + +2010-06-16 18:43 +0000 [r270936] Paul Belanger + + * res/res_agi.c, CHANGES: MSG_OOB flag on HANGUP packet removed. + Per Tilghman's request on IRC (#asterisk-bugs). (closes issue + #17506) Reported by: brycebaril Tested by: pabelanger, tilghman + +2010-06-16 17:36 +0000 [r270867] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 270866 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 + Jun 2010) | 22 lines fixes chan_iax2 race condition There is code + in chan_iax2.c that attempts to guarantee that only a single + active thread will handle a call number at a time. This code + works once the thread is added to an active_list of threads, but + we are not currently guaranteed that a newly activated thread + will enter the active_list immediately because it is left up to + the thread to add itself after frames have been queued to it. + This means that if two frames come in for the same call number at + the same time, it is possible for them to grab two separate + threads because the first thread did not add itself to the + active_list fast enough. This causes some pretty complex + problems. This patch resolves this race condition by immediately + adding an activated thread to the active_list within the network + thread and only depending on the thread to remove itself once it + is done processing the frames queued to it. By doing this we are + guaranteed that if another frame for the same call number comes + in at the same time, that this thread will immediately be found + in the active_list of threads. Review: + https://reviewboard.asterisk.org/r/720/ ........ + +2010-06-16 16:45 +0000 [r270836] Jeff Peeler + + * channels/sig_analog.c: Fix no call waiting caller ID Clearing the + callwaitcas flag in analog_call was causing the incoming D digit + to be ignored which triggers sending the caller ID. + +2010-06-16 15:05 +0000 [r270801] Paul Belanger + + * doc/tex/channelvariables.tex: Update formatting for + channelvariables.tex (closes issue #17511) Reported by: klaus3000 + Patches: channelvariables.tex-patch.txt uploaded by klaus3000 + (license 65) Tested by: pabelanger + +2010-06-15 22:48 +0000 [r270726] Russell Bryant + + * channels/sig_analog.c: Don't blow up if an ast_channel doesn't + get allocated. + +2010-06-15 21:42 +0000 [r270658-270692] Terry Wilson + + * main/http.c: Don't continue sending the file when there has been + an error If there is a problem with a firmware file, Polycom + phones will close the connection. We were continuing to send the + file anyway. There should be no reason to continue sending a file + if there is an error writing it. (closes issue #16682) Reported + by: lmadsen + + * res/res_phoneprov.c: Don't send files twice and remove extra \r\n + from header After the manager http auth changes, we forgot to + remove the manual sending of the file. Also, ast_http_send adds + two \r\n to the header that is passed to it, so a trailing \r\n + is removed from the Content-type header. It might be better to + change ast_http_send, but I don't like changing the behavior of + an API function. (closes issue #17239) Reported by: cjacobsen + Patches: patch2.diff uploaded by cjacobsen (license 1029) Tested + by: lathama, cjacobsen + + * channels/chan_sip.c: Make contactdeny apply to src ip when + nat=yes chan_sip's "contactdeny" feature screens the "to be + registered contact". In case of nat=yes it should not use the + address information from the Contact header (which is not used at + all for routing), but the source IP address of the request. Thus, + if nat=yes and a client sends a request from a denied IP address + (e.g. by spoofing the src-IP address) it can bypass the + screening. This commit makes contactdeny apply to the src ip when + nat=yes instead. (closes issue #17276) Reported by: klaus3000 + Patches: patch-asterisk-trunk-contactdeny.txt uploaded by + klaus3000 (license 65) Tested by: klaus3000 + +2010-06-15 18:26 +0000 [r270519-270584] Tilghman Lesher + + * main/pbx.c, /: Merged revisions 270583 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010) + | 5 lines Variables have always been case-sensitive, so we should + not be removing case-insensitive matches. Bug reported via the + -dev list. See + http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html + ........ + + * res/res_jabber.c: Argh, mixed declarations and code. + + * configs/jabber.conf.sample, include/asterisk/jabber.h, + doc/distributed_devstate-XMPP.txt (added), CHANGES, + res/res_jabber.c: Add distributed devicestate via the XMPP + protocol. (closes issue #15757) Reported by: Marquis Patches: + distributed_devstate-XMPP.txt uploaded by lmadsen (license 10) + Tested by: Marquis, lmadsen, marcelloceschia Review: + https://reviewboard.asterisk.org/r/351/ + +2010-06-15 12:51 +0000 [r270443] Leif Madsen + + * /, configs/voicemail.conf.sample: Merged revisions 270442 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010) + | 1 line Move information about zonemessages into the + [zonemessages] section. ........ + +2010-06-14 21:33 +0000 [r270332] Paul Belanger + + * /, res/res_musiconhold.c: Merged revisions 270331 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon, + 14 Jun 2010) | 14 lines Properly play first file in sort list. + When using sort=alpha we would always skip the first file in the + list first time through. We now check for that properly. (closes + issue #17470) Reported by: pabelanger Patches: sort.aplha.patch + uploaded by pabelanger (license 224) Tested by: lmadsen Review: + https://reviewboard.asterisk.org/r/703/ ........ + +2010-06-14 20:51 +0000 [r270298] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c: + Extract sig_ss7_init_linkset() to sig_ss7. Also found a place + where sig_pri_init_pri() was inlined and called it instead. + +2010-06-14 19:41 +0000 [r270260] Jason Parker + + * channels/chan_agent.c: Add option to get untruncated channel name + from AGENT function. The "channel" option would chop the channel + name at the last '-', which made it useless for something like a + channel transfer from the dialplan. The "fullchannel" option will + return the channel name as-is. ABE-2218 + +2010-06-14 15:55 +0000 [r270219] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, channels/sig_pri.c: Add digit + manipulation tag support to chan_dahdi/sig_pri like chan_misdn. + Add the append_msn_to_cid_tag option to chan_dahdi like + chan_misdn. Review: https://reviewboard.asterisk.org/r/696/ + +2010-06-13 09:16 +0000 [r270184] Tzafrir Cohen + + * autoconf/ast_check_pwlib.m4, configure: bashism in configure + script Theoretically the ./configure script is a pure + bourne-shell script. Practically it may be run by bash if /bin/sh + is not good enough. But we should not count on it. See bug report + for the gory details. (closes issue #17485) Patches: + 0001-remove-bashism-from-ast_check_pwlib.m4.patch uploaded by + tzafrir (license 46) + +2010-06-13 01:53 +0000 [r270042-270151] Paul Belanger + + * configure, include/asterisk/autoconfig.h.in, configure.ac: + Reverting patch and reopening issue #16155, as patch breaks + FreeBSD / OSX builds. + + * /, doc/HOWTO_collect_debug_information.txt: Merged revisions + 270078 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun + 2010) | 2 lines Fix typo in example ........ + + * configure, include/asterisk/autoconfig.h.in, configure.ac: Use + pkg-config to find gmime libraries This way the libraries can be + found even if they are in non-standard locations. (closes issue + #16155) Reported by: jcollie Patches: + 0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch + uploaded by jcollie (license 412) Tested by: jsmith, tilghman, + pabelanger + +2010-06-11 18:31 +0000 [r269936-269976] Tilghman Lesher + + * main/frame.c, /: Merged revisions 269960 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010) + | 8 lines For SpeeX, 0 bits remaining is valid and does not need + an emitted warning. (closes issue #15762) Reported by: nblasgen + Patches: issue15672.patch uploaded by pabelanger (license 224) + Tested by: nblasgen ........ + + * CHANGES, main/db.c: Add DBGetComplete event after a + DBGetResponse. (closes issue #16965) Reported by: rrb3942 + Patches: DBGetComplete.patch uploaded by rrb3942 (license 1003) + + * main/logger.c: Remove lines from the output related to the + backtrace itself. + +2010-06-10 20:30 +0000 [r269889] Paul Belanger + + * Makefile, makeopts.in: Remove ASTBINDIR variable (closes issue + #17031) Reported by: pabelanger Patches: + Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224) + Tested by: pabelanger, tilghman + +2010-06-10 19:34 +0000 [r269749-269822] Mark Michelson + + * main/channel.c, /: Merged revisions 269821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun + 2010) | 19 lines Fix potential crash when writing raw SLIN audio + on a PLC-enabled channel. The issue here was that the frame + created when adjusting for PLC had no offset to its audio data. + If this frame were translated to another format prior to being + sent out an RTP socket, all went well because the translation + code would put an appropriate offset into the frame. However, if + the SLIN audio were not translated before being sent out the RTP + socket, bad things would happen. Specifically, the + ast_rtp_raw_write makes the assumption that the frame has at + least enough of an offset that it can accommodate an RTP header. + This was not the case. As such, data was being written prior to + the allocation, likely corrupting the data the memory allocator + had written. Thus when the time came to free the data, all hell + broke loose. ....Well, Asterisk crashed at least. The fix was + just what one would expect. Offset the data in the frame by a + reasonable amount. The method I used is a bit odd since the data + in the frame is 16 bit integers and not bytes. I left a big ol' + comment about it. This can be improved on if someone is + interested. I was more interested in getting the crash resolved. + ........ + + * doc/tex/plc.tex (added), doc/tex/asterisk.tex: Add documentation + explaining PLC in Asterisk. Review: + https://reviewboard.asterisk.org/r/688/ + +2010-06-10 13:17 +0000 [r269711] Russell Bryant + + * tests/test_heap.c: Fix an off by one error that caused a unit + test to occasionally crash. + +2010-06-10 12:28 +0000 [r269707] Kevin P. Fleming + + * main/logger.c: Ensure that 'logger show channels' works properly + when wildcards are used in logger.conf. + +2010-06-10 08:15 +0000 [r269636] Tilghman Lesher + + * /, main/logger.c, utils/extconf.c, main/asterisk.c: Merged + revisions 269635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010) + | 9 lines Ensure restartable system calls can restart (BSD signal + semantics). This eliminates the annoying on the console. + (closes issue #17477) Reported by: jvandal Patches: + 20100610__issue17477.diff.txt uploaded by tilghman (license 14) + ........ + +2010-06-10 00:32 +0000 [r269417-269602] Russell Bryant + + * channels/chan_dahdi.c: Attempt to fix a FreeBSD build error by + including sys/stat.h. + http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log + + * main/lock.c: Attempt to fix FreeBSD build problem. + + * /, channels/chan_oss.c: Merged revisions 269495 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010) + | 2 lines Don't stop Asterisk if chan_oss fails to register + 'Console' (due to another channel driver already claiming it). + ........ + + * include/asterisk/event.h, main/event.c: Resolve an invalid memory + read on an event. Valgrind pointed out that attempting to get an + IE value from an event that has no IEs produces an invalid memory + read past the end of the event. Thanks to mmichelson for pointing + the problem out to me and then testing the fix. + +2010-06-09 17:32 +0000 [r269346] Paul Belanger + + * contrib/init.d/rc.debian.asterisk, /, main/term.c: Merged + revisions 269334 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun + 2010) | 12 lines Fix Debian init script to not use -c. When using + the init script as-is currently, it could cause issues on Debian + such as high CPU usage. This fix has worked for several people so + I'm implementing the change. We now handle color displays + properly. (closes issue #16784) Reported by: pabelanger Patches: + 20100530__issue16784__2.diff.txt uploaded by tilghman (license + 14) Tested by: pabelanger, tilghman ........ + +2010-06-09 17:06 +0000 [r269307-269308] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c: + Add missing API function to sig_ss7: sig_ss7_fixup(). + + * channels/chan_dahdi.c: Eliminate deadlock potential in + dahdi_fixup(). Calling dahdi_indicate() within dahdi_fixup() + while the owner pointers are in a potentially inconsistent state + is a potentially bad thing in principle. However, calling + dahdi_indicate() when the channel private lock is already held + can cause a deadlock if the PRI lock is needed because + dahdi_indicate() will also get the channel private lock. The + pri_grab() function assumes that the channel private lock is held + once to avoid deadlock. + +2010-06-09 15:09 +0000 [r269271] David Vossel + + * res/res_musiconhold.c: fixes crash in moh when cachertclasses + flag is used The result for moh_register was not verified to + guarantee the mohclass as added to the container. (closes issue + #16993) Reported by: dmitri Patches: + res_musiconhold_rtclass2.patch uploaded by dmitri (license 1001) + moh_crash2.diff uploaded by dvossel (license 671) Tested by: + dmitri + +2010-06-09 13:17 +0000 [r269238] Tzafrir Cohen + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES: + dial by name in chan_dahdi * chan_dahdi supports dialing + configuring and dialing by device file name. + DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . + Likewise it may appear in chan_dahdi.conf as 'channel => + span-name!local!1'. * A new options for chan_dahdi.conf: + 'ignore_failed_channels'. Boolean. False by default. If set, + chan_dahdi will ignore failed 'channel' entries. Handy for the + above name-based syntax as it does not depend on initialization + order. * have my_pri_make_cc_dialstring() only manupulate + dial-strings of group (gGrR) dialing, which make it lsightly more + complicated. https://reviewboard.asterisk.org/r/535/ + +2010-06-09 10:55 +0000 [r269187-269205] Russell Bryant + + * contrib/scripts/install_prereq: Add libjack-dev to + install_prereq. + + * contrib/scripts/install_prereq: Add libpopt-dev, libical-dev, and + libspandsp-dev to install_prereq. + + * contrib/scripts/install_prereq: Add libnewt-dev to + install-prereq. + + * contrib/scripts/install_prereq: Add libopenais-dev to + install_prereq. + + * contrib/scripts/install_prereq: Add an "install-unpackaged" + command to install_prereq for installing unpackaged dependencies + (such as NBS and libresample). + + * contrib/scripts/install_prereq: Add libcurl to install_prereq. + + * contrib/scripts/install_prereq: Add freetds-dev to + install_prereq. + + * contrib/scripts/install_prereq: Add libradiusclient-ng-dev to + install_prereq. + + * contrib/scripts/install_prereq: Add libbluetooth-dev to + install_prereq. + + * contrib/scripts/install_prereq: Add libmysqlclient-dev to + install_prereq. + + * contrib/scripts/install_prereq: Add libgtk2.0-dev to the packages + list for install_prereq. + +2010-06-08 23:48 +0000 [r269153] Bradley Latus + + * configs/cdr_custom.conf.sample, configs/cdr_tds.conf.sample, + cdr/cdr_sqlite.c, configs/cdr_sqlite3_custom.conf.sample, + funcs/func_cdr.c, configs/cdr_syslog.conf.sample, UPGRADE.txt, + cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, cdr/cdr_pgsql.c, + CHANGES, cdr/cdr_odbc.c, cdr/cdr_tds.c, + configs/cdr_odbc.conf.sample: Add High Resolution Times to CDRs + for Asterisk People expressed an interest in having access to the + exact length of calls to a finer degree than seconds. See the + CHANGES and UPGRADE.txt for usage also updated the sample configs + to note the change. Patch by snuffy. (closes issue #16559) + Reported by: cianmaher Tested by: cianmaher, snuffy Review: + https://reviewboard.asterisk.org/r/461/ + +2010-06-08 22:45 +0000 [r269119] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/localtime.h: Fix build on Mac OS X (and maybe + FreeBSD, too) + +2010-06-08 18:50 +0000 [r269083] Matthew Nicholson + + * apps/app_fax.c: Don't pass null to manager_event() (closes issue + #17087) Reported by: bklang Patches: app-fax-null-sprintf1.diff + uploaded by mnicholson (license 96) Tested by: bklang + +2010-06-08 15:41 +0000 [r269008] Russell Bryant + + * Makefile.rules: Ensure CONFIG_FLAGS makes it into the build rules + when doing out of tree builds. (closes issue #16685) Reported by: + pprindeville + +2010-06-08 15:39 +0000 [r269007] Sean Bright + + * /, cdr/cdr_tds.c: Merged revisions 269006 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r269006 | seanbright | 2010-06-08 11:28:49 -0400 (Tue, 08 Jun + 2010) | 11 lines Reduce startup time for cdr_tds with large CDR + tables. Since we are just checking for table existence, add a + WHERE clause that will return no rows but will raise an error if + the table doesn't exist. (closes issue #17380) Reported by: + kkwong Patches: issue17380-01.patch uploaded by seanbright + (license 71) Tested by: kkwong ........ + +2010-06-08 15:23 +0000 [r268969-268988] Leif Madsen + + * configs/sip.conf.sample: Update note in sip.conf.sample. Update + note in sip.conf.sample about externip and externhost with STUN. + (closes issue #16323) Reported by: klaus3000 Patches: + sip.conf.sample-patch.txt uploaded by klaus3000 (license 65) + + * apps/app_meetme.c, main/ccss.c, include/asterisk/data.h, + res/res_jabber.c, res/res_config_sqlite.c, + include/asterisk/callerid.h, channels/chan_dahdi.c, + include/asterisk/bridging_technology.h, + include/asterisk/doxyref.h, include/asterisk/event.h, + include/asterisk/astmm.h, main/ast_expr2f.c, main/features.c, + include/asterisk/timing.h, include/asterisk/rtp_engine.h, + include/asterisk/ccss.h, include/asterisk/threadstorage.h, + include/asterisk/xml.h, main/pbx.c, channels/chan_sip.c, + include/asterisk/astobj2.h, include/asterisk/channel.h, + include/asterisk/calendar.h, include/asterisk/manager.h, + include/asterisk/features.h, include/asterisk/logger.h, + include/asterisk/http.h, channels/sig_pri.h, + include/asterisk/app.h, main/audiohook.c, include/asterisk/pbx.h, + include/asterisk/dnsmgr.h, include/asterisk/smdi.h, + apps/app_voicemail.c: Fix some doxygen warnings. (closes issue + #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded + by snuffy (license 35) Tested by: russell + +2010-06-08 06:57 +0000 [r268896-268933] Tilghman Lesher + + * res/res_config_sqlite.c: Release list lock before returning on + error. + + * utils/extconf.c: Fix trunk build on Mac OS X. + +2010-06-08 05:29 +0000 [r268894] Terry Wilson + + * channels/sip/sdp_crypto.c (added), res/res_rtp_asterisk.c, + main/global_datastores.c, main/rtp_engine.c, + include/asterisk/res_srtp.h (added), channels/sip/srtp.c (added), + channels/chan_sip.c, include/asterisk/autoconfig.h.in, + res/res_srtp.exports.in (added), configure.ac, CHANGES, + channels/chan_iax2.c, res/res_srtp.c (added), main/channel.c, + build_tools/menuselect-deps.in, main/asterisk.exports.in, + configure, funcs/func_channel.c, + channels/sip/dialplan_functions.c, + channels/sip/include/sdp_crypto.h (added), + doc/tex/secure-calls.tex (added), + include/asterisk/global_datastores.h, channels/sip/include/srtp.h + (added), makeopts.in, include/asterisk/rtp_engine.h, + include/asterisk/frame.h, doc/tex/asterisk.tex, + channels/sip/include/sip.h: Add SRTP support for Asterisk After 5 + years in mantis and over a year on reviewboard, SRTP support is + finally being comitted. This includes generic CHANNEL dialplan + functions that work for getting the status of whether a call has + secure media or signaling as defined by the underlying channel + technology and for setting whether or not a new channel being + bridged to a calling channel should have secure signaling or + media. See doc/tex/secure-calls.tex for examples. Original patch + by mikma, updated for trunk and revised by me. (closes issue + #5413) Reported by: mikma Tested by: twilson, notthematrix, + hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ + +2010-06-08 00:45 +0000 [r268857] Richard Mudgett + + * channels/sip/dialplan_functions.c: Make SIP tests compile again. + +2010-06-07 22:56 +0000 [r268817-268818] Tilghman Lesher + + * channels/chan_sip.c: Use the mailbox destructor function, + instead. + + * channels/chan_sip.c, channels/sip/include/sip.h: Mailbox list + would previously grow at each reload, containing duplicates. + Also, optimize the allocation of mailboxes to avoid additional + memory structures. (closes issue #16320) Reported by: Marquis + Patches: 20100525__issue16320.diff.txt uploaded by tilghman + (license 14) + +2010-06-07 20:04 +0000 [r268774] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_ss7.h + (added), channels/Makefile, channels/sig_pri.c, + channels/sig_ss7.c (added): Extract sig_ss7 out of chan_dahdi. + Extract the SS7 specific code out of chan_dahdi like what was + done to ISDN/PRI and analog signaling. The new SS7 structures + were modeled on sig_pri. The changes to sig_pri are an + enhancement and a bug fix made possible because SS7 was + extracted. 1) The sig_pri TRANSFERCAPABILITY channel variable + should have been set unconditionally in + sig_pri_new_ast_channel(). 2) SS7/PRI transfer capability + interaction in dahdi_new() fixed because of SS7 extraction. 3) + Module ref count error in dahdi_new() if startpbx failed to start + the PBX for some reason. Review: + https://reviewboard.asterisk.org/r/661/ + +2010-06-07 19:52 +0000 [r268773] Tilghman Lesher + + * main/rtp_engine.c, channels/chan_sip.c, + channels/sip/dialplan_functions.c, include/asterisk/rtp_engine.h: + Seems strange (and the code backs up) that if the max and min of + a statistic is expressed as a double, the last value would not + also need to be a double. (closes issue #15807) Reported by: + klaus3000 + +2010-06-07 19:06 +0000 [r268734] Richard Mudgett + + * channels/sig_pri.c: Moved AOC request code out of the middle of + code parsing the dialed number. + +2010-06-07 18:59 +0000 [r268731] Tilghman Lesher + + * main/manager.c: Event well was going dry. (issue #17234) + +2010-06-07 17:34 +0000 [r268690] Paul Belanger + + * main/dsp.c: Set threshold for silence detection defaults to 256 + (closes issue #15685) Reported by: david_s5 Patches: + dsp-silence-threshold-init.diff uploaded by dant (license 670) + issue15685.patch.v5 uploaded by pabelanger (license 224) Tested + by: danti Review: https://reviewboard.asterisk.org/r/670/ + +2010-06-07 17:14 +0000 [r268653] Tilghman Lesher + + * res/res_smdi.c: Avoid unloading res_smdi twice. (closes issue + #17237) Reported by: pabelanger + +2010-06-07 15:51 +0000 [r268578] Richard Mudgett + + * main/file.c: Suppress warning in waitstream_core(). Suppress the + warning about unexpected control subclass frames for + AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and + AST_CONTROL_AOC in file.c:waitstream_core(). + +2010-06-06 05:29 +0000 [r268454-268534] Tilghman Lesher + + * contrib/init.d/rc.redhat.asterisk: Take advantage of variable + substitution already in the Makefile to specify the correct + location for files in init.d. (closes issue #16979) Reported by: + jw-asterisk (issue #15691) Reported by: itamarjp + + * channels/chan_iax2.c: Finally track down and eliminate the + "FRACK! warnings from chan_iax2". + + * main/dsp.c: Fix crash in DTMF detection. What I did not + originally see in my previous commit was that even though the + next digit could be detected before the previous was considered + ended, the detection of the next digit effectively ends the + detection of the previous. Therefore, the length moves in + lockstep with the digit, and no separate counter is needed for + the length alone. (closes issue #17371) Reported by: alecdavis + (closes issue #17474) Reported by: kenner + + * main/manager.c: Verify event is not NULL before attempting to + lower its usecount. (closes issue #17234) Reported by: mav3rick + +2010-06-05 05:23 +0000 [r268395-268417] Kevin P. Fleming + + * CHANGES: Typo fix. + + * CHANGES: Grammatical error fix. + +2010-06-05 02:51 +0000 [r268321] Tilghman Lesher + + * /, configs/voicemail.conf.sample: Merged revisions 268320 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010) + | 3 lines Rest In Peace + http://www.outandaboutnewspaper.com/article/4061 ........ + +2010-06-04 22:37 +0000 [r268205-268281] David Vossel + + * channels/chan_sip.c: fixes compile error from uninitialized + variable + + * channels/chan_sip.c: RFC3261 compliant sip unreliable retransmit + timing + 'registerattempts' option tweak Changes. 1. RFC 3261 + states in section 17.1.2.2 and 17.1.1.2 that retransmission + timers should initially be set to timer T1. T1 by default is + 500ms. Asterisk was starting the retransmission timers at T1*2 + which shouldn't cause any problems, but is not RFC compliant. 2. + RFC 3261 states in section 17.1.2.2 that for a non-INVITE client + transaction, if the retransmit timer fires while in the + proceeding state that the request must be retransmitted. Asterisk + currently ack's requests for both INVITE and non-INVITE + transactions when a 1XX response is received, this patch changes + this for non-INVITE requests. 3. The 'registerattempts' option in + sip.conf is supposed to set how many registry attempts will be + made before giving up. When this option is set to 1, I would + expect only one registry attempt to be made before stopping + because of a failure, but instead two are made. In my opinion + this is not expected behavior. This option does not indicate that + these are re-attempts. The logic behind this option has been + changed to only attempt registers the exact number of times this + option is set to. If this option is 0, it still continues to + re-attempt the registration forever. Review: + https://reviewboard.asterisk.org/r/687/ + +2010-06-04 20:42 +0000 [r267972-268127] Tilghman Lesher + + * /, configure, configure.ac: Merged revisions 268126 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r268126 | tilghman | 2010-06-04 15:41:24 -0500 (Fri, 04 + Jun 2010) | 2 lines AC_CONFIG_SUBDIRS has a bad side-effect on + cross-compiles. ........ + + * Makefile, /, makeopts.in: Merged revisions 268050 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r268050 | tilghman | 2010-06-04 14:38:57 -0500 (Fri, 04 + Jun 2010) | 6 lines Build menuselect with the build environment's + compiler, not the host (target)'s compiler. (closes issue #17464) + Reported by: pprindeville Tested by: tilghman ........ + + * /, configure, configure.ac, autoconf/libcurl.m4: Merged revisions + 267971 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r267971 | tilghman | 2010-06-04 11:27:02 -0500 (Fri, 04 Jun 2010) + | 2 lines As-fixiate the build process ........ + +2010-06-04 14:45 +0000 [r267928] Richard Mudgett + + * channels/sig_pri.c: Incoming overlap dialing no longer works + after sig_pri extraction. The problem would manifest itself if + your dialplan matching could accept more digits to match than + were actually dialed. The time out waiting for overlap digits + disconnected the call instead of matching any accumulated digits + to the dialplan. Accidental conversion of a break out of loop as + a break out of switch. (closes issue #17401) Reported by: + avalentin Patches: issue17401_digit_timeout.patch uploaded by + rmudgett (license 664) Tested by: avalentin, rmudgett + +2010-06-04 03:20 +0000 [r267877] Tilghman Lesher + + * include/asterisk/slin.h: As signed linear audio data is accessed + as 16-bit values, certain processors require the values to be + aligned in memory. (closes issue #16912) Reported by: + michaelevdokimov Patches: asterisk.patch uploaded by + michaelevdokimov (license 997) Tested by: michaelevdokimov + +2010-06-04 03:11 +0000 [r267863] Terry Wilson + + * channels/chan_sip.c: Send an ACK for every final response + received for an INVITE From issue ABE-2247. RFC 3261 compliance + for sections 13.2.24 and 17.1.1.2. Review: + https://reviewboard.asterisk.org/r/692/ + +2010-06-04 02:58 +0000 [r267775-267862] Tilghman Lesher + + * include/asterisk/slin.h: As signed linear audio data is accessed + as 16-bit values, certain processors require the values to be + aligned in memory. (closes issue #16912) Reported by: + michaelevdokimov + + * configure, autoconf/ast_ext_lib.m4: If there's a default, turn it + on, even when the option isn't specified. + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 267759 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r267759 | tilghman | 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010) + | 7 lines Make the default install path appear to be /usr on + Linux, instead of /usr/local. Also, reorganize the options, so + that they're more alphabetical. (closes issue #17013) Reported + by: klaus3000 ........ + +2010-06-03 20:41 +0000 [r267714] Russell Bryant + + * main/ccss.c: Remove a LOG_WARNING. This came up when using the + sample configs, and just indicates expected behavior. + +2010-06-03 19:46 +0000 [r267669] Tilghman Lesher + + * funcs/func_odbc.c: Handle OOM errors more gracefully. (closes + issue #17084) Reported by: falves11 Patches: + issue17084_162_A.diff uploaded by falves11 (license 374) Tested + by: falves11 + +2010-06-03 18:53 +0000 [r267624] Leif Madsen + + * UPGRADE.txt, CHANGES: Update UPGRADE.txt and CHANGE for CDR + functionality changes. Updated the UPGRADE.txt and CHANGES file + stating that CDR records will not be explicity written unless + cdr.conf exists and is configured. (closes issue #17373) Reported + by: wdoekes Tested by: pabelanger + +2010-06-03 18:38 +0000 [r267622] Richard Mudgett + + * codecs/codec_dahdi.c: Make compile again. + +2010-06-03 17:31 +0000 [r267537] Russell Bryant + + * channels/chan_usbradio.c: Don't stop Asterisk if chan_usbradio + isn't configured. + +2010-06-03 17:09 +0000 [r267492] Mark Michelson + + * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c, + codecs/codec_alaw.c, main/translate.c, codecs/codec_g726.c, + codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_dahdi.c, + include/asterisk/translate.h: Remove unnecessary code relating to + PLC. The logic for handling generic PLC is now handled in + ast_write in channel.c instead of in translation code. Review: + https://reviewboard.asterisk.org/r/683/ + +2010-06-03 17:05 +0000 [r267445-267490] Russell Bryant + + * channels/chan_usbradio.c: Remove a line that was killing Asterisk + on startup. + + * channels/h323/Makefile.in: Comment out a rule that likes to run + implicitly unnecessarily, breaking builds + +2010-06-03 00:02 +0000 [r267399] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, configure, + include/asterisk/autoconfig.h.in, configure.ac, CHANGES, + channels/sig_pri.c: Add ETSI Message Waiting Indication (MWI) + support. Add the ability to report waiting messages to ISDN + endpoints (phones). Relevant specification: EN 300 650 and EN 300 + 745 Review: https://reviewboard.asterisk.org/r/599/ + +2010-06-02 22:46 +0000 [r267352] Russell Bryant + + * channels/Makefile, channels/h323/Makefile.in: try to fix some + random chan_h323 compilation failures After some debugging, the + random chan_h323 build failures appear to be due to complications + introduced by some chan_h323 specific build stuff getting + triggered during a clean. Simplify this by moving the h323 clean + commands down into channels/makefile. + +2010-06-02 22:28 +0000 [r267350] Richard Mudgett + + * main/channel.c, configure, include/asterisk/autoconfig.h.in, + configure.ac, include/asterisk/channel.h, CHANGES, + channels/sig_pri.c: Add ETSI Malicious Call ID support. Add the + ability to report malicious callers as an AMI event in the call + event class. Relevant specification: EN 300 180 Review: + https://reviewboard.asterisk.org/r/576/ + +2010-06-02 21:44 +0000 [r267303-267305] Russell Bryant + + * utils/extconf.c: Fix a build error on mac. + + * main/Makefile: Ensure the -Wno-strict-aliasing flag makes it, + even if ASTCFLAGS has been specified. When ASTCFLAGS was + specified with the make command, Makefile.rules was using the + specified value from the command line and not the one here, + making it so this flag would go missing. + +2010-06-02 21:05 +0000 [r267261] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, configure, + include/asterisk/autoconfig.h.in, configure.ac, CHANGES, + channels/sig_pri.c: Add ETSI Call Waiting support. Add the + ability to announce a call to an endpoint when there are no B + channels available. A call waiting call is a SETUP message with + no B channel selected. Relevant specification: EN 300 056, EN 300 + 057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan + function now supports the "no_media_path" option. * Returns "0" + if there is a B channel associated with the call. * Returns "1" + if no B channel is associated with the call. The call is either + on hold or is a call waiting call. If you are going to allow + incoming call waiting calls then you need to use + CHANNEL(no_media_path) do determine if you must drop a call to + accept the new call. Review: + https://reviewboard.asterisk.org/r/568/ + +2010-06-02 19:33 +0000 [r267181] David Vossel + + * CHANGES, doc/advice_of_charge.txt: Update CHANGES and aoc help + doc to reflect AOC additions + +2010-06-02 18:53 +0000 [r267138] Russell Bryant + + * main/cli.c: Add a CLI command that blocks until Asterisk has + fully booted. Review: https://reviewboard.asterisk.org/r/684/ + +2010-06-02 18:13 +0000 [r267097] Mark Michelson + + * channels/chan_sip.c: Prevent use of uninitialized values. Two + struct sockaddr_ins are created when applying directmedia host + access rules. The addresses of these are passed to the RTP engine + to be filled in. However, the RTP engine inspects the fields of + the structs before actually taking action. This inspection caused + valgrind to be a bit unhappy. + +2010-06-02 18:10 +0000 [r267096] Richard Mudgett + + * apps/app_dial.c, configs/chan_dahdi.conf.sample, + include/asterisk/aoc.h (added), channels/chan_sip.c, + configs/manager.conf.sample, main/aoc.c (added), + apps/app_queue.c, channels/sig_pri.c, doc/advice_of_charge.txt + (added), main/channel.c, channels/sig_pri.h, + channels/chan_dahdi.c, main/manager.c, main/features.c, + tests/test_aoc.c (added), configs/sip.conf.sample, + include/asterisk/frame.h, main/asterisk.c, + channels/sip/include/sip.h: Generic Advice of Charge. Asterisk + Generic AOC Representation - Generic AOC encode/decode routines. + (Generic AOC must be encoded to be passed on the wire in the + AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent + generic encoded AOC data - Manager events for AOC-S, AOC-D, and + AOC-E messages Asterisk App Support - app_dial AOC-S pass-through + support on call setup - app_queue AOC-S pass-through support on + call setup AOC Unit Tests - AOC Unit Tests for encode/decode + routines - AOC Unit Test for manager event representation. SIP + AOC Support - Pass-through of generic AOC-D and AOC-E messages to + snom phones via the snom AOC specification. - Creation of + chan_sip page3 flags for the addition of the new + 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively + supports AOC pass-through through the use of the new + AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC + Pass-through support - 'aoc_enable' chan_dahdi.conf option for + independently enabling pass-through of AOC-S, AOC-D, AOC-E. - + 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - + DAHDI A() dial string option for requesting AOC services. example + usage: ;requests AOC-S, AOC-D, and AOC-E on call setup + exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: + https://reviewboard.asterisk.org/r/552/ + +2010-06-02 17:57 +0000 [r267093] Russell Bryant + + * apps/app_voicemail.c: Silence a compiler warning. + +2010-06-02 17:29 +0000 [r267065] Jeff Peeler + + * include/asterisk/slin.h: Fix infinite loop when loading codec + speex This changes the sample slinear frame data to contain + non-zero data so that translation calculations for speex works + when preprocessing and VAD is turned on. The encoder expects + samples to be returned, but when attempted with the mentioned two + options and silent sample frames everything was discarded. + (closes issue #17240) Reported by: seandarcy Review: + https://reviewboard.asterisk.org/r/682/ + +2010-06-02 17:25 +0000 [r267041] Paul Belanger + + * /, main/ast_expr2.y: Merged revisions 267009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r267009 | pabelanger | 2010-06-02 13:14:37 -0400 (Wed, 02 Jun + 2010) | 7 lines Cleanup error/warning messages in AEL2 parser + (closes issue #16684) Reported by: Silmaril Patches: + patch_ael2_logmsg.diff uploaded by Silmaril (license 979) + ........ + +2010-06-02 17:13 +0000 [r266926-267008] Richard Mudgett + + * main/manager.c, configure, include/asterisk/autoconfig.h.in, + configure.ac, configs/manager.conf.sample, CHANGES, + channels/sig_pri.c, include/asterisk/manager.h: Add ETSI Advice + Of Charge (AOC) event reporting. This feature generates AMI + events in the new aoc event class from the events passed up by + libpri. Review: https://reviewboard.asterisk.org/r/537/ + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, configure, + include/asterisk/autoconfig.h.in, configure.ac, CHANGES, + channels/sig_pri.c: Add ETSI Explicit Call Transfer (ECT) + support. Added ability to send and receive ETSI Explicit Call + Transfer (ECT) messages to eliminate tromboned calls. Note: + Asterisk already supported initiating the transfer of calls to + eliminate tromboned calls to libpri so there was nothing to do + for the asterisk portion. Review: + https://reviewboard.asterisk.org/r/520/ + +2010-06-02 13:32 +0000 [r266877] Paul Belanger + + * main/bridging.c: pthread_join to assure the thread is really gone + (closes issue #15465) Reported by: fnordian Patches: + bridging.patch uploaded by fnordian (license 110) Tested by: + lmadsen, fnordian, peterh Review: + https://reviewboard.asterisk.org/r/679/ + +2010-06-01 22:14 +0000 [r266832] Terry Wilson + + * res/res_calendar_exchange.c: Use the correct ical.h file + +2010-06-01 21:28 +0000 [r266828] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, tests/test_locale.c + (added), configure.ac, configs/voicemail.conf.sample, + include/asterisk/localtime.h, main/stdtime/localtime.c, CHANGES, + apps/app_voicemail.c: Support setting locale per-mailbox (changes + date/time languages for email, pager messages). (closes issue + #14333) Reported by: klaus3000 Patches: + 20090515__issue14333.diff.txt uploaded by tilghman (license 14) + app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by + klaus3000 (license 65) Tested by: klaus3000 + +2010-06-01 21:12 +0000 [r266786] Terry Wilson + + * apps/app_dial.c, UPGRADE.txt: Set app and appdata fields when a + Dial is redirected (closes issue #17204) Reported by: one47 + Tested by: twilson, one47 + +2010-06-01 18:02 +0000 [r266592-266735] Tilghman Lesher + + * res/res_smdi.c: Don't register functions until the last possible + point, so they're not unloaded unnecessarily. (closes issue + #15996) Reported by: junky Patches: sdmi_wait.diff uploaded by + junky (license 177) + + * main/manager.c: Eliminate stale manager events after a set + interval, even if AMI clients don't query for them. Actions (or + failures to act) by external clients should not cause memory + leaks in Asterisk, especially when those continued leaks could + cause Asterisk to misbehave later. (closes issue #17234) Reported + by: mav3rick Patches: 20100510__issue17234.diff.txt uploaded by + tilghman (license 14) 20100517__issue17234__trunk.diff.txt + uploaded by tilghman (license 14) Tested by: mav3rick, davidw + (closes issue #17365) Reported by: davidw + + * /, main/asterisk.c: Merged revisions 266585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) + | 11 lines Prevent CLI prompt from distorting output of lines + shorter than the prompt. Uses the VT100 method of clearing the + line from the cursor position to the end of the line: Esc-0K + (closes issue #17160) Reported by: coolmig Patches: + 20100531__issue17160.diff.txt uploaded by tilghman (license 14) + Tested by: coolmig ........ + +2010-05-30 20:18 +0000 [r266438-266522] Tilghman Lesher + + * funcs/func_env.c: Needs to be wrapped in + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 266437 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 May 2010) + | 2 lines Reverting patch and reopening issue #16784, as patch + breaks color display. ........ + +2010-05-28 22:54 +0000 [r266386] Terry Wilson + + * res/res_calendar_icalendar.c, configure, configure.ac, + res/res_calendar_caldav.c: Fix ical library handling (again) + Newer versions of libical (which we require) store the header + file in a libical/ subfolder and include an ical.h file that does + a #warning for deprecation and then #includes . + Since we now test for libical/ical.h, we can change the #includes + back to and remove the test which specifically + adds /usr/include/libical as an include directory. + +2010-05-28 22:50 +0000 [r266337-266385] Tilghman Lesher + + * funcs/func_env.c, UPGRADE.txt, main/asterisk.c: Setup environment + variables for the benefit of child processes and disallow + changing them. (closes issue #14899) Reported by: jmls Patches: + 20090916__issue14899.diff.txt uploaded by tilghman (license 14) + Tested by: jmls + + * main/asterisk.c: Only report swap on platforms which can examine + those statistics + +2010-05-28 17:55 +0000 [r266292] David Vossel + + * channels/chan_sip.c: fixes crash when creation of UDPTL fails + (closes issue #17264) Reported by: falves11 Patches: + issue_17264_reviewboard_fix.diff uploaded by dvossel (license + 671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel + (license 671) Tested by: falves11 + +2010-05-28 17:34 +0000 [r266289] Terry Wilson + + * configure, configure.ac, makeopts.in: More build fixes for + ical/neon and res_calendar_ews + +2010-05-27 20:08 +0000 [r266240] Jeff Peeler + + * pbx/pbx_realtime.c: fix compile error + +2010-05-27 19:25 +0000 [r266146-266238] Tilghman Lesher + + * pbx/pbx_realtime.c, CHANGES: Cache query results for one second. + Queries from the PBX core come in 3's. Caching avoids the + additional performance penalty from those two additional queries + hitting the database. (closes issue #16521) Reported by: tilghman + Patches: 20091229__issue16521.diff.txt uploaded by tilghman + (license 14) Tested by: Hubguru, tilghman + + * /, main/logger.c, utils/extconf.c, main/asterisk.c: Merged + revisions 266142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) + | 14 lines Use sigaction for signals which should persist past + the initial trigger, not signal. If you call signal() in a + Solaris signal handler, instead of just resetting the signal + handler, it causes the signal to refire, because the signal is + not marked as handled prior to the signal handler being called. + This effectively causes Solaris to immediately exceed the + threadstack in recursive signal handlers and crash. (closes issue + #17000) Reported by: rmcgilvr Patches: + 20100526__issue17000.diff.txt uploaded by tilghman (license 14) + Tested by: rmcgilvr ........ + +2010-05-26 20:17 +0000 [r266092-266098] Mark Michelson + + * apps/app_dial.c: Remove redundant ast_conntected_line_free call. + This wouldn't cause any problems, but it's certainly not needed + either. + + * res/res_musiconhold.c: Remove unrelated MOH change from previous + commit. Thanks Kevin! + + * main/channel.c, res/res_musiconhold.c: Fix misspelling of macro + args. + +2010-05-26 19:46 +0000 [r266006-266090] David Vossel + + * channels/chan_sip.c, main/app.c, channels/sip/config_parser.c, + channels/sip/include/sip.h: do all sip registry parsing before + transmit_register This patch breaks up every part of the sip + registry string during config parsing and removes all parsing + from transmit_register(). Thanks to Nick_Lewis for contributing + this patch! (closes issue #14331) Reported by: Nick_Lewis + Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis + (license 657) chan_sip.c.patch uploaded by Nick Lewis (license + 657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis + (license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis + (license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis + (license 657) nicklewispatch.diff uploaded by dvossel (license + 671) Tested by: Nick_Lewis, dvossel Review: + https://reviewboard.asterisk.org/r/628/ + + * channels/chan_sip.c: fixes failed SIP Directed pickup resulting + in dead channel (closes issue #17339) Reported by: one47 Patches: + sip_magic_pickup2 uploaded by one47 (license 23) Tested by: + one47, dvossel + +2010-05-26 16:23 +0000 [r265894-265923] Tilghman Lesher + + * res/res_config_pgsql.c, /: Merged revisions 265910 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 + May 2010) | 7 lines Not finding rows in the DB does not rise to + the level of a warning. (closes issue #17062) Reported by: + drookie Patches: 20100525__issue17062.diff.txt uploaded by + tilghman (license 14) ........ + + * res/res_config_pgsql.c, configs/res_pgsql.conf.sample: Construct + socket name, according to the Postgres docs, and document as + such. (closes issue #17392) Reported by: dps Patches: + 20100525__issue17392.diff.txt uploaded by tilghman (license 14) + Tested by: dps + +2010-05-26 14:45 +0000 [r265842-265844] Mark Michelson + + * channels/chan_sip.c: ....... + + * channels/chan_sip.c: Re-enable "always" option for videosupport + option in sip.conf. (closes issue #17016) Reported by: twilson + Patches: 17016.patch uploaded by mmichelson (license 60) Tested + by: devmod + +2010-05-26 05:33 +0000 [r265793] Terry Wilson + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, + res/res_calendar_ews.c: Ensure that libneon > 0.29.0 is installed + for res_calendar_ews This uses a modified version of pabelanger's + patch that checks for NTLM support instead, which was added in + 0.29.0 which is what is required for res_calendar_ews. (closes + issue #17391) Reported by: loloski Patches: issue17391.patch.v2 + uploaded by pabelanger (license 224) Tested by: twilson + +2010-05-26 00:29 +0000 [r265747] Tilghman Lesher + + * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, + configure, include/asterisk/autoconfig.h.in, configure.ac, + pbx/pbx_lua.c, res/res_calendar_caldav.c, res/res_calendar_ews.c: + Use configure to determine the prefixes and include directories + properly. This ensures cross-platform compatibility, even among + Linux distributions, which don't always put headers in the same + place. (closes issue #17391) Reported by: loloski + +2010-05-25 20:59 +0000 [r265698] Mark Michelson + + * channels/chan_sip.c: Properly use peer's outboundproxy for + outbound REGISTERs. The logic used in transmit_register to get + the outboundproxy for a peer was flawed since this value would be + overridden shortly afterwards when create_addr was called. In + addition, this also fixes some logic used when parsing users.conf + so that the peer name is placed in the internally-generated + register string so that an outboundproxy set in the Asterisk GUI + will be used for outbound REGISTERs. + +2010-05-25 17:00 +0000 [r265611] Matthew Nicholson + + * /, apps/app_queue.c: Merged revisions 265610 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May + 2010) | 8 lines Don't mark the cdr records of unanswered queue + calls with "NOANSWER". This restores the behavior prior to + r258670. (closes issue #17334) Reported by: jvandal Patches: + queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested + by: aragon, jvandal ........ + +2010-05-25 16:23 +0000 [r265608] Richard Mudgett + + * main/channel.c: Memory leak in connected line data when SIP blond + transfer done. The handling of the control subclass + AST_CONTROL_READ_ACTION frame leaked connected line string memory + in __ast_read(). Also in __ast_read() the frame type switch + should not have had a case for AST_CONTROL_READ_ACTION. + AST_CONTROL_READ_ACTION is not a frame type. + +2010-05-25 08:31 +0000 [r265525] Tzafrir Cohen + + * addons/ooh323c/src/oochannels.c: Typos: 'succesful' (lintian) + +2010-05-24 22:21 +0000 [r265467] Terry Wilson + + * doc/manager_1_1.txt, main/manager.c, main/asterisk.c: Merge the + rest of the FullyBooted patch + +2010-05-24 22:16 +0000 [r265449-265453] Mark Michelson + + * apps/app_senddtmf.c: Allow SendDTMF to play digits to a specified + channel. Patch supplied by reporter was modified to use + autoservice and prevent a potential channel ref leak but is + otherwise as the reporter uploaded it. (closes issue #17182) + Reported by: rcasas Patches: app_senddtmf.c.patch_trunk uploaded + by rcasas (license 641) + + * channels/h323/ast_h323.cxx: Print openh323 log to the Asterisk + console. (closes issue #17109) Reported by: under Patches: + logstream.diff uploaded by under (license 914) + + * channels/chan_sip.c: Allow type=user SIP endpoints to be loaded + properly from realtime. (closes issue #16021) Reported by: + Guggemand Patches: realtime-type-fix.patch uploaded by Guggemand + (license 897) (altered by me slightly to avoid ref leaks) Tested + by: Guggemand + +2010-05-24 20:08 +0000 [r265367] Richard Mudgett + + * apps/app_rpt.c: Make app_rpt.c able to compile again. + +2010-05-24 19:42 +0000 [r265366] David Vossel + + * channels/chan_sip.c: reverses incorrect logic introduced by + r243200 The decoding of the replace_id did not need to be broken + up in this instance. This was brought to my attention again + because it caused a segfault when the from or to tags were not + present in the "Replaces" header. + +2010-05-24 19:06 +0000 [r265317-265320] Terry Wilson + + * doc/tex/manager.tex: Add the FullyBooted AMI event It is possible + to connect to the manager interface before all Asterisk modules + are loaded. To ensure that an application does not send AMI + actions that might require a module that has not yet loaded, the + application can listen for the FullyBooted manager event. It will + be sent upon connection if all modules have been loaded, or as + soon as loading is complete. The event: Event: FullyBooted + Privilege: system,all Status: Fully Booted Review: + https://reviewboard.asterisk.org/r/639/ + + * CREDITS, configs/calendar.conf.sample, CHANGES, + res/res_calendar_ews.c (added), res/res_calendar.c: Calendaring + support for Exchange Server 2007+ via EWS This commit adds + support for calendaring with Exchange Server 2007+ via Exchange + Web Services. Full write support and for querying attendees. Many + thanks to Jan Kaláb for the feature. (closes issue #17022) + Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel + (license 1008) Tested by: pitel, twilson Review: + https://reviewboard.asterisk.org/r/557/ Review: + https://reviewboard.asterisk.org/r/668/ + +2010-05-24 18:19 +0000 [r265316] Tilghman Lesher + + * main/asterisk.c: On systems with a LOT of RAM, a signed integer + sometimes printed negative. (closes issue #16837) Reported by: + jlpedrosa Patches: 20100504__issue16837.diff.txt uploaded by + tilghman (license 14) + +2010-05-24 16:10 +0000 [r265273] David Vossel + + * main/channel.c: fixes segfault when using generic plc + +2010-05-23 18:23 +0000 [r265227] Alexandr Anikin + + * addons/chan_ooh323.c: small changes to avoiding 'freeing unused + memory...' + +2010-05-21 22:46 +0000 [r265174] Richard Mudgett + + * main/channel.c: Channel initialization failure causes crashes. + __ast_channel_alloc_ap() has several points in the initialization + of a new channel structure where it could fail. Since the channel + structure is now an ao2 object, the destructor callback needs to + be able to handle clean up when the structure setup is + incomplete. Problems corrected: 1) Failing to setup the alertpipe + would not unreference the structure but free it directly. Doing + this to an ao2_object is very bad. 2) File descriptors need to be + initialized to -1 before a construction failure could occur so + the destructor will not close unopened descriptors. 3) The + destructor needs to check that the string field has been + initialized before using any string field values. Crashes + expected. 4) The destructor should not notify devstate if the + device name is empty. It is a waste of cycles and a couple ERROR + log messages are generated. Review: + https://reviewboard.asterisk.org/r/675/ + +2010-05-21 21:08 +0000 [r264953-265090] Mark Michelson + + * include/asterisk/file.h, /, apps/app_queue.c: Merged revisions + 265089 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May + 2010) | 8 lines Don't hang up on a queue caller if the file we + attempt to play does not exist. This also fixes a documentation + mistake in file.h that made my original attempt to correct this + problem not work correctly. (closes issue #17061) Reported by: + RoadKill ........ + + * channels/chan_sip.c: Be sure to set the sin_family on the proxy + when allocating. (closes issue #17157) Reported by: stuarth + + * /, include/asterisk/channel.h: Merged revisions 264999 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, 21 May + 2010) | 3 lines Fix grammatical error in comment. ........ + + * main/channel.c, main/autoservice.c, /, + include/asterisk/channel.h: Merged revisions 264996 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, + 21 May 2010) | 32 lines Allow ast_safe_sleep to defer specific + frames until after the sleep has concluded. From reviewboard + Background: A Digium customer discovered a somewhat odd bug. The + setup is that parties A and B are bridged, and party A places + party B on hold. While party B is listening to hold music, he + mashes a bunch of DTMF. Party A takes party B off hold while this + is happening, but party B continues to hear hold music. I could + reproduce this about 1 in 5 times. The issue: When DTMF features + are enabled and a user presses keys, the channel that the DTMF is + streamed to is placed in an ast_safe_sleep for 100 ms, the + duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is + read from the channel during the sleep, the frame is dropped. + Thus the unhold indication is never made to the channel that was + originally placed on hold. The fix: Originally, I discussed with + Kevin possible ways of fixing the specific problem reported. + However, we determined that the same type of problem could happen + in other situations where ast_safe_sleep() is used. Using + autoservice as a model, I modified ast_safe_sleep_conditional() + to defer specific frame types so they can be re-queued once the + sleep has finished. I made a common function for determining if a + frame should be deferred so that there are not two identical + switch blocks to maintain. Review: + https://reviewboard.asterisk.org/r/674/ ........ + + * res/res_fax.c, include/asterisk/res_fax.h, + res/res_fax.exports.in, res/res_fax_spandsp.c: Log spandsp's fax + debug output to the FAX logger level. Review: + https://reviewboard.asterisk.org/r/658 + +2010-05-21 01:00 +0000 [r264905] Terry Wilson + + * channels/chan_sip.c: Take dup'd code for directmedia ACLs and + make utility func The same code was repeated in lots of different + places, so I made a utility fuction for it. This should make the + merge in the v6-new branch easier. + +2010-05-20 23:29 +0000 [r264828] Richard Mudgett + + * /, main/callerid.c: Merged revisions 264820 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010) + | 6 lines ast_callerid_parse() had a path that left name + uninitialized. Several callers of ast_callerid_parse() do not + initialize the name parameter before calling thus there is the + potential to use an uninitialized pointer. ........ + +2010-05-20 22:23 +0000 [r264752-264779] Tilghman Lesher + + * main/pbx.c: Let ExtensionState resolve dynamic hints. (closes + issue #16623) Reported by: tilghman Patches: + 20100116__issue16623.diff.txt uploaded by tilghman (license 14) + Tested by: lmadsen + + * apps/app_stack.c: Error message fix. (closes issue #17356) + Reported by: kenner Patches: app_stack.c.diff uploaded by kenner + (license 1040) + +2010-05-20 20:49 +0000 [r264669-264711] Richard Mudgett + + * main/ccss.c: Avoid crash in generic CC agent init if caller name + or number is NULL. + + * apps/app_dial.c, apps/app_queue.c: Dial and queue connected line + update macro not always run when expected. The connected line + update macro would not get run if the connected line number + string was empty. The number could be empty if the connected line + update did not update a number but the name. It should be run if + there was an AST_CONTROL_CONNECTED_LINE frame received for + pending dials and queues. Renamed and added some more comments + for some confusing identifiers directly connected to the related + code. Also fixed a memory leak in app_queue. Review: + https://reviewboard.asterisk.org/r/669/ + +2010-05-20 17:54 +0000 [r264626] Terry Wilson + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: Add support for direct media ACLs + directmediapermit/directmediadeny support to restrict which peers + can do directmedia based on ip address. In some networks not all + phones are fully routed, i.e. not all phones can ping each other. + This patch adds a way to restrict directmedia for certain peers + between certain networks. (closes issue #16645) Reported by: + raarts Patches: directmediapermit.patch uploaded by raarts + (license 937) Tested by: raarts Review: + https://reviewboard.asterisk.org/r/467/ + +2010-05-20 15:30 +0000 [r264497-264540] Kevin P. Fleming + + * addons/ooh323c/src/h323, addons/ooh323c/src: Ignore pre-processed + source files generated during DONT_OPTIMIZE dev-mode builds. + + * main/logger.c: Correct 'all logger levels' patch to work + properly. Nick Lewis pointed out that the patch as committed + wouldn't actually include dynamic logger levels, which was missed + by the other reviewers. Thanks! + +2010-05-19 21:29 +0000 [r264452] Mark Michelson + + * main/channel.c, channels/chan_sip.c, include/asterisk/_private.h, + include/asterisk/options.h, main/asterisk.c, main/loader.c: Fix + transcode_via_sln option with SIP calls and improve PLC usage. + From reviewboard: The problem here is a bit complex, so try to + bear with me... It was noticed by a Digium customer that generic + PLC (as configured in codecs.conf) did not appear to actually be + having any sort of benefit when packet loss was introduced on an + RTP stream. I reproduced this issue myself by streaming a file + across an RTP stream and dropping approx. 5% of the RTP packets. + I saw no real difference between when PLC was enabled or disabled + when using wireshark to analyze the RTP streams. After analyzing + what was going on, it became clear that one of the problems faced + was that when running my tests, the translation paths were being + set up in such a way that PLC could not possibly work as + expected. To illustrate, if packets are lost on channel A's read + stream, then we expect that PLC will be applied to channel B's + write stream. The problem is that generic PLC can only be done + when there is a translation path that moves from some codec to + SLINEAR. When I would run my tests, I found that every single + time, read and write translation paths would be set up on channel + A instead of channel B. There appeared to be no real way to + predict which channel the translation paths would be set up on. + This is where Kevin swooped in to let me know about the + transcode_via_sln option in asterisk.conf. It is supposed to work + by placing a read translation path on both channels from the + channel's rawreadformat to SLINEAR. It also will place a write + translation path on both channels from SLINEAR to the channel's + rawwriteformat. Using this option allows one to predictably set + up translation paths on all channels. There are two problems with + this, though. First and foremost, the transcode_via_sln option + did not appear to be working properly when I was placing a SIP + call between two endpoints which did not share any common + formats. Second, even if this option were to work, for PLC to be + applied, there had to be a write translation path that would go + from some format to SLINEAR. It would not work properly if the + starting format of translation was SLINEAR. The one-line change + presented in this review request in chan_sip.c fixed the first + issue for me. The problem was that in sip_request_call, the + jointcapability of the outbound channel was being set to the + format passed to sip_request_call. This is nativeformats of the + inbound channel. Because of this, when + ast_channel_make_compatible was called by app_dial, both channels + already had compatibly read and write formats. Thus, no + translation path was set up at the time. My change is to set the + jointcapability of the sip_pvt created during sip_request_call to + the intersection of the inbound channel's nativeformats and the + configured peer capability that we determined during the earlier + call to create_addr. Doing this got the translation paths set up + as expected when using transcode_via_sln. The changes presented + in channel.c fixed the second issue for me. First and foremost, + when Asterisk is started, we'll read codecs.conf to see the value + of the genericplc option. If this option is set, and ast_write is + called for a frame with no data, then we will attempt to fill in + the missing samples for the frame. The implementation uses a + channel datastore for maintaining the PLC state and for creating + a buffer to store PLC samples in. Even when we receive a frame + with data, we'll call plc_rx so that the PLC state will have + knowledge of the previous voice frame, which it can use as a + basis for when it comes time to actually do a PLC fill-in. So, + reviewers, now I ask for your help. First off, there's the one + line change in chan_sip that I have put in. Is it right? By my + logic it seems correct, but I'm sure someone can tell me why it + is not going to work. This is probably the change I'm least + concerned about, though. What concerns me much more is the set of + changes in channel.c. First off, am I even doing it right? When I + run tests, I can clearly see that when PLC is activated, I see a + significant increase in RTP traffic where I would expect it to + be. However, in my humble opinion, the audio sounds kind of + crappy whenever the PLC fill-in is done. It sounds worse to me + than when no PLC is used at all. I need someone to review the + logic I have used to be sure that I'm not misusing anything. As + far as I can see my pointer arithmetic is correct, and my use of + AST_FRIENDLY_OFFSET should be correct as well, but I'm sure + someone can point out somewhere where I've done something + incorrectly. As I was writing this review request up, I decided + to give the code a test run under valgrind, and I find that for + some reason, calls to plc_rx are causing some invalid reads. + Apparently I'm reading past the end of a buffer somehow. I'll + have to dig around a bit to see why that is the case. If it's + obvious to someone reviewing, speak up! Finally, I have one other + proposal that is not reflected in my code review. Since without + transcode_via_sln set, one cannot predict or control where a + translation path will be up, it seems to me that the current + practice of using PLC only when transcoding to SLINEAR is not + useful. I recommend that once it has been determined that the + method used in this code review is correct and works as expected, + then the code in translate.c that invokes PLC should be removed. + Review: https://reviewboard.asterisk.org/r/622/ + +2010-05-19 20:30 +0000 [r264400] David Vossel + + * main/udptl.c: fixes infinite loop during udptl.c's + decode_open_type When decode_length returns the length there is a + check to see if that length is negative, if so the decode loop + breaks as this means the limit has been reached. The problem here + is that length is an unsigned int, so length can never be + negative. This resulted in an infinite loop. (issue #17352) + +2010-05-19 20:26 +0000 [r264335-264379] Matthew Nicholson + + * main/udptl.c: Cast an unsigned int to a signed int when comparing + it with 0. (AST-377) + + * /, apps/app_speech_utils.c: Merged revisions 264334 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, + 19 May 2010) | 5 lines Set quieted flag when receiving a dtmf + tone during playback in speechbackground. (closes issue #16966) + Reported by: asackheim ........ + +2010-05-19 19:21 +0000 [r264331] David Vossel + + * channels/chan_sip.c: fixes crash in check_rtp_timeout During + deadlock avoidance the sip dialog pvt is locked and unlocked. + When this occurs we have no guarantee the pvt's owner is still + valid. We were trying to access the pvt's owner after this + without checking to see if it still existed first. (closes issue + #17271) Reported by: under Patches: check_rtp_timeout.diff + uploaded by under (license 914) Tested by: dvossel + +2010-05-19 17:48 +0000 [r264204-264249] Tilghman Lesher + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/options.h: Merged revisions 264248 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 + May 2010) | 17 lines Internal timing is now on by default, if + you're using DAHDI 2.3 or above. The reason for ensuring DAHDI + 2.3 or above is that this version ensures that a timer is always + available, whereas in previous versions, it was possible for + DAHDI to be loaded, but have no drivers to actually generate + timing. If internal_timing was turned on in this circumstance, a + complete lack of audio would result. This is the reason why + internal_timing was not on by default. However, now that DAHDI + ensures the availability of a timer, there is no reason for this + setting to be off (and in fact, it solves a great many initial + user problems). (closes issue #15932) Reported by: dimas Patches: + 20100519__issue15932.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ + + * main/dsp.c: Keep track of digit duration, when we're decoding + inband to pass DTMF frames. (closes issue #17235) Reported by: + frawd Patches: new_dtmf_dsp_len.patch uploaded by frawd (license + 610) 20100518__issue17235.diff.txt uploaded by tilghman (license + 14) Tested by: frawd + +2010-05-19 15:39 +0000 [r264161] Leif Madsen + + * main/cli.c: Fix compilation problem with previous commit. (issue + #16009) + +2010-05-19 15:29 +0000 [r264160] Kevin P. Fleming + + * main/logger.c, configs/logger.conf.sample: Add ability for logger + channels to include *all* levels. Now that Asterisk modules can + dynamically create and destroy logger levels on demand, it's + useful to be able to configure a logger channel (console, file, + whatever) to be able to accept log messages from *all* levels, + even levels created dynamically. This patch adds support for + this, by allowing the '*' level name to be used in logger.conf. + Review: https://reviewboard.asterisk.org/r/663/ + +2010-05-19 15:12 +0000 [r264117] Leif Madsen + + * CHANGES, main/cli.c: Add ability to hangup all channels from the + CLI. Added the keyword 'all' to the 'channel hangup request' CLI + command so that you can request all channels to be hungup without + having to restart Asterisk. (closes issue #16009) Reported by: + moy Patches: hangup-all-rev-221688.patch uploaded by moy (license + 222) Tested by: moy, russell + +2010-05-19 14:38 +0000 [r264114] David Vossel + + * res/res_rtp_asterisk.c: fixes crash during dtmf During the + processing of Cisco dtmf the dtmf samples were not being + calculated correctly. In an attempt to determine what sample rate + was being used, a NULL frame was processed which caused a crash. + This patch resolves this. (closes issue #17248) Reported by: + falves11 Patches: issue_17248.diff uploaded by dvossel (license + 671) + +2010-05-19 08:09 +0000 [r264031] Alec L Davis + + * configs/indications.conf.sample: fix incorrectly typed + indications for [nz] stutter and dialrecall (closes issue #17359) + Reported by: alecdavis Patches: bug17359.diff.txt uploaded by + alecdavis (license 585) + +2010-05-19 06:41 +0000 [r263905-263950] Tilghman Lesher + + * /, main/dsp.c: Merged revisions 263949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010) + | 8 lines Because progress is called multiple times, across + several frames, we must persist states when detecting multitone + sequences. (closes issue #16749) Reported by: dant Patches: + dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by: + dant ........ + + * configure, configure.ac, build_tools/sha1sum-sh (added), + makeopts.in, sounds/Makefile: Add an sha1sum-workalike for + platforms which don't have it (like Mac OS X) + +2010-05-18 22:48 +0000 [r263904] David Vossel + + * main/strings.c: fixes segfault on logging (closes issue #17331) + Reported by: under Patches: utils.diff uploaded by under (license + 914) segfault_on_logging.diff uploaded by dvossel (license 671) + Tested by: under, dvossel + +2010-05-18 21:09 +0000 [r263860] Mark Michelson + + * channels/chan_sip.c: Be sure to heap-allocate the redirecting to + tag so as not to cause crashiness. + +2010-05-18 20:49 +0000 [r263858] Tilghman Lesher + + * res/res_timing_kqueue.c: Make happy green color come back + +2010-05-18 20:09 +0000 [r263810] Mark Michelson + + * channels/chan_sip.c: Fix memory leaks in redirecting structures + in chan_sip.c Thanks to Richard for pointing this out. + +2010-05-18 19:30 +0000 [r263807-263808] Jeff Peeler + + * CHANGES: put changes with the correct version + + * /, CHANGES, apps/app_directory.c: Merged revisions 263769 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) + | 10 lines Modify directory name reading to be interrupted with + operator or pound escape. In the case of accidentally entering + the wrong first three letters for the reading, users could be + very frustrated if the name listing is very long. This allows + interrupting the reading by pressing 0 or #. 0 will attempt to + execute a configured operator (o) extension and # will exit and + proceed in the dialplan. ABE-2200 ........ + +2010-05-17 23:49 +0000 [r263724] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + makeopts.in, sounds/Makefile, autoconf/ast_ext_lib.m4: Cache + sound tarfiles in a common directory, such that a clean reinstall + does not force a re-download of the tarballs. (closes issue + #15370) Reported by: pprindeville Patches: + asterisk-trunk-bugid15370.patch uploaded by pprindeville (license + 347) Tested by: pprindeville, tilghman, seanbright + +2010-05-17 22:08 +0000 [r263640] Mark Michelson + + * /, main/devicestate.c: Merged revisions 263639 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May + 2010) | 10 lines Fix logic error when checking for a devstate + provider. When using strsep, if one of the list of specified + separators is not found, it is the first parameter to strsep + which is now NULL, not the pointer returned by strsep. This issue + isn't especially severe in that the worst it is likely to do is + waste some cycles when a device with no '/' and no ':' is passed + to ast_device_state. ........ + +2010-05-17 19:31 +0000 [r263589] Tilghman Lesher + + * apps/app_voicemail.c: With IMAP backend, messages in INBOX were + counted twice for MWI. (closes issue #17135) Reported by: + edhorton Patches: 20100513__issue17135.diff.txt uploaded by + tilghman (license 14) 17135_2.diff uploaded by ebroad (license + 878) Tested by: edhorton, ebroad + +2010-05-17 15:36 +0000 [r263541] Mark Michelson + + * apps/app_dial.c, channels/chan_local.c, main/rtp_engine.c, + channels/chan_sip.c, include/asterisk/channel.h, + configs/misdn.conf.sample, apps/app_queue.c, + funcs/func_redirecting.c, channels/misdn_config.c, + main/channel.c, main/dial.c, channels/chan_dahdi.c, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, + channels/misdn/chan_misdn_config.h, main/features.c, + funcs/func_connectedline.c, include/asterisk/frame.h, + funcs/func_callerid.c, channels/sip/include/sip.h: Enhancements + to connected line and redirecting work. From reviewboard: Digium + has a commercial customer who has made extensive use of the + connected party and redirecting information present in later + versions of Asterisk Business Edition and which is to be in the + upcoming 1.8 release. Through their use of the feature, new + problems and solutions have come about. This patch adds several + enhancements to maximize usage of the connected party and + redirecting information functionality. First, Asterisk trunk + already had connected line interception macros. These macros + allow you to manipulate connected line information before it was + sent out to its target. This patch adds the same feature except + for redirecting information instead. Second, the ast_callerid and + ast_party_id structures have been enhanced to provide a "tag." + This tag can be set with func_callerid, func_connectedline, + func_redirecting, and in the case of DAHDI, mISDN, and SIP + channels, can be set in a configuration file. The idea behind the + callerid tag is that it can be set to whatever value the + administrator likes. Later, when running connected line and + redirecting macros, the admin can read the tag off the + appropriate structure to determine what action to take. You can + think of this sort of like a channel variable, except that + instead of having the variable associated with a channel, the + variable is associated with a specific identity within Asterisk. + Third, app_dial has two new options, s and u. The s option lets a + dialplan writer force a specific caller ID tag to be placed on + the outgoing channel. The u option allows the dialplan writer to + force a specific calling presentation value on the outgoing + channel. Fourth, there is a new control frame subclass called + AST_CONTROL_READ_ACTION added. This was added to correct a very + specific situation. In the case of SIP semi-attended (blond) + transfers, the party being transferred would not have the + opportunity to run a connected line interception macro to + possibly alter the transfer target's connected line information. + The issue here was that during a blond transfer, the SIP transfer + code has no bridged channel on which to queue the connected line + update. The way this was corrected was to add this new control + frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on + the channel on which the connected line interception macro should + be run. When ast_read is called to read the frame, ast_read + responds by calling a callback function associated with the + specific read action the control frame describes. In this case, + the action taken is to run the connected line interception macro + on the transferee's channel. Review: + https://reviewboard.asterisk.org/r/652/ + +2010-05-17 15:14 +0000 [r263375-263460] Leif Madsen + + * main/manager.c: Missing newlines added to Set-Cookie line in + manager.c Sean Bright pointed out that we lost a set of newline + characters in commit 190349 on a line I had recently changed. Yay + for code review on commits. (issue #17231, #10961) + + * main/manager.c, /: Recorded merge of revisions 263456 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010) + | 11 lines Manager cookies are not compatible with RFC2109. The + Version field in the cookies we're setting contain quotes around + the version number which is not compatible with RFC2109 and + breaks some implementations. (closes issue #17231) Reported by: + ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by + ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by + ecarruda (license 559) Tested by: ecarruda, russell ........ + + * /, sounds/Makefile: Merged revisions 263374 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010) + | 8 lines Update link to new version of core sounds. The latest + version of the core sounds files 1.4.19 now includes the missing + queue-minute sound file which is called by app_queue but which + has been missing. (closes issue #17123) Reported by: n8ideas + ........ + +2010-05-17 13:05 +0000 [r263294] David Vossel + + * CHANGES: Update CHANGES to reflect DAHDI buffer dialstring option + backport to 1.6.2 + +2010-05-16 16:31 +0000 [r263250] Tzafrir Cohen + + * contrib/scripts/live_ast: live_ast: add commands 'rsync' and + 'gen-live-asterisk' This adds the following two commands to + live_ast: * rsync [user]@host directory Copy over all generated + files to at remote host. Would allow running live_ast + there. Hence allows separating a build machine from a test + machine. * gen-live-asteris: regenerate live/asterisk . Useful if + copying over files to a different directory. + +2010-05-16 11:14 +0000 [r263208] Kevin P. Fleming + + * main/astobj2.c: Improve some very confusing structure names in + astobj2.c As pointed out by 'akshayb' on #asterisk-dev, the code + here called a list of bucket entries a 'bucket', and the entries + within the bucket were called 'bucket_list'. This made the code + very hard to understand without reading all of it... so I've + renamed 'bucket_list' to 'bucket_entry' to clarify the purpose of + the structure. + +2010-05-14 18:53 +0000 [r263151] David Vossel + + * channels/chan_iax2.c: fix iax_frame double free Very unfortunate + things happen if we add an iax_frame to the frame queue and let + go of the lock before scheduling the frame's transmit... There is + a race condition that exists where the frame can be removed from + the frame_queue and freed before the transmit is scheduled if we + do not hold on to that lock. This results in a freed frame being + scheduled for transmit later. + +2010-05-13 22:01 +0000 [r263069] Richard Mudgett + + * channels/chan_dahdi.c: Fix inverted logic in cli command: ss7 set + debug on/off + +2010-05-13 20:25 +0000 [r263028] Tzafrir Cohen + + * configure, configure.ac: Remove "untested" feature PRI_VERSION + Nobody seems to actually test PRI_VERSION. It is only useful for + failing PRI support in chan_dahdi. + +2010-05-13 17:49 +0000 [r262940-262987] Tilghman Lesher + + * res/res_timing_kqueue.c: For FreeBSD + + * res/res_timing_kqueue.c: Hmmm, probably should have read the + manpage more thoroughly. + +2010-05-13 15:36 +0000 [r262895-262897] Russell Bryant + + * channels/chan_console.c: Fix an off by one error that causes a + crash. Thanks to Raymond Burke for pointing it out. + + * main/stdtime/localtime.c: Fix build on linux. + + * pbx/pbx_spool.c: Fix build on linux. + +2010-05-13 05:37 +0000 [r262852] Tilghman Lesher + + * Makefile, pbx/pbx_spool.c, tests/test_time.c, + build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, + main/stdtime/localtime.c, res/res_timing_kqueue.c (added): Add + kqueue(2) implementation to Asterisk in various places. This will + save a considerable amount of CPU on the BSDs, including Mac OS + X, as it eliminates several places in the code that we previously + used a busy loop. Additionally, this adds a res_timing interface, + using kqueue timers. Review: + https://reviewboard.asterisk.org/r/543/ + +2010-05-12 19:59 +0000 [r262800] Paul Belanger + + * main/loader.c, main/cli.c: Notify CLI when modules is loaded / + unloaded (closes issue #17308) Reported by: pabelanger Patches: + cli.modules.patch uploaded by pabelanger (license 224) Tested by: + pabelanger, russell + +2010-05-12 19:53 +0000 [r262796-262798] Leif Madsen + + * res/ael/pval.c: Revert previous WARNING message removal. + Marquis42 suggested a better method of doing what I wanted + because I ended up removing the WARNING message for all instances + when really I just wanted to remove it for the 'return' keyword, + not everything. (issue #17145) + + * res/ael/pval.c: Remove unnecessary WARNING message in ael/pval.c + (closes issue #17145) Reported by: okrief + +2010-05-12 18:01 +0000 [r262744] David Vossel + + * /, apps/app_meetme.c: Merged revisions 262662 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) + | 11 lines fixes app_meetme dsp error We attempted to detect + silence after translating a frame from signed linear. This caused + a flooding of errors. To resolve this the code to detect silence + was moved before the translation. (closes issue #17133) Reported + by: jsdyer ........ + +2010-05-12 17:57 +0000 [r262661-262743] Richard Mudgett + + * channels/chan_dahdi.c: Don't crash when destroying chan_dahdi + pseudo channels. Must do a deep copy of the cc_params in + duplicate_pseudo(). Otherwise, when the duplicate pseudo channel + is destroyed, it frees the original pseudo channel cc_params. The + original pseudo channel is then left with a dangling pointer for + when the next duplicated pseudo channel is created. + + * channels/chan_misdn.c: Merged revisions 262657,262660 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed, + 12 May 2010) | 4 lines Forgot some conditionals around the + callrerouting facility help text. JIRA ABE-2223 .......... + r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010) + | 22 lines Add mISDN Call rerouting facility for point-to-point + ISDN lines (exchange line) In the case of ISDN + point-to-multipoint (multidevice) you can use the mISDN "facility + calldeflect" application for call diversions from external (PSTN) + to external (PSTN). In that case this is the only way to get rid + of the two call legs to the PBX and let the calling number at the + C party become the number of the A party. In the case of ISDN + point-to-point (exchange line) the call deflection facility may + not be used. Instead a call rerouting facility has to be used. + This patch for chan_misdn.c is an extension to realize this + service (facility rerouting application). It can accept either + spelling: "callrerouting" or "callrerouteing". The patch is + tested towards Deutsche Telekom and requires a modified version + of mISDN from Digium, Inc. Patches: + misdn_rerouteing_corrected.patch (Slightly modified.) JIRA + ABE-2223 + +2010-05-12 16:23 +0000 [r262656] Tilghman Lesher + + * apps/app_privacy.c: Ensure the arguments are initialized. Also + miscellaneous CG cleanup. (closes issue #16576) Reported by: + uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman + (license 14) Tested by: uxbod + +2010-05-12 01:00 +0000 [r262613] Paul Belanger + + * channels/chan_sip.c, include/asterisk/cli.h: Convert to + AST_CLI_YESNO and AST_CLI_ONOFF Clean up chan_sip.c to use new + AST_CLI functions (closes issue #17287) Reported by: pabelanger + Patches: issue17287.patch uploaded by pabelanger (license 224) + Tested by: russell + +2010-05-11 23:18 +0000 [r262569] Richard Mudgett + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + channels/sig_pri.c: Dialing an invalid extension causes + incomplete hangup sequence. Revision -r1489 of the libpri 1.4 + branch corrected a deviation from Q.931 Section 5.3.2. However, + this resulted in an unexpected behaviour change to the upper + layer (Asterisk). This change uses pri_hangup_fix_enable() to + follow Q.931 Section 5.3.2 call hangup better if the version of + libpri supports it. (issue #17104) Reported by: shawkris Tested + by: rmudgett + +2010-05-11 21:25 +0000 [r262513] Tilghman Lesher + + * include/asterisk/causes.h: Move cause 200 to cause 26, as + specified in Q.850. Also cleanup the formatting and add a few + more that seem like good candidates. (closes issue #16157) + Reported by: wimpy + +2010-05-11 19:57 +0000 [r262422] Jason Parker + + * /, res/Makefile: Merged revisions 262421 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | + 11 lines Use a less silly method for modifying a flex-generated + file. The sed syntax that was used wasn't actually valid, causing + some versions to choke. This is the method that is used in 1.6.x+ + for similar changes. (closes issue #16696) Reported by: bklang + Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested + by: qwell ........ + +2010-05-11 19:40 +0000 [r262414-262419] Paul Belanger + + * pbx/pbx_config.c: Improve logging by displaying line number + (closes issue #16303) Reported by: dant Patches: + issue16303.patch.v2 uploaded by pabelanger (license 224) Tested + by: dant, lmadsen, pabelanger + + * channels/chan_sip.c: Improve logging information for + misconfigured contexts (closes issue #17238) Reported by: + pprindeville Patches: chan_sip-bug17238.patch uploaded by + pprindeville (license 347) Tested by: pprindeville + +2010-05-11 17:23 +0000 [r262330] Tilghman Lesher + + * /, Makefile.rules, apps/app_voicemail.c: Merged revisions 262321 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010) + | 2 lines Fix issue #17302 a slightly different way (mad props to + Qwell) ........ + +2010-05-11 16:43 +0000 [r262299] Jason Parker + + * bootstrap.sh: Allow bootstrap script to work on Solaris. As + usual, the way they do things is different, so we need to account + for that. automake is versioned ala BSD/Linux, but autoconf is + not. We don't actually need to specify a version there, since + AC_PREREQ will cover it for us. Things will fail pretty loudly if + AC_PREREQ isn't met. (closes issue #16341) Reported by: bklang + Patches: opensolaris_bootstrap.sh uploaded by bklang (license + 919) + +2010-05-10 19:06 +0000 [r262236-262240] David Vossel + + * apps/app_directed_pickup.c: fixes PickupChan application (closes + issue #16863) Reported by: schern Patches: + app_directed_pickup.c.patch uploaded by schern (license 995) + for_trunk.diff uploaded by cjacobsen (license 1029) Tested by: + Graber, cjacobsen, lathama, rickead2000, dvossel + + * channels/chan_console.c: fixes crash in chan_console There is a + race condition between console_hangup() and start_stream(). It is + possible for console_hangup() to be called and then the stream + thread to begin after the hangup. To avoid this a check in + start_stream() to make sure the pvt-owner still exists while the + pvt lock is held is made. If the owner is gone that means the + channel hung up and start_stream should be aborted. + +2010-05-10 16:36 +0000 [r262152] Tilghman Lesher + + * /, Makefile.rules: Merged revisions 262151 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010) + | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes + issue #17297) Reported by: jcovert Patches: + 20100506__issue17297.diff.txt uploaded by tilghman (license 14) + (closes issue #17302) Reported by: jcovert ........ + +2010-05-09 02:14 +0000 [r262048-262102] Tilghman Lesher + + * autoconf/ast_c_define_check.m4, configure, + include/asterisk/autoconfig.h.in, autoconf/ast_ext_lib.m4, + autoconf/ast_c_compile_check.m4: Cleanup a bit more by getting + rid of useless version defines. Also make library detection use + passed CFLAGS. (closes issue #17309) Reported by: stuarth + + * configure, configure.ac: Use CPPFLAGS to pass PTHREAD_CFLAGS for + vpb only + +2010-05-07 23:54 +0000 [r262005] Alec L Davis + + * UPGRADE.txt, apps/app_voicemail.c: VoicemailMain and + VMauthenticate, allow escape to the 'a' extension when a single + '*' is entered Where a site uses VoicemailMain(mailbox) the users + have to be at their own extension to clear their voicemail, they + have no way of escaping VoicemailMain to allow entry of new + boxnumber. This patch, allows a site to include to 'a' priority + in the VoicemailMain context, to allow an escape. If the 'a' + priority doesn't exist in the context that VoicemailMain was + called from then it acts as the old behaviour. Reported by: + alecdavis Tested by: alecdavis Patch vm_a_extension.diff2.txt + uploaded by alecdavis (license 585) Review: + https://reviewboard.asterisk.org/r/489/ + +2010-05-07 22:09 +0000 [r261913-261964] Tilghman Lesher + + * addons/ooh323c/src/ooh323.c: Fix build on Linux + + * funcs/func_odbc.c: Double free crash (closes issue #17245) + Reported by: thedavidfactor Patches: + 20100426__issue17245.diff.txt uploaded by tilghman (license 14) + Tested by: murraytm + + * configure, include/asterisk/autoconfig.h.in, configure.ac: Use + the detected pthread building flags in every place, instead of + hardcoding -lpthread. We nicely detect the right flags on each + system for building Asterisk with pthreads, then ignore it for + every other build option that requires us to build with pthreads. + This caused some items to return a false negative. Also cleanup + some minor naming issues that caused "library library" redundancy + in the output. (closes issue #17303) Reported by: stuarth + Patches: 20100507__issue17303.diff.txt uploaded by tilghman + (license 14) Tested by: stuarth + +2010-05-07 16:05 +0000 [r261867] Leif Madsen + + * UPGRADE-1.6.txt: Update UPGRADE-1.6.txt stating insecure=very has + been removed. (closes issue #17282) Reported by: stuarth Tested + by: stuarth + +2010-05-07 15:33 +0000 [r261866] Jeff Peeler + + * channels/sig_pri.c: Fix deadlock in sig_pri when hanging up. The + pri_dchannel thread currently violates locking order by locking + the private and then attempting to queue a frame, which needs to + lock the channel. Queueing a frame is unneccesary though and is + actually a regression since sig_pri. All the places that + currently use ast_softhangup_nolock now will just set the + softhangup value directly as before. (closes issue #17216) + Reported by: lmsteffan Patches: bug17216.patch uploaded by + jpeeler (license 325) + +2010-05-06 23:41 +0000 [r261822] Richard Mudgett + + * channels/sig_pri.c: Some code optimizations. * Made more places + use pri_queue_control() instead of pri_queue_frame() and a local + frame variable. * Made pri_queue_frame() use + sig_pri_lock_owner(). pri_queue_frame() no longer releases the + libpri access lock unless it is required. * Made the + pri_queue_frame() and pri_queue_control() parameter list similar + to sig_pri_lock_owner(). + +2010-05-06 20:11 +0000 [r261736] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 261735 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 + May 2010) | 8 lines Only allow the operator key to be accepted + after leaving a voicemail. Or rather disallow the operator key + from being accepted when not offered, such as after finishing a + recording from within the mailbox options menu. ABE-2121 SWP-1267 + ........ + +2010-05-06 17:06 +0000 [r261609] Jason Parker + + * /, sounds/Makefile: Merged revisions 261608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) | + 4 lines Use the versioned MOH tarballs, now that we have them. + This makes for more reproducibility. Prompted by a discussion in + #asterisk-dev ........ + +2010-05-06 15:39 +0000 [r261560] Tilghman Lesher + + * channels/sip/include/sip.h: Permit more lines within a SIP body + to be parsed. The example given within the related issue showed + 120 lines, which was mostly a result of the body being XML. + (closes issue #17179) Reported by: khw + +2010-05-06 14:15 +0000 [r261496-261500] Russell Bryant + + * tests/test_heap.c: Add test case for removing random elements + from a heap. I modified the original patch for trunk to use the + unit test API. (issue #17277) Reported by: cappucinoking Patches: + test_heap.diff uploaded by cappucinoking (license 1036) Tested + by: cappucinoking, russell + + * main/heap.c: Fix handling of removing nodes from the middle of a + heap. This bug surfaced in 1.6.2 and does not affect code in any + other released version of Asterisk. It manifested itself as SIP + qualify not happening when it should, causing peers to go + unreachable. This was debugged down to scheduler entries + sometimes not getting executed when they were supposed to, which + was in turn caused by an error in the heap code. The problem only + sometimes occurs, and it is due to the logic for removing an + entry in the heap from an arbitrary location (not just popping + off the top). The scheduler performs this operation frequently + when entries are removed before they run (when ast_sched_del() is + used). In a normal pop off of the top of the heap, a node is + taken off the bottom, placed at the top, and then bubbled down + until the max heap property is restored (see max_heapify()). This + same logic was used for removing an arbitrary node from the + middle of the heap. Unfortunately, that logic is full of fail. + This patch fixes that by fully restoring the max heap property + when a node is thrown into the middle of the heap. Instead of + just pushing it down as appropriate, it first pushes it up as + high as it will go, and _then_ pushes it down. Lastly, fix a + minor problem in ast_heap_verify(), which is only used for + debugging. If a parent and child node have the same value, that + is not an error. The only error is if a parent's value is less + than its children. A huge thanks goes out to cappucinoking for + debugging this down to the scheduler, and then producing an + ast_heap test case that demonstrated the breakage. That made it + very easy for me to focus on the heap logic and produce a fix. + Open source projects are awesome. (closes issue #16936) Reported + by: ib2 Tested by: cappucinoking, crjw (closes issue #17277) + Reported by: cappucinoking Patches: heap-fix.rev2.diff uploaded + by russell (license 2) Tested by: cappucinoking, russell + +2010-05-06 07:27 +0000 [r261451] Tzafrir Cohen + + * channels/chan_dahdi.c: When failing to configure, don't destroy + 'cfg' twice Fixes a crash when some config section had an + incorrect channel config. + +2010-05-05 22:22 +0000 [r261405] Richard Mudgett + + * channels/chan_dahdi.c: Avoid a crash on SS7 channels. + +2010-05-05 20:48 +0000 [r261364] Russell Bryant + + * Makefile, configs/asterisk.conf.sample: Restore previous + asterisk.conf syntax, where the directories aren't commented out. + This fixes some breakage in the test suite, that uses the + contents of asterisk.conf to discover the install layout on the + system. + +2010-05-05 19:13 +0000 [r261316] David Vossel + + * channels/chan_sip.c: fixes sip native transfer The Refer-To + header field containing the Replaces header in the URI was not + being decoded properly. This caused invalid parsing between the + caller id field and the domain resulting in a failed transfer. + (closes issue #17284) Reported by: dvossel + +2010-05-05 18:43 +0000 [r261314] Paul Belanger + + * /, channels/chan_sip.c: Merged revisions 261274 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May + 2010) | 12 lines Registration fix for SIP realtime. Make sure + realtime fields are not empty. (closes issue #17266) Reported by: + Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick + Lewis (license 657) Tested by: Nick_Lewis, sberney Review: + https://reviewboard.asterisk.org/r/643/ ........ + +2010-05-05 18:28 +0000 [r261313] Mark Michelson + + * channels/sip/dialplan_functions.c: Prevent unnecessary warnings + when getting rtpsource or rtpdest. If a recognized media type was + present, but the media type was not enabled for the channel, then + a warning would be emitted. For instance, attempting to get + CHANNEL(rtpsource,video) on a call with no video would cause a + warning message to appear. With this change, the warning will + only appear if the stream argument is not recognized as being a + media type that can be specified. + +2010-05-05 15:42 +0000 [r261124-261232] Paul Belanger + + * apps/app_queue.c: 'queue reset stats' erroneously clears + wrapuptime configuration. Resets each member's lastcall to 0 now. + (closes issue #17262) Reported by: rain Patches: + wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested + by: rain + + * main/manager.c, include/asterisk/cli.h, CHANGES, + include/asterisk/manager.h: New 'manager show settings' CLI + command. See the CHANGES file for more details. (closes issue + #16343) Reported by: pabelanger Patches: issue16343.patch.v5 + uploaded by pabelanger (license 224) Tested by: pabelanger, + tilghman, lmadsen Review: https://reviewboard.asterisk.org/r/630/ + + * Makefile, configs/asterisk.conf.sample (added): New static + asterisk.conf.sample file. This simply moves the functionality + from the Makefile (cleaning it up) into an external + asterisk.conf.samples file. Also updates formatting (easier to + read) and grammar changes to asterisk.conf.samples. (closes issue + #17027) Reported by: pabelanger Patches: + 0017027.asterisk.conf.v6.patch uploaded by pabelanger (license + 224) Tested by: qwell, lmadsen, pabelanger, chappell Review: + https://reviewboard.asterisk.org/r/616/ + +2010-05-04 23:51 +0000 [r261095] Tilghman Lesher + + * main/channel.c, /: Merged revisions 261093-261094 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 + May 2010) | 7 lines Protect against overflow, when calculating + how long to wait for a frame. (closes issue #17128) Reported by: + under Patches: d.diff uploaded by under (license 914) ........ + r261094 | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) + | 2 lines Add a tiny corner case to the previous commit ........ + +2010-05-04 22:46 +0000 [r261051] Mark Michelson + + * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add new + possible value to autopause option to allow members to be + autopaused in all queues. See the CHANGES file and + queues.conf.sample for more details. (closes issue #17008) + Reported by: jlpedrosa Patches: queues.autopause_en_review.diff + uploaded by jlpedrosa (license 1002) Review: + https://reviewboard.asterisk.org/r/581/ + +2010-05-04 21:10 +0000 [r261007] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h, channels/sig_pri.c: The inalarm flag is + not passed up from the sig_analog and sig_pri submodules. The CLI + "dahdi show channel" command was not correctly reporting the + InAlarm status. The inalarm flag is now consistently passed + between chan_dahdi and submodules. + +2010-05-04 18:51 +0000 [r260924] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 260923 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 + May 2010) | 12 lines Voicemail transfer to operator should occur + immediately, not after main menu. There were two scenarios in the + advanced options that while using the operator=yes and review=yes + options, the transfer occurred only after exiting the main menu + (after sending a reply or leaving a message for an extension). + Now after the audio is processed for the reply or message the + transfer occurs immediately as expected. ABE-2107 ABE-2108 + ........ + +2010-05-04 15:49 +0000 [r260802] Jason Parker + + * /, build_tools/make_build_h: Merged revisions 260801 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May + 2010) | 1 line Fix fallout from removing from configure script. + Pointed out by philipp64 on #asterisk-dev ........ + +2010-05-03 22:13 +0000 [r260757] Jeff Peeler + + * apps/app_meetme.c, CHANGES: Add new admin features to meetme: + Roll call, eject all, mute all, record in-conf This patch adds + the following in-conference admin DTMF features: *81 - Roll call + (or simply user count if INTROUSER isn't enabled) *82 - Eject all + non-admins *83 - Mute/unmute all non-admins *84 - Start recording + the conference on the fly FWIW, this code uses newly recorded + prompts. (closes issue #16379) Reported by: rfinnie Patches: + meetme-enhancements-232771-v1.patch uploaded by rfinnie (license + 940) modified slightly by me + +2010-05-03 17:06 +0000 [r260663] Paul Belanger + + * Makefile, /: Merged revisions 260661-260662 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May + 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend + libdir when executing mkpkgconfig allowing non-root installs to + work. (closes issue #17268) Reported by: pabelanger Patches: + issue17268.patch uploaded by pabelanger (license 224) Tested by: + pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41 + -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/ + part. Thanks Qwell. ........ + +2010-05-03 14:58 +0000 [r260570] Leif Madsen + + * doc/HOWTO_collect_debug_information.txt: Merged revisions 260569 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010) + | 1 line Minor typo pointed out by pabelanger on IRC. ........ + +2010-05-02 02:52 +0000 [r260521] Eliel C. Sardanons + + * main/data.c, include/asterisk/data.h: Avoid making AstData depend + on libxml2 to compile. We have some functions inside the AstData + API to get the tree in XML form, but it is not required at the + moment to compile asterisk and we can disable that part of the + API if we don't have libxml2 support. + +2010-04-30 22:36 +0000 [r260437] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Merged revisions 260434 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) + | 11 lines Ensure channel state is not incorrectly set in the + case of a very early answer. The needringing bit was being read + in dahdi_read after answering thereby setting the state to + ringing from up. This clears needringing upon answering so that + is no longer possible. (closes issue #17067) Reported by: tzafrir + Patches: needringing.diff uploaded by tzafrir (license 46) + ........ + +2010-04-30 22:24 +0000 [r260435] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7, + and MFCR2 users. Created SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS + SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS SIG_MFCR2_MAX_CHANNELS + Also fixed the declaration of pollers[] in mfcr2_monitor(). It + was dimensioned to the number of bytes in struct + dahdi_mfcr2.pvts[] and not to the same dimension of the struct + dahdi_mfcr2.pvts[]. + +2010-04-30 20:11 +0000 [r260344-260346] Mark Michelson + + * /, res/res_musiconhold.c: Merged revisions 260345 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, + 30 Apr 2010) | 18 lines Fix potential crash from race condition + due to accessing channel data without the channel locked. In + res_musiconhold.c, there are several places where a channel's + stream's existence is checked prior to calling ast_closestream on + it. The issue here is that in several cases, the channel was not + locked while checking the stream. The result was that if two + threads checked the state of the channel's stream at + approximately the same time, then there could be a situation + where both threads attempt to call ast_closestream on the + channel's stream. The result here is that the refcount for the + stream would go below 0, resulting in a crash. I have added + proper channel locking to res_musiconhold.c to ensure that we do + not try to check chan->stream without the channel locked. A + Digium customer has been using this patch for several weeks and + has not had any crashes since applying the patch. ABE-2147 + ........ + + * apps/app_queue.c: Fix logic reversal error when queue callers + join the queue. When a specific position is specified for the + queue, the idea was that the caller cannot be placed ahead of + higher-priority callers. Unfortunately, the logic was reversed so + that the caller could ONLY be placed ahead of higher priority + callers. Discovered while writing a unit test. + +2010-04-30 06:19 +0000 [r260280-260292] Tilghman Lesher + + * main/strcompat.c: Don't allow file descriptors to go above 64k, + when we're closing them in a fork(2). This saves time, when, even + though the system allows the process limit to be that high, the + practical limit is much lower. Also introduce an additional + optimization, in the form of using the CLOEXEC flag to close + descriptors at the right time. (closes issue #17223) Reported by: + dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by + tilghman (license 14) Tested by: dbackeberg + + * configs/extensions.conf.sample: Logic fixups for a sample FREENUM + dialplan context. (closes issue #17263) Reported by: pprindeville + Patches: freenum-dialplan.patch#3 uploaded by pprindeville + (license 347) + +2010-04-29 22:44 +0000 [r260231] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 260195 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) + | 26 lines DTMF CallerID detection problems. The code handling + DTMF CallerID drops digits on long CallerID numbers and may + timeout waiting for the first ring with shorter numbers. The DTMF + emulation mode was not turned off when processing DTMF CallerID. + When the emulation code gets behind in processing the DTMF digits + it can skip a digit. For shorter numbers, the timeout may have + been too short. I increased it from 2 seconds to 4 seconds. Four + seconds is a typical time between rings for many countries. + (closes issue #16460) Reported by: sum Patches: issue16460.patch + uploaded by rmudgett (license 664) issue16460_v1.6.2.patch + uploaded by rmudgett (license 664) Tested by: sum, rmudgett + Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA + AST-334 JIRA SWP-901 ........ + +2010-04-29 18:15 +0000 [r260148] Tilghman Lesher + + * configs/extensions.conf.sample: Pattern match fail. + +2010-04-29 15:33 +0000 [r260050] David Vossel + + * /, include/asterisk/audiohook.h, main/audiohook.c: Merged + revisions 260049 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) + | 14 lines Fixes crash in audiohook_write_list The middle_frame + in the audiohook_write_list function was being freed if a + audiohook manipulator returned a failure. This is incorrect + logic. This patch resolves this and adds detailed descriptions of + how this function should work and why manipulator failures must + be ignored. (closes issue #17052) Reported by: dvossel Tested by: + dvossel (closes issue #16196) Reported by: atis Review: + https://reviewboard.asterisk.org/r/623/ ........ + +2010-04-29 00:35 +0000 [r260007] Richard Mudgett + + * include/asterisk/extconf.h: Fix comment. + +2010-04-28 22:34 +0000 [r259957] Mark Michelson + + * channels/chan_sip.c, channels/sip/include/sip.h: Don't override + peer context with domain context. (closes issue #17040) Reported + by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded + by pprindeville (license 347) Tested by: pprindeville Review: + https://reviewboard.asterisk.org/r/565/ + +2010-04-28 21:20 +0000 [r259870] David Vossel + + * main/channel.c, channels/chan_local.c, /: Merged revisions 259858 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) + | 33 lines resolves deadlocks in chan_local Issue_1. In the + local_hangup() 3 locks must be held at the same time... pvt, + pvt->chan, and pvt->owner. Proper deadlock avoidance is done when + the channel to hangup is the outbound chan_local channel, but + when it is not the outbound channel we have an issue... We + attempt to do deadlock avoidance only on the tech pvt, when both + the tech pvt and the pvt->owner are locked coming into that loop. + By never giving up the pvt->owner channel deadlock avoidance is + not entirely possible. This patch resolves that by doing deadlock + avoidance on both the pvt->owner and the pvt when trying to get + the pvt->chan lock. Issue_2. ast_prod() is used in + ast_activate_generator() to queue a frame on the channel and make + the channel's read function get called. This function is used in + ast_activate_generator() while the channel is locked, which + mean's the channel will have a lock both from the generator code + and the frame_queue code by the time it gets to chan_local.c's + local_queue_frame code... local_queue_frame contains some of the + same crazy deadlock avoidance that local_hangup requires, and + this recursive lock prevents that deadlock avoidance from + happening correctly. This patch removes ast_prod() from the + channel lock so only one lock is held during the + local_queue_frame function. (closes issue #17185) Reported by: + schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel + (license 671) issue_17185_v2.diff uploaded by dvossel (license + 671) Tested by: schmoozecom, GameGamer43 Review: + https://reviewboard.asterisk.org/r/631/ ........ + +2010-04-28 21:08 +0000 [r259853] Leif Madsen + + * /, config.guess: Merged revisions 259852 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010) + | 6 lines Update config.guess. Updating config.guess because + after installing Ubuntu Server 9.10 and running all the update + scripts, running ./configure would not continue because it was + unable to determine what kind of system I had. After updating + config.guess things started working again. ........ + +2010-04-28 20:32 +0000 [r259760-259848] Jason Parker + + * /, configure, configure.ac: Merged revisions 259847 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr + 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so + systems without install can use install-sh from our source dir. + ........ + + * /, makeopts.in: Merged revisions 259833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) | + 1 line Missed this when removing $ID ........ + + * Makefile, /, configure, configure.ac: Merged revisions 259748 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) | + 7 lines Remove usage of `id` since it isn't useful and was + causing breakge. Solaris `id` doesn't support the -u argument. + Instead of figuring out how to fix this to work on Solaris, I + decided to check why it was necessary and where else it was used. + It was only used in one place, and it hasn't been needed for a + very long time (I question whether it was ever needed). ........ + +2010-04-28 17:18 +0000 [r259672] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 259664 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 + Apr 2010) | 4 lines Do not play goodbye prompt after timeout of + message review. ABE-2124 ........ + +2010-04-27 22:47 +0000 [r259587-259617] Jason Parker + + * res/res_agi.c: Fix compile on systems without + HAVE_NULLSAFE_PRINTF defined. + + * channels/sip/dialplan_functions.c: Be more explicit about field + naming in a test. + +2010-04-27 22:18 +0000 [r259538] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 259531 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 + Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and + vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed + failed: Success" Changed the warning to "Failed to decode + CallerID on channel 'name'". The message before it is likely more + specific about why the CallerID decode failed. SWP-501 AST-283 + ........ + +2010-04-27 22:11 +0000 [r259533] Mark Michelson + + * main/ccss.c: Shuffle some casts to make builds on bamboo happier. + +2010-04-27 21:49 +0000 [r259527] Leif Madsen + + * /, sounds/Makefile: Merged revisions 259526 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010) + | 15 lines Update sounds files. * Add additional sounds prompts + for say_enumeration * Update the English conference sounds + prompts so they are better quality and all sound more consistent + * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files + to include all present sound files Both core (en, fr, es) and + extra (en, fr) sounds files have been updated. (closes issue + #16200) Reported by: murf (closes issue #17137) Reported by: + lmadsen ........ + +2010-04-27 21:18 +0000 [r259439-259451] Jason Parker + + * /: Block 259441 instead of recording it as merged. + + * /: Recorded merge of revisions 259441 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259441 | qwell | 2010-04-27 16:15:46 -0500 (Tue, 27 Apr 2010) | + 1 line Add gar to the check for AR for those silly OSes (Solaris) + that don't have ar. ........ + + * main/editline/configure, main/editline/Makefile.in, + main/editline/configure.in: Add gar to the check for AR for those + silly OSes (Solaris) that don't have ar. autoconf2.13 couldn't + handle AC_PROG_GREP, so I removed it. This is fine, since we + don't need to use anything that the configure script doesn't. + +2010-04-27 21:10 +0000 [r259438] Leif Madsen + + * include/asterisk/doxygen/mantisworkflow.h: Update the Mantis + Workflow document in doxygen. (closes issue #17175) Reported by: + lmadsen Patches: Bug_Tracker_Workflow.v2.txt uploaded by + pabelanger (license 224) Tested by: pabelanger, lmadsen + +2010-04-27 19:52 +0000 [r259357] Mark Michelson + + * main/ccss.c: Change cc_ref and cc_unref from macros to inline + functions. The hope is that Solaris won't be as whiny after this + change. + +2010-04-27 19:31 +0000 [r259353] Jason Parker + + * /, configure, configure.ac: Merged revisions 259352 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr + 2010) | 5 lines Support the silly OSes that don't have ar and + strip. Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path + isn't specified, and AC_PATH_TOOLS doesn't exist, we'll just + switch to AC_CHECK_TOOLS. ........ + +2010-04-27 18:29 +0000 [r259229-259307] Richard Mudgett + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged + revisions 259270 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) + | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue + #7321 implements a new chan_dahdi configuration option. However, + a change mentioned in the issue was never implemented. This is + the change that will allow the feature to work. I added a note to + chan_dahdi.conf.sample about the feature. (closes issue #17143) + Reported by: djensen99 Patches: diff.txt uploaded by djensen99 + (license NA) (One line change) Tested by: djensen99 ........ + + * channels/chan_dahdi.c: Re-fix dahdi_request() iflist locking + since CCSS merged. + +2010-04-27 15:25 +0000 [r259189] Tilghman Lesher + + * contrib/init.d/etc_default_asterisk (added): Add missing file + (pointed out by TheDavidFactor on #asterisk-dev) referenced by + revision 239231. + +2010-04-26 21:45 +0000 [r259023-259105] Mark Michelson + + * main/channel.c, /: Merged revisions 259104 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr + 2010) | 3 lines Let compilation succeed warning-free when + DONT_OPTIMIZE is turned off. ........ + + * main/channel.c, /: Merged revisions 259018 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr + 2010) | 13 lines Prevent Newchannel manager events for dummy + channels. No Newchannel manager event will be fired for channels + that are allocated to not match a registered technology type. + Thus bogus channels allocated solely for variable substitution or + CDR operations do not result in a Newchannel event. (closes issue + #16957) Reported by: atis Review: + https://reviewboard.asterisk.org/r/601 ........ + +2010-04-26 19:05 +0000 [r258974] David Ruggles + + * contrib/valgrind.supp: Line 24 missed in compatibility fix in + revision 233577 added a "fun:" prefix line 24 + +2010-04-26 15:59 +0000 [r258934] Leif Madsen + + * channels/chan_sip.c: Small error in the T.140 RTP port verbose + log. (closes issue #16988) Reported by: frawd Patches: + chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610) + Tested by: russell + +2010-04-26 14:18 +0000 [r258896] Matthew Nicholson + + * res/res_fax.c, include/asterisk/res_fax.h, res/res_fax_spandsp.c: + Update res_fax and res_fax_spandsp to be compatible with Fax For + Asterisk 1.2. The fax session initilization code for T.38 faxes + has been rewritten. T.38 session initialization was removed from + generic_fax_exec, and split into two different code paths for + receive and send. Also the 'z' option (to send a T.38 reinvite if + we do not receive one) was added to sendfax. In the output of + 'fax show sessions', the 'Type' column has been renamed to 'Tech' + and replaced with a new 'Tech' column that will report 'G.711' or + 'T.38'. Control of ECM defaults has been added to res_fax A 'fax + show settings' CLI command has been added. Support of the new + AST_T38_REQUEST_PARMS control method request to handle channels + that have already received a T.38 reinvite before the FAX + application is start has been added. Support for the 'fax show + settings' command has been added to res_fax_spandsp and handling + of the ECM flag has been slightly altered. + +2010-04-25 18:51 +0000 [r258838-258855] Alexandr Anikin + + * addons/chan_ooh323.c: additional checking related to issue 17186 + + * addons/chan_ooh323.c: Don't pass zero length callerid to ooh323 + stack Don't pass zero callerid string to ooh323 stack because it + can't encode this properly and can't generate setup message. + (closes issue #17186) Reported by: vmikhelson Patches: + zero_callerid_num.patch uploaded by may213 (license 454) Tested + by: may213 + +2010-04-25 18:12 +0000 [r258776] Tilghman Lesher + + * /, res/res_monitor.c: Merged revisions 258775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) + | 6 lines When StopMonitor is called, ensure that it will not be + restarted by a channel event. (closes issue #16590) Reported by: + kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm + (license 888) ........ + +2010-04-22 22:19 +0000 [r258685] Jason Parker + + * utils/extconf.c: Add another random function that does nothing to + make the utils/ dir happy. + +2010-04-22 22:11 +0000 [r258675] Matthew Nicholson + + * main/channel.c: Fix previous commit. + +2010-04-22 22:10 +0000 [r258673-258674] Jason Parker + + * utils/Makefile, utils/extconf.c: Make utils/ stuff *actually* + compile this time. + + * utils/Makefile, utils/extconf.c: Let utils/ dir compile when + DEBUG_THREADS is not enabled. + +2010-04-22 21:57 +0000 [r258671] Matthew Nicholson + + * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions + 193391,258670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May + 2009) | 8 lines Set the proper disposition on originated calls. + (closes issue #14167) Reported by: jpt Patches: + call-file-missing-cdr2.diff uploaded by mnicholson (license 96) + Tested by: dlotina, rmartinez, mnicholson ........ r258670 | + mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11 + lines Fix broken CDR behavior. This change allows a CDR record + previously marked with disposition ANSWERED to be set as BUSY or + NO ANSWER. Additionally this change partially reverts r235635 and + does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated + from ast_call(). To preserve proper CDR behavior, the + AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in + ast_bridge_call(). (closes issue #16797) Reported by: + VarnishedOtter Tested by: mnicholson ........ (closes issue + #16222) Reported by: telles Tested by: mnicholson + +2010-04-22 21:06 +0000 [r258632] Russell Bryant + + * tests/test_event.c, main/event.c: Add ast_event subscription unit + test and fix some ast_event API bugs. This patch introduces + another test in test_event.c that exercises most of the + subscription related ast_event API calls. I made some minor + additions to the existing event allocation test to increase API + coverage by the test code. Finally, I made a list in a comment of + API calls not yet touched by the test module as a to-do list for + future test development. During the development of this test + code, I discovered a number of bugs in the event API. 1) + subscriptions to AST_EVENT_ALL were not handled appropriately in + a couple of different places. The API allows a subscription to + all event types, but with IE parameters, just as if it was a + subscription to a specific event type. However, the parameters + were being ignored. This affected ast_event_check_subscriber() + and event distribution to subscribers. 2) Some of the logic in + ast_event_check_subscriber() for checking subscriptions against + query parameters was wrong. Review: + https://reviewboard.asterisk.org/r/617/ + +2010-04-22 20:04 +0000 [r258595] Eliel C. Sardanons + + * apps/app_voicemail.c: Pass interactive = 0 and fix a compile + error. + +2010-04-22 19:08 +0000 [r258557] Jason Parker + + * main/lock.c (added), include/asterisk/res_odbc.h, + include/asterisk/astobj2.h, main/heap.c, include/asterisk/lock.h, + main/astobj2.c, res/res_odbc.c, include/asterisk/heap.h: Remove + ABI differences that occured when compiling with DEBUG_THREADS. + "Bad Things" would happen if Asterisk was compiled with + DEBUG_THREADS, but a loaded module was not (or vice versa). This + also immensely simplifies the lock code, since there are no + longer 2 separate versions of them. Review: + https://reviewboard.asterisk.org/r/508/ + +2010-04-22 18:07 +0000 [r258517] Eliel C. Sardanons + + * doc/manager_1_1.txt, main/channel.c, include/asterisk/doxyref.h, + include/asterisk/xml.h, main/data.c (added), main/xml.c, + include/asterisk/channel.h, include/asterisk/_private.h, + include/asterisk/data.h (added), CHANGES, apps/app_queue.c, + main/asterisk.c, apps/app_voicemail.c: Asterisk data retrieval + API. This module implements an abstraction for retrieving and + exporting asterisk data. Developed by: Brett Bryant + Eliel C. Sardanons (LU1ALY) + For the Google Summer of code 2009 Project. + Documentation can be found in doxygen format and inside the + header include/asterisk/data.h Review: + https://reviewboard.asterisk.org/r/275/ + +2010-04-22 17:36 +0000 [r258515] Russell Bryant + + * doc/tex/channelvariables.tex: Add MEETMEBOOKID from r256019. + +2010-04-21 21:56 +0000 [r258433] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 258432 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 + Apr 2010) | 8 lines Fix looping forever when no input received in + certain voicemail menu scenarios. Specifically, prompting for an + extension (when leaving or forwarding a message) or when + prompting for a digit (when saving a message or changing + folders). ABE-2122 SWP-1268 ........ + +2010-04-21 19:45 +0000 [r258351-258387] Leif Madsen + + * doc/tex/asterisk.tex: Missed this when reverting the bad version + change in asterisk.tex. + + * doc/tex/asterisk.tex: Fix change in asterisk.tex that got merged + in after testing. (issue #17220) + + * Makefile, doc/tex/security-events.tex, configure, + include/asterisk/autoconfig.h.in, doc/tex/Makefile, configure.ac, + doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex, + build_tools/prep_tarball, doc/tex/localchannel.tex, + doc/tex/enum.tex, makeopts.in, doc/tex/asterisk.tex, + doc/tex/cel-doc.tex: Add ability to generate ASCII documentation + from the TeX files. These changes add the ability to run 'make + asterisk.txt' just like the existing 'make asterisk.pdf' commands + to generate a text document from the TeX files we have in the + doc/tex/ directory. I've also updated a few of the .tex files + because they weren't properly escaping certain characters so they + would show up as Unicode characters (like [U+021C]). Made changes + to the configure scripts so it would detect the catdvi program + which is required to convert the .dvi file generated by latex. + I've also added a few lines to the build_tools/prep_tarball + script so that the text documentation gets generated and added to + future tarballs of Asterisk releases. (closes issue #17220) + Reported by: lmadsen Patches: asterisk.txt.patch uploaded by + lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger + (license 224) Tested by: lmadsen, pabelanger + +2010-04-21 19:07 +0000 [r258345] Mark Michelson + + * funcs/func_callcompletion.c: Add small documentation update to + func_callcompletion.c. This directs users to documents which can + help explain the concepts and configuration options settable with + the function. + +2010-04-21 19:02 +0000 [r258344] Leif Madsen + + * UPGRADE.txt, CHANGES, channels/chan_iax2.c: IAXpeers output now + matches SIPpeers format for manager (AMI). (closes issue #17100) + Reported by: secesh Tested by: pabelanger Review: + https://reviewboard.asterisk.org/r/594/ + +2010-04-21 18:13 +0000 [r258305] David Vossel + + * channels/chan_sip.c: fixes issue with double "sip:" in header + field This is a clear mistake in logic. Future discussions about + how to avoid having to handle uri's like this should take place + in the future, but this fix needs to go in for now. (closes issue + #15847) Reported by: ebroad Patches: doublesip.patch uploaded by + ebroad (license 878) + +2010-04-21 13:26 +0000 [r258265] Leif Madsen + + * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, + res/res_calendar_caldav.c: Fix the \brief description in the + res_calendar_*.c files. + +2010-04-21 13:24 +0000 [r258190-258256] Julian Lyndon-Smith + + * doc/manager_1_1.txt: fix whitespace issue + + * doc/manager_1_1.txt, doc/tex/manager.tex: Added NEW ACTIONS entry + for new MixMonitorMute AMI command. Added State and Direction + variables for new MixMonitorMute AMI command. + + * CHANGES: Added CHANGES entry for new MixMonitorMute AMI command. + + * main/frame.c, include/asterisk/audiohook.h, main/audiohook.c, + include/asterisk/frame.h, apps/app_mixmonitor.c, + res/res_mutestream.c: Added MixMonitorMute manager command Added + a new manager command to mute/unmute MixMonitor audio on a + channel. Added a new feature to audiohooks so that you can mute + either read / write (or both) types of frames - this allows for + MixMonitor to mute either side of the conversation without + affecting the conversation itself. (closes issue #16740) Reported + by: jmls Review: https://reviewboard.asterisk.org/r/487/ + +2010-04-20 19:02 +0000 [r258106-258149] Leif Madsen + + * configs/cli_aliases.conf.sample: Add 'soft hangup' alias per + Steve Johnson on asterisk-users. + + * configs/extensions.conf.sample: Add example dialplan for dialing + ISN numbers (http://www.freenum.org). Minor tweaks and + documentation added by me. (closes issue #17058) Reported by: + pprindeville Patches: freenum.patch#5 uploaded by pprindeville + (license 347) Tested by: lmadsen + + * contrib/scripts/sip-friends.sql: Add missing 'useragent' field to + sip-friends.sql file. (closes issue #17171) Reported by: thehar + Patches: sip-friends.patch uploaded by thehar (license 831) + Tested by: pabelanger, thehar + +2010-04-20 17:06 +0000 [r258065] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 258029 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 + Apr 2010) | 11 lines Play correct prompt when voicemail store + failure occurs after attempted forward. If a user's mailbox was + full and a message was attempted to be forwarded to said box, + warnings on the console would indicate failure. However, the + played prompt was that of success (vm-msgsaved). Now storage + failure is taken into account and the correct prompt + (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262 + ........ + +2010-04-20 12:38 +0000 [r257988] Leif Madsen + + * formats/format_pcm.c: Update supported file extensions in + doxygen. Updated the doxygen \arg line after looking at the file + for some other Asterisk documentation and noticing they weren't + up to date. Thanks to seanbright for looking at the code for me + :) + +2010-04-19 21:57 +0000 [r257947-257949] Jason Parker + + * main/indications.c: Change log message to match severity. + + * main/indications.c: Don't consider a missing indications.conf to + be a critical error. There were many changes in revision 176627 + which would avoid the error that a missing config would have + caused. Other than this, there are no other config files + (including asterisk.conf, surprisingly) that are required. + +2010-04-19 19:23 +0000 [r257883] Tilghman Lesher + + * apps/app_voicemail.c: Bad merge fix + +2010-04-19 18:42 +0000 [r257851] Mark Michelson + + * funcs/func_srv.c: Commit compromise I suggested on review 608. + This allows for multiple SRV queries to be done from the dialplan + for the same service on a single call while still allowing one to + bypass the call to SRVQUERY if they so please. Taking action + since no comments had been left for a while. This can easily be + reverted if needed. External tests still pass. + +2010-04-19 17:57 +0000 [r257810] Terry Wilson + + * main/features.c: Fix incomplete CDR merge from r195881 Because + res/res_features.c was removed and main/cdr.c added, these + changes didn't make it to trunk and the 1.6.x branches + +2010-04-18 17:25 +0000 [r257768] Tilghman Lesher + + * configs/cdr_odbc.conf.sample: Removing unused configuration + parameters + +2010-04-16 21:22 +0000 [r257713] Dwayne M. Hubbard + + * /, apps/app_mixmonitor.c: Merged revisions 257686 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 + Apr 2010) | 21 lines Make the mixmonitor thread process audio + frames faster Mantis issue 17078 reports MixMonitor recordings + have shorter durations than the call duration. This was because + the mixmonitor thread was not processing frames from the + audiohook fast enough. The mixmonitor thread would slowly fall + behind the most recent audio frame and when the channel hangs up, + the mixmonitor thread would exit without processing the same + number of frames as the channel; leaving the mixmonitor recording + shorter than actual call duration. This revision fixes this issue + by moving the ast_audiohook_trigger_wait() and the subsequent + audiohook.status check into the block where the + ast_audiohook_read_frame() function returns NULL. (closes issue + #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded + by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review: + https://reviewboard.asterisk.org/r/611/ ........ + +2010-04-16 19:50 +0000 [r257646] Mark Michelson + + * channels/chan_sip.c: Make sure to fail a monitor if we receive a + negative response for a CC SUBSCRIBE. + +2010-04-16 19:25 +0000 [r257642] Dwayne M. Hubbard + + * channels/chan_dahdi.c: Enable PRI SERVICE message support in + chan_dahdi for the 'national' switchtype Revision 1072 of libpri + added SERVICE message support for the 'national' switchtype. The + attached patch enables the use of 'pri service' CLI commands on + dahdi channels that are configured for the 'national' switchtype. + (closes issue #17142) Reported by: dhubbard Patches: dw-ni2.patch + uploaded by dhubbard (license 733) Tested by: elguero, dhubbard + Review: https://reviewboard.asterisk.org/r/612/ + +2010-04-15 21:26 +0000 [r257493-257560] Tilghman Lesher + + * include/asterisk/app.h, /, tests/test_app.c, main/app.c: Merged + revisions 257544 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) + | 6 lines Allow application options with arguments to contain + parentheses, through a variety of escaping techniques. Fixes + SWP-1194 (ABE-2143). Review: + https://reviewboard.asterisk.org/r/604/ ........ + + * /, channels/chan_sip.c: Merged revisions 257467 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) + | 13 lines Don't recreate peer, when responding to a repeated + deregistration attempt. When a reply to a deregistration is lost + in transmit, the client retries the deregistration. Previously, + this would cause a realtime/autocreate peer to be loaded back + into memory, after it had already been correctly purged. Instead, + we just want to resend the reply without loading the peer. + (closes issue #16908) Reported by: kkm Patches: + 20100412__issue16908.diff.txt uploaded by tilghman (license 14) + Tested by: kkm ........ + +2010-04-15 19:41 +0000 [r257343-257427] Leif Madsen + + * /, doc/backtrace.txt: Merged revisions 257426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) + | 13 lines Update backtrace.txt documentation. Update the + backtrace.txt documentation so it conforms to the same layout as + other documents we've been working on recently. Additionally, add + a bunch of new information about gathering backtraces for crashes + and deadlocks, along with ways of verifying your file before + uploading it. Create a couple of one line commands for people to + generate the files we need. (closes issue #17190) Reported by: + lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen + (license 10) Tested by: lmadsen, pabelanger ........ + + * /, doc/backtrace.txt: Merged revisions 257342 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) + | 1 line Update address of the bug tracker. ........ + +2010-04-14 22:57 +0000 [r257262] Tilghman Lesher + + * main/features.c, configs/features.conf.sample: Yet another issue + where the conversion of the application delimiter to comma caused + an issue. Application arguments within the feature map could + possibly contain a comma, which conflicts with the syntax of the + features.conf configuration file. This patch allows the argument + to be wrapped in parentheses or quoted, to allow the application + arguments to be interpreted as a single configuration parameter. + (closes issue #16646) Reported by: pinga-fogo Patches: + 20100414__issue16646.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman Review: + https://reviewboard.asterisk.org/r/547/ + +2010-04-13 19:17 +0000 [r257191] Tilghman Lesher + + * channels/chan_sip.c: Also unref the pvt when we delete the + provisional keepalive job. (closes issue #16774) Reported by: + kowalma Patches: 20100315__issue16774.diff.txt uploaded by + tilghman (license 14) Tested by: falves11, jamicque Review: + https://reviewboard.asterisk.org/r/591/ + +2010-04-13 18:10 +0000 [r257146] Matthew Nicholson + + * main/manager.c, /, configs/manager.conf.sample: Merged revisions + 257070 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr + 2010) | 9 lines Add an option to restore past broken behavor of + the Events manager action Before r238915, certain values for the + EventMask parameter of the Events action would result in no + response being returned. This patch adds an option to restore + that broken behavior. Also while fixing this bug I discovered + that passing an empty EventMasks parameter would also result in + no response being returned, this has been fixed as well while + being preserved when the broken behavior is requested. (closes + issue #17023) Reported by: nblasgen Review: + https://reviewboard.asterisk.org/r/602/ ........ + +2010-04-13 16:33 +0000 [r257065] Tilghman Lesher + + * cdr/cdr_sqlite3_custom.c: Ensure that we can have commas within + cdr values. (closes issue #17001) Reported by: snuffy Patches: + 20100412__issue17001.diff.txt uploaded by tilghman (license 14) + Tested by: snuffy + +2010-04-13 16:18 +0000 [r256985-257032] Mark Michelson + + * configs/sip.conf.sample: Update sample dialstrings in + sip.conf.sample file. + + * funcs/func_srv.c: Address Russell's comments on func_srv from + reviewboard. * Change copyright date * Place channel in + autoservice when doing SRV lookup * Get rid of trailing + whitespace * Change logic in load_module function + + * main/ccss.c: Fix issue where recall would not happen when it + should. Specifically, the situation would happen when multiple + callers would request CC for a single generically-monitored + device. If the monitored device became available but the caller + did not answer the recall, then there was nothing that would poke + the CC core to let it know that it should attempt to recall + someone else instead. After careful consideration, I came to the + conclusion that the only area of Asterisk that needed to be + touched was the generic CC monitor. All other types of CC would + require something outside of Asterisk to invoke a recall for a + separate device. This was accomplished by changing the generic + monitor destructor to poke other generic monitor instances if the + device is currently available and the specific instance was + currently not suspended. In order to not accidentally trigger + recalls at bad times, the fit_for_recall flag was also added to + the generic_monitor_instance_list struct. This gets set as soon + as a monitored device becomes available. It gets cleared if a + CCNR request triggers the creation of a new generic monitor + instance. By doing this, we don't accidentally try to recall a + device when the monitored device was being monitored for CCNR and + never actually became available for recall in the first place. + This error was discovered by Steve Pitts during in-house testing + at Digium. + +2010-04-12 17:29 +0000 [r256860-256901] Leif Madsen + + * /, doc/HOWTO_collect_debug_information.txt (added): Merged + revisions 256900 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) + | 15 lines Add How-To document on collecting debugging info for + issues.asterisk.org Paul Belanger has been helping a lot with bug + tracking recently and created this document that we can now point + to when additional debugging information is required. This + document will help those filing issues to know how to get the + information required when filing their issues. This will make + things easier on the developers. Initial text and changes by + pabelanger. Tweaks and editing by myself. (closes issue #17159) + Reported by: pabelanger Patches: + HOWTO_collect_debug_information.txt.patch uploaded by lmadsen + (license 10) Tested by: tzafrir, pabelanger, lmadsen ........ + + * apps/app_voicemail.c: Remove silly debug message that is not + useful. (issue #17159) + +2010-04-12 14:47 +0000 [r256823] David Vossel + + * channels/chan_sip.c: gives channel reference before unlocking it + and using setvar helper. To guarantee the channel is valid when + calling setvar on the MASTER_CHANNEL dialplan function, a channel + reference must be taken before unlocking. Thanks to russell for + pointing out the error. + +2010-04-12 14:39 +0000 [r256821] Leif Madsen + + * main/logger.c: CLI command logger set level auto complete. A + simple patch to enable auto tab complete. (closes issue #17152) + Reported by: pabelanger Patches: 0017152.patch uploaded by + pabelanger (license 224) + +2010-04-12 02:19 +0000 [r256745-256783] Russell Bryant + + * tests/test_substitution.c: test_substitution expects func_curl to + be present to work. + + * tests/test_pbx.c: Add ASTERISK_FILE_VERSION() macro + +2010-04-10 08:33 +0000 [r256704] Tzafrir Cohen + + * contrib/scripts/safe_asterisk.8, doc/asterisk.8, + contrib/scripts/autosupport.8, contrib/scripts/astgenkey.8: fix + hyphen vs. minus in man pages In troff '-' is used for a hyphen. + A minus is denoted by '\-' . This is normally also used for a + dash. This patch converts all '-'-s that are minuses or dashes to + '\-'. + +2010-04-09 22:20 +0000 [r256646-256661] Mark Michelson + + * channels/chan_sip.c, main/ccss.c: Remove status_response + callbacks where they are not needed. + + * channels/chan_local.c: Prevent crash when originating a call to a + local channel. Call completion code tries to grab the call + completion parameters from the requesting channel during + local_request. When originating a call to a local channel, + however, this channel is NULL. This was causing an issue for me + when trying to run a test script. + +2010-04-09 19:46 +0000 [r256569-256608] Richard Mudgett + + * doc/CCSS_architecture.pdf (added): Merge CCSS architecture + document from CCSS branch. + + * channels/sig_pri.h, configure, include/asterisk/autoconfig.h.in: + Remove PRI CCSS BUGBUG message and update configure script. + +2010-04-09 16:04 +0000 [r256485-256530] Mark Michelson + + * channels/sip/reqresp_parser.c, channels/sip/include/sip.h, + channels/sip/include/reqresp_parser.h: Add routines for parsing + SIP URIs consistently. From the original issue report opened by + Nick Lewis: Many sip headers in many sip methods contain the ABNF + structure name-andor-addr = name-addr / addr-spec Examples + include the to-header, from-header, contact-header, + replyto-header At the moment chan_sip.c makes various different + attempts to parse this name-andor-addr structure for each header + type and for each sip method with sometimes limited degrees of + success. I recommend that this name-andor-addr structure be + parsed by a dedicated function and that it be used irrespective + of the specific method or header that contains the + name-andor-addr structure Nick has also included unit tests for + verifying these routines as well, so...heck yeah. (closes issue + #16708) Reported by: Nick_Lewis Patches: + reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis + (license 657 Review: https://reviewboard.asterisk.org/r/549 + + * channels/chan_sip.c, tests/test_gosub.c, funcs/func_srv.c: Fix + some compiler errors that popped up after the CCSS merge. + + * apps/app_dial.c, configs/chan_dahdi.conf.sample, + include/asterisk/devicestate.h, include/asterisk/xml.h, + channels/chan_local.c, doc/tex/ccss.tex (added), main/ccss.c + (added), channels/chan_sip.c, configure.ac, main/xml.c, + include/asterisk/channel.h, configs/manager.conf.sample, + include/asterisk/channelstate.h (added), + include/asterisk/manager.h, CHANGES, channels/sig_pri.c, + channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c, + main/manager.c, funcs/func_callcompletion.c (added), + channels/sig_analog.c, channels/sig_analog.h, + configs/ccss.conf.sample (added), include/asterisk/rtp_engine.h, + include/asterisk/frame.h, include/asterisk/ccss.h (added), + doc/tex/asterisk.tex, main/asterisk.c, + channels/sip/include/sip.h: Merge Call completion support into + trunk. From Reviewboard: CCSS stands for Call Completion + Supplementary Services. An admittedly out-of-date overview of the + architecture can be found in the file doc/CCSS_architecture.pdf + in the CCSS branch. Off the top of my head, the big differences + between what is implemented and what is in the document are as + follows: 1. We did not end up modifying the Hangup application at + all. 2. The document states that a single call completion monitor + may be used across multiple calls to the same device. This proved + to not be such a good idea when implementing protocol-specific + monitors, and so we ended up using one monitor per-device + per-call. 3. There are some configuration options which were + conceived after the document was written. These are documented in + the ccss.conf.sample that is on this review request. For some + basic understanding of terminology used throughout this code, see + the ccss.tex document that is on this review. This implements + CCBS and CCNR in several flavors. First up is a "generic" + implementation, which can work over any channel technology + provided that the channel technology can accurately report device + state. Call completion is requested using the dialplan + application CallCompletionRequest and can be canceled using + CallCompletionCancel. Device state subscriptions are used in + order to monitor the state of called parties. Next, there is a + SIP-specific implementation of call completion. This method uses + the methods outlined in draft-ietf-bliss-call-completion-06 to + implement call completion using SIP signaling. There are a few + things to note here: * The agent/monitor terminology used + throughout Asterisk sometimes is the reverse of what is defined + in the referenced draft. * Implementation of the draft required + support for SIP PUBLISH. I attempted to write this in a + generic-enough fashion such that if someone were to want to write + PUBLISH support for other event packages, such as dialog-state or + presence, most of the effort would be in writing callbacks + specific to the event package. * A subportion of supporting + PUBLISH reception was that we had to implement a PIDF parser. The + PIDF support added is a bit minimal. I first wrote a validation + routine to ensure that the PIDF document is formatted properly. + The rest of the PIDF reading is done in-line in the + call-completion-specific PUBLISH-handling code. In other words, + while there is PIDF support here, it is not in any state where it + could easily be applied to other event packages as is. Finally, + there are a variety of ISDN-related call completion protocols + supported. These were written by Richard Mudgett, and as such I + can't really say much about their implementation. There are notes + in the CHANGES file that indicate the ISDN protocols over which + call completion is supported. Review: + https://reviewboard.asterisk.org/r/523 + + * main/srv.c, channels/chan_sip.c, funcs/func_srv.c (added), + CHANGES, include/asterisk/srv.h: func_srv and explicit + specification of a remote IP for SIP. From Review Board: There + are two interrelated changes here. First, there is the + introduction of func_srv. This adds two new read-only dialplan + functions, SRVQUERY and SRVRESULT. They work very similarly to + the ENUMQUERY and ENUMRESULT functions, except that this allows + one to query SRV records instead. In order to facilitate this + work, I added a couple of new API calls to srv.h. + ast_srv_get_record_count tells the number of records returned by + an SRV lookup. This number is calculated at the time of the SRV + lookup. ast_srv_get_nth_record allows one to get a numbered SRV + record. Second, there is the modification to chan_sip that allows + one to specify a hostname or IP address (along with a port) to + send an outgoing INVITE to when dialing a SIP peer. This goes + hand-in-hand with func_srv. You can query SRV records and then + use the host and port from the results to dial via a specific + host instead of what is configured in sip.conf. Review: + https://reviewboard.asterisk.org/r/608 SWP-1200 + +2010-04-08 16:35 +0000 [r256428] Kevin P. Fleming + + * /, Makefile.rules, build_tools/make_linker_version_script: Ensure + that linker version scripts (used for symbol export control) + always exist. Using wildcard matching in the Makefile is not + adequate to determine whether an export file should exist for a + module or not, so instead we'll just create one if the module + needs one, or copy the default one if it does not. + +2010-04-06 19:28 +0000 [r256370] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/lock.h: Mac OS X does not support comparing a + mutex to its initializer. Create a test for this. + +2010-04-06 14:42 +0000 [r256319] David Vossel + + * channels/chan_sip.c: fixes deadlock in chan_sip caused by usage + of MASTER_CHANNEL dialplan function (closes issue #16767) + Reported by: lmsteffan Patches: deadlock_16767v3.diff uploaded by + dvossel (license 671) Review: + https://reviewboard.asterisk.org/r/606/ + +2010-04-06 00:39 +0000 [r256265] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 256225 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 + Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not + protected by PRI lock. SWP-1231 ABE-2163 ........ + +2010-04-05 15:14 +0000 [r256161] Leif Madsen + + * doc/tex/localchannel.tex: Fix for localchannel.tex to allow PDFs + to be generated again. + +2010-04-03 02:12 +0000 [r256103-256104] Richard Mudgett + + * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c, + include/asterisk/channel.h, main/cel.c, channels/sig_pri.c, + channels/chan_iax2.c, apps/app_queue.c, channels/chan_oss.c, + funcs/func_redirecting.c, main/channel.c, main/dial.c, + channels/chan_dahdi.c, channels/chan_misdn.c, + apps/app_dumpchan.c, res/res_agi.c, channels/chan_h323.c, + res/snmp/agent.c, apps/app_amd.c, funcs/func_callerid.c: + Consolidate ast_channel.cid.cid_rdnis into + ast_channel.redirecting.from.number. SWP-1229 ABE-2161 * Ensure + chan_local.c:local_call() will not leak cid.cid_dnid when + copying. + + * apps/app_dial.c: Using the Dial application f option when the + call is forwarded will likely crash. Fix app_dial.c:do_forward() + OPT_FORCECLID setting cid.cid_num with a stack allocated string + instead of a heap allocated string. + +2010-04-02 23:55 +0000 [r256010-256019] Russell Bryant + + * apps/app_meetme.c: Export MEETMEBOOKID and fix pin-less + conferences with realtime conferences (closes issue #16866) + Reported by: DEA Patches: rt-meetme-options.txt uploaded by DEA + (license 3) Tested by: DEA Review: + https://reviewboard.asterisk.org/r/582/ + + * channels/chan_local.c, /: Merged revisions 256014 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 + Apr 2010) | 9 lines Resolve a deadlock that occurs due to a + pointless call to ast_bridged_channel() (closes issue #16840) + Reported by: bzing2 Patches: patch.txt uploaded by bzing2 + (license 902) issue_16840.rev1.diff uploaded by russell (license + 2) Tested by: bzing2, russell ........ + + * main/channel.c, /: Merged revisions 256009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010) + | 2 lines Remove extremely verbose debug message. ........ + +2010-04-02 20:19 +0000 [r255952] Tilghman Lesher + + * main/asterisk.c: Pass the PID of the Asterisk process, not the + PID of the canary. (closes issue #17065) Reported by: + globalnetinc Patches: astcanary.patch uploaded by makoto (license + 38) Tested by: frawd, globalnetinc + +2010-04-02 18:57 +0000 [r255906] Kevin P. Fleming + + * res/res_ael_share.exports.in (added), codecs, + res/res_pktccops.exports.in (added), utils, + res/res_monitor.exports.in (added), Makefile.moddir_rules, + res/res_smdi.exports.in (added), Makefile.rules, cdr, + res/res_agi.exports.in (added), formats, main/asterisk.exports + (removed), res/res_odbc.exports (removed), + res/res_calendar.exports (removed), apps/app_voicemail.exports + (removed), bridges, res/res_odbc.exports.in (added), + main/asterisk.exports.in (added), apps/app_voicemail.exports.in + (added), res/res_calendar.exports.in (added), + res/res_features.exports (removed), res/res_fax.exports.in + (added), pbx, res/res_adsi.exports.in (added), + res/res_jabber.exports (removed), res/res_pktccops.exports + (removed), channels, res/res_jabber.exports.in (added), + main/Makefile, res/res_smdi.exports (removed), tests, apps, cel, + res/res_agi.exports (removed), addons, res/res_speech.exports + (removed), Makefile, funcs, res/res_speech.exports.in (added), + res/res_fax.exports (removed), main, res/res_adsi.exports + (removed), res/res_features.exports.in (added), + res/res_ael_share.exports (removed), + build_tools/make_linker_version_script (added), res, + res/res_monitor.exports (removed): Allow symbol export filtering + to work properly on platforms that have symbol prefixes. Some + platforms prefix externally-visible symbols in object files + generated from C sources (most commonly, '_' is the prefix). On + these platforms, the existing symbol export filtering process + ends up suppressing all the symbols that are supposed to be left + visible. This patch allows the prefix string to be supplied to + the top-level Makefile in the LINKER_SYMBOL_PREFIX variable, and + then generates the linker scripts as required to include the + prefix supplied. + +2010-04-02 06:45 +0000 [r255850-255851] Michiel van Baak + + * channels/chan_skinny.c: Ignore Redial softkey when no previous + dialed number is known (closes issue #17126) Reported by: wedhorn + Patches: skinny79xx_redial1.diff uploaded by wedhorn (license 30) + + * channels/chan_skinny.c: Cleanup transmit_* functions Bulk lot of + generally trivial changes for cleaning up the transmit stuff. + Line state request has been modified for line only responses. + (closes issue #16994) Reported by: wedhorn Patches: + skinny-clean07.diff uploaded by wedhorn (license 30) Tested by: + wedhorn + +2010-04-01 18:16 +0000 [r255796] Tilghman Lesher + + * include/asterisk/lock.h: Fix DEBUG_THREADS build on Darwin. + (closes issue #16828) Reported by: oej Patches: + 20100331__issue16828.diff.txt uploaded by tilghman (license 14) + +2010-04-01 16:09 +0000 [r255751] Matthew Nicholson + + * configs/sip.conf.sample: Removed documentation of the non + existent 'both' option to 'faxdetect' in sip.conf + +2010-03-31 22:35 +0000 [r255701] Mark Michelson + + * channels/chan_sip.c: Fix improper comaparison of anonymous URI + when getting P-Asserted-Identity. There was a bug where we split + the URI on the @ sign and then attempted to compare to + "anonymous@anonymous.invalid" afterwards. This comparison could + never evaluate true. So now we keep a copy of the URI prior to + the split so that the comparison is valid. + +2010-03-31 19:13 +0000 [r255592] Tilghman Lesher + + * /, apps/app_voicemail.c: Recorded merge of revisions 255591 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) + | 15 lines Ensure line terminators in email are consistent. Fixes + an issue with certain Mail Transport Agents, where attachments + are not interpreted correctly. (closes issue #16557) Reported by: + jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by + tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt + uploaded by tilghman (license 14) + 20100308__issue16557__trunk.diff.txt uploaded by tilghman + (license 14) Tested by: ebroad, zktech Reviewboard: + https://reviewboard.asterisk.org/r/544/ ........ + +2010-03-31 17:48 +0000 [r255504] Leif Madsen + + * apps/app_dial.c, /, configs/sip.conf.sample: Add documentation + clarifying when 't' and 'T' can be used. (closes issue #17021) + Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad + +2010-03-30 20:56 +0000 [r255323-255410] Russell Bryant + + * /, channels/chan_h323.c: Merged revisions 255409 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 + Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does + not start. ........ + + * /, pbx/pbx_dundi.c: Merged revisions 255322 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010) + | 2 lines Don't make Asterisk not start if pbx_dundi fails to + initialize. ........ + +2010-03-29 14:07 +0000 [r255281] Jared Smith + + * apps/app_confbridge.c, CHANGES: This patch adds custom device + state handling for ConfBridge conferences, matching the devstate + handling of the MeetMe conferences. Review: + https://reviewboard.asterisk.org/r/572/ Closes issue #16972 + +2010-03-29 05:10 +0000 [r255240] Russell Bryant + + * main/event.c: Remove a debugging log entry. + +2010-03-27 23:51 +0000 [r255199] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, + addons/chan_ooh323.c, addons/ooh323c/src/ooh323.h, + addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: + corrections in gk interface, small fixes in call clearing. + +2010-03-27 14:44 +0000 [r255158] Sean Bright + + * apps/app_voicemail.c: We need to inclde sys/wait.h on OpenBSD to + get WEXITSTATUS. + +2010-03-27 06:09 +0000 [r255117] Tilghman Lesher + + * pbx/pbx_spool.c: inotify support for pbx_spool This should give a + good speed boost, in that one particular thread isn't waking up + once a second to read directory contents. Reviewboard: + https://reviewboard.asterisk.org/r/137/ + +2010-03-26 19:27 +0000 [r255021-255066] Leif Madsen + + * configs/sip.conf.sample: Replace some documentation from 1.6.x + back into trunk. This documentation associated wth tlsbindaddr is + still useful so lets synchronize it between trunk and 1.6.x + branches. (issue #17054) + + * configs/sip.conf.sample: Update confusing documentation for + tlsbindaddr. Update some confusing documentation for the + tlsbindaddr option in sip.conf.sample. Point at a link instead + which has better documentation. (closes issue #17054) Reported + by: klaus3000 + +2010-03-26 16:27 +0000 [r254976] Sean Bright + + * contrib/scripts/live_ast: Work around a bug in dash on Ubuntu by + checking the number of arguments before shift'ing. Reported and + tested by pabelanger. + +2010-03-25 23:38 +0000 [r254931] Kevin P. Fleming + + * addons/chan_ooh323.h, addons/ooh323c/src/ooasn1.h, + addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c, + addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooTimer.c, + addons/ooh323c/src/dlist.c, addons/ooh323c/src/eventHandler.c, + addons/ooh323c/src/ooCapability.c, addons/ooh323cDriver.c, + addons/mp3/interface.c, addons/ooh323cDriver.h, + addons/ooh323c/src/rtctype.c, addons/ooh323c/src/ooCalls.c, + addons/ooh323c/src/encode.c, addons/ooh323c/src/ooUtils.c, + addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooh323ep.c, + addons/ooh323c/src/ooports.c, addons/mp3/decode_ntom.c, + addons/ooh323c/src/memheap.c, addons/ooh323c/src/ooh323.c, + addons/ooh323c/src/ooh245.c, addons/mp3/common.c, + addons/ooh323c/src/decode.c, addons/ooh323c/src/context.c, + addons/ooh323c/src/perutil.c, addons/mp3/layer3.c, + addons/ooh323c/src/oochannels.c, + addons/ooh323c/src/ooCmdChannel.c, + addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c, + addons/ooh323c/src/ootrace.c: Use "local" instead of "system" + header file inclusion. Now that these files are in the tree, they + should prefer the tree's local copy of all Asterisk headers over + any that may be installed. + +2010-03-25 21:39 +0000 [r254884] Russell Bryant + + * addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooSocket.h: Fix + a number of other build problems on Mac OS X. + +2010-03-25 20:41 +0000 [r254802] Jason Parker + + * utils/Makefile, /: Merged revisions 254800 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) | + 1 line Don't remove local copies of utils in uninstall. ........ + +2010-03-25 20:41 +0000 [r254718-254801] Russell Bryant + + * addons/chan_ooh323.h: Resolve compiler warning on FreeBSD. + + * addons/ooh323c/src/ooh323.c, addons/Makefile, + addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ootrace.c: Fix + chan_ooh323 so it works on Mac OS X, as well. + + * channels/chan_usbradio.c: chan_usbradio depends on alsa. + +2010-03-25 18:38 +0000 [r254636-254638] Kevin P. Fleming + + * .cleancount: Bump cleancount due to ast_channel change. + + * include/asterisk/channel.h: Remove no-longer-used (and unsafe) + field in ast_channel for linked lists. The ast_channel structure + had a field used for linking a channel into a linked list, but + now that ast_channel structures are ao2 objects, this is no + longer needed, and could be harmful as ao2 objects really + shouldn't ever be placed into linked lists (since those lists + don't assist with reference count management on the objects). + + * addons/Makefile: Get chan_ooh323 building again after recent + build system changes. + +2010-03-25 17:52 +0000 [r254454-254557] Mark Michelson + + * tests/test_acl.c (added): Add unit test for testing ACL + functionality. There are two unit tests contained here. 1. + "Invalid ACL" This attempts to read a bunch of badly formatted + ACL entries and add them to a host access rule. The goal of this + test is to be sure that all invalid entries are rejected as they + should be. 2. "ACL" This sets up four ACLs. One is a permit all, + one is a deny all, and the other two have specific rules about + which subnets are allowed and which are not. Then a set of test + addresses is used to determine whether we would allow those + addresses to access us when each ACL is applied. This test, by + the way, was what resulted in AST-2010-003's creation. Review: + https://reviewboard.asterisk.org/r/532 + + * include/asterisk/acl.h, /: Merged revisions 254552 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, + 25 Mar 2010) | 5 lines Add doxygen for acl.h Review: + https://reviewboard.asterisk.org/r/528 ........ + + * channels/sip/dialplan_functions.c: Add new rtpsource options to + the CHANNEL function. This adds rtpsource options analogous to + the rtpdest functions that already exist. In addition, this fixes + potential crashes which could result due to trying to read values + from nonexistent RTP streams. + + * res/res_rtp_asterisk.c, /: Recorded merge of revisions 254452 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar + 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection. + Here is a copy and paste of the details from my request on + reviewboard that dealt with these changes: Fix 1. The first + change in place is to fix Mantis issue 15811, which deals with a + situation where Asterisk will incorrectly interpret out of order + RFC2833 frames as duplicate DTMF digits. For instance, we would + receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: + DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 + seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch + when we received the frame with seqno 5, we would interpret this + as a new DTMF 1. With this patch, we will check the seqno of the + incoming digit and not process the frame if the seqno is lower + than the last recorded seqno. Note that we do not record the + seqno of the dropped DTMF frame for future processing. While the + above situation is what was designed to be fixed, the patch is + written in such a way that the following would also be fixed too: + seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) + seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno + 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In + this second situation, the beginning of the DTMF 2 arrives before + the final end frame of the DTMF 1. With the patch, seqno 12 is no + processed and thus we properly interpret the DTMF. Fix 2. The + second change in place is to fix an issue like the following: + seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet + lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end) + *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had + code in place that was supposed to properly end the previously + unended DTMF 1. The problem was that the code was essentially a + no-op. The code would set up an end frame for the DTMF 1 but + would immediately overwrite the frame with the begin for DTMF 2. + I changed process_dtmf_rfc2833() so that instead of returning a + single frame, it is given as an output parameter a list of + frames. Each frame that needs to be returned is appended to this + list. Fix 3. The final change is a minor one where an + AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco + DTMF or an RFC 3389 frame and no frame was returned, then we + would return &ast_null_frame. The problem is that earlier in the + function, we may have generated an AST_CONTROL_SRCCHANGE frame + and put it in the list of frames we wish to return. This frame + would be lost in such a case. The patch fixes this problem + ........ + +2010-03-25 16:03 +0000 [r254453] Terry Wilson + + * /, main/file.c: Merged revisions 254451 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010) + | 2 lines Handle new SRCCHANGE control message here too ........ + +2010-03-25 15:27 +0000 [r254450] Kevin P. Fleming + + * main/channel.c, channels/chan_sip.c, res/res_fax.c, + configs/sip.conf.sample, include/asterisk/frame.h, + channels/sip/include/sip.h: Improve handling of T.38 re-INVITEs + that arrive before a T.38-capable application is executing on a + channel. This patch addresses an issue found during working with + end-users using res_fax. If an incoming call is answered in the + dialplan, or jumps to the 'fax' extension due to reception of a + CNG tone (with faxdetect enabled), and then the remote endpoint + sends a T.38 re-INVITE, it is possible for the channel's T.38 + state to be 'T38_STATE_NEGOTIATING' when the application starts + up. Unfortunately, even if the application wants to use T.38, it + can't respond to the peer's negotiation request, because the + AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent + originally has been lost, and the application needs the content + of that frame to be able to formulate a reply. This patch adds a + new 'request' type to AST_CONTROL_T38_PARAMETERS, + AST_T38_REQUEST_PARMS. If the application sends this request, + chan_sip will re-send the original control frame (with + AST_T38_REQUEST_NEGOTIATE as the request type), and the + application can respond as normal. If this occurs within the five + second timeout in chan_sip, the automatic cancellation of the + peer reinvite will be stopped, and the application will 'own' the + negotiation process from that point onwards. This also improves + the code path in chan_sip to allow sip_indicate(), when called + for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero + response, which should have been in place before since the + control frame *can* fail to be processed properly. It also + modifies ast_indicate() to return whatever result the channel + driver returned for this control frame, rather than converting + all non-zero results into '-1'. Finally, the new request type + intentionally returns a positive value, so that an application + that sends AST_T38_REQUEST_PARMS can know for certain whether the + channel driver accepted it and will be replying with a control + frame of its own, or whether it was ignored (if the + sip_indicate()/ast_indicate() path had properly supported failure + responses before, this would not be necessary). This patch also + modifies res_fax to take advantage of the new request. In + addition, this patch makes sip_t38_abort() actually lock the + private structure before doing its work... bad programmer, no + donut. This patch also enhances chan_sip's 'faxdetect' support to + allow triggering on T.38 re-INVITEs received as well as CNG tone + detection. Review: https://reviewboard.asterisk.org/r/556/ + +2010-03-25 15:21 +0000 [r254446] Leif Madsen + + * res/res_agi.c: handle_speechset has 4 arguments. Update code to + reflect that handle_speechset has 4 arguments. (closes issue + #17093) Reported by: gpatri Patches: res_agi.patch uploaded by + gpatri (license 1014) Tested by: pabelanger, mmichelson + +2010-03-25 10:09 +0000 [r254406] Tzafrir Cohen + + * channels/chan_dahdi.c: remove unneeded explicit channel in dahdi + ioctls This patch removes some cases where the channel number for + an ioctl was passed as a member in a struct rather then through + the file descriptor. The gain setting functions passed around a + channel which is always 0, and thus this parameter is simply + dropped. Review: https://reviewboard.asterisk.org/r/584/ + +2010-03-24 21:10 +0000 [r254362] Mark Michelson + + * main/pbx.c: Fix potential invalid reads that could occur in pbx.c + Here is a cut and paste of my review request for this change: + This past weekend, Russell ran our current suite of unit tests + for Asterisk under valgrind. The PBX pattern match test caused + valgrind to spew forth two invalid read errors. This patch + contains two changes that shut valgrind up and do not cause any + new memory leaks. Change 1: In + ast_context_remove_extension_callerid2, valgrind reported an + invalid read in the for loop close to the function's end. + Specifically, one of the the strcmp calls in the loop control was + reading invalid memory. This was because the caller of + ast_context_remove_extension_callerid2 (__ast_context destroy in + this case) passed as a parameter a shallow copy of an ast_exten's + exten field. This same ast_exten was what was destroyed inside + the for loop, thus any iterations of the for loop beyond the + destruction of the ast_exten would result in invalid reads. My + fix for this is to make a copy of the ast_exten's exten field and + pass the copy to ast_context_remove_extension_callerid2. In + addition, I have also acted similarly with the ast_exten's + matchcid field. Since in this case a NULL is handled quite + differently than an empty string, I needed to be a bit more + careful with its handling. Change 2: In __ast_context_destroy, we + iterated over a hashtab and called + ast_context_remove_extension_callerid2 on each item. + Specifically, the hashtab over which we were iterating was an + ast_exten's peer_table. Inside of + ast_context_remove_extension_callerid2, we could possibly destroy + this ast_exten, which also caused the hashtab to be freed. + Attempting to call ast_hashtab_end_traversal on the hashtab + iterator caused an invalid read to occur when trying to read the + iterator->tab->do_locking field since iterator->tab had already + been freed. My handling of this problem is a bit less + straightforward. With each iteration over the hashtab's contents, + we set a variable called "end_traversal" based on the return of + ast_context_remove_extension_callerid2. If 0 is ever returned, + then we know that the extension was found and destroyed. Because + of this, we cannot call ast_hashtab_end_traversal because we will + be guaranteeing a read of invalid memory. In such a case, we + forego calling ast_hashtab_end_traversal and instead call + ast_free on the hashtab iterator. Review: + https://reviewboard.asterisk.org/r/585 + +2010-03-24 18:13 +0000 [r254277-254321] Jeff Peeler + + * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: + Allow configuration of minsecs and nextaftercmd per mailbox. + Previously only configurable globally. A unit test has also been + written to provide protection against parse failures for + supported mailbox options. (closes issue #16864) Reported by: + kobaz Patches: voicemail2.patch uploaded by kobaz (license 834) + Review: https://reviewboard.asterisk.org/r/555/ + + * /, res/res_monitor.c: Merged revisions 254235 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) + | 72 lines Ensure that monitor recordings are written to the + correct location (again) This is an extension to 248860. As such + the dialplan test has been extended: ; non absolute path, not + combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test) + exten => 5040, n, dial(sip/5001) ; absolute path, not combined + exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten => + 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1, + monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ; + combined: changemonitor from non absolute to no path (leaves + tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m) + exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n, + dial(sip/5001) ; combined: changemonitor from no path to non + absolute path exten => 5044, 1, monitor(wav,monitor_test6,m) + exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this + wasn't possible before exten => 5044, n, dial(sip/5001) ; non + absolute path, combined exten => 5045, 1, + monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n, + dial(sip/5001) ; absolute path, combined exten => 5046, 1, + monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n, + dial(sip/5001) ; no path, combined exten => 5047, 1, + monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ; + combined: changemonitor from non absolute to absolute (leaves + tmp/jeff) exten => 5048, 1, + monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n, + changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n, + dial(sip/5001) ; combined: changemonitor from absolute to non + absolute (leaves /tmp/jeff) exten => 5049, 1, + monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n, + changemonitor(tmp/jeff/monitor_test14) exten => 5049, n, + dial(sip/5001) ; combined: changemonitor from no path to absolute + exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n, + changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n, + dial(sip/5001) ; combined: changemonitor from absolute to no path + (leaves /tmp/jeff) exten => 5051, 1, + monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n, + changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ; + not combined: changemonitor from non absolute to no path (leaves + tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19) + exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n, + dial(sip/5001) ; not combined: changemonitor from no path to non + absolute exten => 5053, 1, monitor(wav,monitor_test21) exten => + 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n, + dial(sip/5001) ; not combined: changemonitor from non absolute to + absolute (leaves tmp/jeff) exten => 5054, 1, + monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n, + changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n, + dial(sip/5001) ; not combined: changemonitor from absolute to non + absolute (leaves /tmp/jeff) exten => 5055, 1, + monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n, + changemonitor(tmp/jeff/monitor_test25) exten => 5055, n, + dial(sip/5001) ; not combined: changemonitor from no path to + absolute exten => 5056, 1, monitor(wav,monitor_test26) exten => + 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056, + n, dial(sip/5001) ; not combined: changemonitor from absolute to + no path (leaves /tmp/jeff) exten => 5057, 1, + monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n, + changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001) + ........ + +2010-03-23 22:48 +0000 [r254162] Tzafrir Cohen + + * main/asterisk.c: make 'core show settings' should show all + settable directories (closes issue #17086) Reported by: tzafrir + Patches: asterisk_extra_settings_dirs.diff uploaded by tzafrir + (license 46) + +2010-03-23 22:35 +0000 [r254159] Russell Bryant + + * main/test.c: Put test output for a failure in a CDATA section in + the XML results. + +2010-03-23 21:17 +0000 [r254050] Jeff Peeler + + * main/channel.c: Exit native bridging early for greater timing + accuracy with warnings This changes native bridging to break one + millisecond early so that the more accurate timeval calculations + done in the generic bridge can be performed using the bridge + config. Currently the time between exiting native bridging + slightly late can sometimes cause a large enough discrepancy for + warnings to be missed. For the record, 1.4 does not attempt to + native bridge at all when warnings are enabled. (closes issue + #15815) Reported by: adomjan Review: + https://reviewboard.asterisk.org/r/577/ + +2010-03-23 20:52 +0000 [r254045] Sean Bright + + * apps/app_queue.c: Remove unused structure member in app_queue. + (closes issue #15494) Reported by: makoto + +2010-03-23 19:19 +0000 [r254001] Tzafrir Cohen + + * tests/Makefile: Change the name of the category 'TEST' to match + the name of the subdir + +2010-03-23 16:52 +0000 [r253958] Terry Wilson + + * main/http.c: Don't act like an http write failed when it didn't + fwrite returns the number of items written, not the number of + bytes + +2010-03-23 14:22 +0000 [r253917] Kevin P. Fleming + + * codecs/Makefile, include/asterisk/logger.h, main/Makefile, + Makefile.moddir_rules, pbx/Makefile, res/Makefile, CHANGES, + channels/Makefile, include/asterisk/options.h, main/cli.c: Change + per-file debug and verbose levels to be per-module, the way users + expect them to work. 'core set debug' and 'core set verbose' can + optionally change the level for a specific filename; however, + this is actually for a specific source file name, not the module + that source file is included in. With examples like chan_sip, + chan_iax2, chan_misdn and others consisting of multiple source + files, this will not lead to the behavior that users expect. If + they want to set the debug level for chan_sip, they want it set + for all of chan_sip, and not to have to also set it for + reqresp_parser and other files that comprise the chan_sip module. + This patch changes this functionality to be module-name based + instead of file-name based. To make this work, some Makefile + modifications were required to ensure that the AST_MODULE + definition is present in each object file produced for each + module as well. Review: https://reviewboard.asterisk.org/r/574/ + +2010-03-22 20:32 +0000 [r253872] Mark Michelson + + * main/asterisk.c: Initialize channels prior to loading "preload" + modules. We can have bad results when a module, upon being + loaded, attempts to reference the channels container if the + container hasn't yet been initialized. I saw this happen by + trying to preload pbx_config.so and having a hint defined which + referenced a non-existent SIP peer. + +2010-03-22 19:52 +0000 [r253800] Matthew Nicholson + + * /, main/features.c: Merged revisions 253799 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar + 2010) | 4 lines Unconditionally copy the caller's account code to + the called party. (related to issue #16331) ........ + +2010-03-22 19:05 +0000 [r253712-253758] Tilghman Lesher + + * contrib/scripts/dbsep.cgi: Update query should be an UPDATE, not + a SELECT. + + * contrib/scripts/dbsep.cgi: Return the list for later + manipulation. This fixes an issue with the update procedure. + Debugging with mmichelson. + + * contrib/scripts/dbsep.cgi, configs/dbsep.conf.sample: Accomodate + equal signs in DSNs and add documentation, based upon + mmichelson's feedback. + +2010-03-20 16:50 +0000 [r253536-253579] Russell Bryant + + * funcs/func_strings.c: Fix memory corruption found by unit tests. + ast_str_reset() was being called on a potentially uninitialized + pointer. Valgrind is my hero, once again. + + * cel/cel_pgsql.c, main/tcptls.c, main/manager.c, main/features.c, + main/test.c, cdr/cdr_pgsql.c, main/stdtime/localtime.c, + main/cel.c: Resolve more compiler warnings on FreeBSD. + + * apps/app_voicemail.c: Include sys/wait.h on FreeBSD to get the + WEXITSTATUS() macro. + + * apps/app_dial.c, apps/app_followme.c: Resolve compiler warnings + on FreeBSD. + + * pbx/pbx_dundi.c: Resolve a compiler warning on FreeBSD. + + * channels/chan_dahdi.c: Use SHRT_MAX instead of MAXSHORT. These + changes fix build issues I had with this module on FreeBSD. + +2010-03-19 07:37 +0000 [r253490] Alec L Davis + + * main/astobj2.c: prevent segfault if bad magic number is + encountered. internal_ao2_ref uses INTERNAL_OBJ which mzy report + 'bad magic number', but internal_ao2_ref continues on, causing + segfault. Although AO2_MAGIC number is checked by INTERNAL_OBJ + before internal_ao2_ref is called, A02_MAGIC is being destroyed + (or a wrong pointer) by the time internal_ao2_ref uses + INTERNAL_OBJ. internal_ao2_ref now returns -1 if INTERNAL_OBJ + encouters a bad magic number. (issue #17037) Reported by: + alecdavis Patches: bug17037.diff.txt uploaded by alecdavis + (license 585) Tested by: alecdavis + +2010-03-18 18:23 +0000 [r253357-253378] Russell Bryant + + * main/asterisk.c: Update comment to reflect new timeout value. + + * main/asterisk.c: Increase CLI command output timeout for asterisk + -rx to 60 seconds. (closes issue #17049) Reported by: russell + Tested by: russell Review: + https://reviewboard.asterisk.org/r/573/ + +2010-03-18 17:52 +0000 [r253345] Leif Madsen + + * apps/app_userevent.c: Change usage of pipe to comma in UserEvent + docs. Change the example usage of pipe as a separator to comma in + the UserEvent documentation. (closes issue #16961) Reported by: + jlpedrosa + +2010-03-18 15:59 +0000 [r253261] Philippe Sultan + + * res/res_jabber.c: Prevent a crash when a buddy gets offline. + (closes issue #16760) Reported by: fiddur Patches: 248394.diff + uploaded by fiddur (license 678)i with modifications by me Tested + by: fiddur, phsultan + +2010-03-18 15:46 +0000 [r253256] Leif Madsen + + * /, doc/tex/localchannel.tex: Update to new Local channel + documentation. Add same changes as commit to 1.4, but convert to + TeX. (issue #16963) Reported by: kobaz Patches: + localchannel-2.txt uploaded by kobaz (license 834) + +2010-03-18 15:45 +0000 [r253255] Tilghman Lesher + + * main/stdtime/localtime.c: Just in case of a race, send the signal + on interrupt. + +2010-03-17 19:06 +0000 [r253205] Leif Madsen + + * main/test.c: main/test.c reports erroneous CLI message. (closes + issue #17051) Reported by: Nick_Lewis + +2010-03-17 14:16 +0000 [r253113] Tilghman Lesher + + * tests/test_gosub.c: Switch to using intptr_t, as suggested by + Kevin Fleming on the -dev list + +2010-03-17 00:40 +0000 [r253028-253032] Leif Madsen + + * main/xmldoc.c: Fix a typo. + + * configs/say.conf.sample: Merged revisions 253018 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 + Mar 2010) | 6 lines Add french snipset to say.conf. Add the + french snipset to say.conf. (Closes issue #15799) ........ + +2010-03-17 00:23 +0000 [r252976-253004] Tilghman Lesher + + * tests/test_gosub.c: Argh. + + * configure, include/asterisk/autoconfig.h.in, tests/test_gosub.c, + configure.ac: Fix bamboo compile error by calculating an integer + with the same size as a pointer. + + * tests/test_gosub.c (added), apps/app_stack.c: Mask out previous + arguments on each nested invocation of Gosub. (closes issue + #16758) Reported by: wdoekes Patches: + 20100316__issue16758.diff.txt uploaded by tilghman (license 14) + Review: https://reviewboard.asterisk.org/r/561/ + +2010-03-16 19:36 +0000 [r252849] Russell Bryant + + * tests/test_time.c: Re-enable test_time on non-Linux. + +2010-03-16 19:36 +0000 [r252848] Sean Bright + + * res/res_clialiases.c: Include an extra newline after "Aliased CLI + command" to get back the prompt. The other issue mentioned in + this bug will be more difficult to resolve since we have no idea + (right now) of knowing if the command that is aliased has been + installed yet. (issue #16978) Reported by: jw-asterisk Tested by: + seanbright + +2010-03-16 19:34 +0000 [r252846] Tilghman Lesher + + * tests/test_time.c, include/asterisk/localtime.h, + main/stdtime/localtime.c: Fix test_time on Mac OS X (and other + platforms without inotify) Reviewboard: + https://reviewboard.asterisk.org/r/554/ + +2010-03-16 19:01 +0000 [r252767] Russell Bryant + + * utils/Makefile, /: Merged revisions 252766 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010) + | 6 lines Don't treat warnings as errors for muted. muted + supports OS X, but uses functions marked as deprecated in 10.6. + However, the functions are still supported, so just ignore the + warnings for now and allow the build to proceed. ........ + +2010-03-16 18:48 +0000 [r252762] Leif Madsen + + * configs/extensions.ael.sample: Merged revisions 252761 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) + | 7 lines Additional extensions.ael global variable fixes. Fixing + up a couple more overlapping global variable namespaces shared + with extensions.conf.sample. Also noticed a few of the lines that + were commented out didn't have the closing semi-colon so I added + that as well. (issue #17035) ........ + +2010-03-16 18:40 +0000 [r252760] Tilghman Lesher + + * codecs/gsm/Makefile: OSARCH is not inherited to this directory + +2010-03-16 18:36 +0000 [r252759] Russell Bryant + + * tests/test_time.c: Disable this test on non-Linux for now. + +2010-03-15 22:48 +0000 [r252709] Kevin P. Fleming + + * res/res_fax.c: Improve handling of values supplied to + FAXOPT(ecm). Previously, values that began with whitespace were + silently treated as 'no', and all non-'yes' values were also + treated as 'no'. Now the supplied value is specifically checked + for a 'yes' or 'no' (or equivalent) value, after skipping leading + whitespace. If the value is not valid, then a warning message is + generated. + +2010-03-15 22:14 +0000 [r252627] Russell Bryant + + * channels/chan_sip.c: Tell the RTP engine API about the initial + read and write format. Peer reviewed out-of-band by file. + +2010-03-15 21:55 +0000 [r252623] Sean Bright + + * apps/app_meetme.c: Resolve a crash in SLATrunk when the specified + trunk doesn't exist. Reported by philipp64 in #asterisk-dev. + +2010-03-15 21:51 +0000 [r252619] Tilghman Lesher + + * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions + 252617 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010) + | 2 lines Uh, yeah. Umask. I'm stupid. ........ + +2010-03-15 20:52 +0000 [r252534] Leif Madsen + + * /, configs/extensions.ael.sample: Merged revisions 252533 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) + | 7 lines Update extensions.ael file to not overlap + extensions.conf. Updated the extensions.ael file so the global + variables don't overlap those that we have in extensions.conf + (sample files). This way unexpected things won't happed hopefully + if both pbx_ael and res_config are loaded. (closes issue #17035) + Reported by: pprindeville ........ + +2010-03-15 16:27 +0000 [r252362-252488] Tilghman Lesher + + * codecs/gsm/Makefile: Make the Makefile logic more explicit and + move the Snow Leopard logic down to where it's not executed on + non-Darwin systems. (closes issue #17028) Reported by: pabelanger + Patches: issue17028_20100315.patch uploaded by seanbright + (license 71) 20100315__issue17028.diff.txt uploaded by tilghman + (license 14) Tested by: tilghman, pabelanger + + * channels/chan_sip.c: THIS IS NOT PYTHON. Indentation doesn't + matter, only braces do. (closes issue #17025) Reported by: + smurfix Patches: sip.patch uploaded by smurfix (license 547) + + * /: Recorded merge of revisions 252366 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252366 | tilghman | 2010-03-14 20:39:00 -0500 (Sun, 14 Mar 2010) + | 2 lines Typo ........ + + * Makefile, contrib/init.d/org.asterisk.asterisk.plist (added), /, + main/asterisk.c: Merged revisions 252361 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010) + | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard: + https://reviewboard.asterisk.org/r/551/ ........ + +2010-03-14 17:43 +0000 [r252314] Sean Bright + + * cdr/cdr_sqlite3_custom.c, cel/cel_sqlite3_custom.c: Fix building + CDR and CEL SQLite3 modules. They added a sqlite3_log() function + which was conflicting with our function names. (closes issue + #17017) Reported by: alephlg + +2010-03-14 14:42 +0000 [r252277] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c, + addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h, + configs/chan_ooh323.conf.sample, addons/ooh323c/src/ooh245.h, + addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ootypes.h, + addons/ooh323c/src/ooq931.c: generate roundtrip delay requests + and responses added response to roundtrip delay requests from + opposite side added roundtrip delay request sending to opposite + side after answer, added options for sending request (interval + between request and count of unreplied requests before forced + call hangup) (closes issue #16976) Reported by: vmikhelson + Patches: rtdr-1.6.0-2.patch uploaded by may213 (license 454) + Tested by: vmikhelson, may213 + +2010-03-13 22:21 +0000 [r252229-252241] Russell Bryant + + * main/app.c: Resolve unit test failure that occurred on Mac OSX. + On Linux (glibc), regcomp() does not return an error for an empty + string. However, the version on OSX will return an error. The + test for channel group matching by regex now passes on the mac, + as well. + + * tests/test_time.c: Resolve compiler warning by paying attention + to system() return value. This resolves the last compile failure + on bamboo. + +2010-03-12 23:18 +0000 [r252133] Tilghman Lesher + + * tests/test_time.c (added): Test script to verify that timezone + cache is properly removed on zonefile alteration. + +2010-03-12 22:04 +0000 [r252089] Terry Wilson + + * main/channel.c, res/res_rtp_asterisk.c, addons/chan_ooh323.c, + main/rtp_engine.c, channels/chan_sip.c, channels/chan_skinny.c, + channels/chan_h323.c, configs/sip.conf.sample, + include/asterisk/frame.h, include/asterisk/rtp_engine.h, + channels/sip/include/sip.h, channels/chan_mgcp.c: Only change the + RTP ssrc when we see that it has changed This change basically + reverts the change reviewed in + https://reviewboard.asterisk.org/r/374/ and instead limits the + updating of the RTP synchronization source to only those times + when we detect that the other side of the conversation has + changed the ssrc. The problem is that SRCUPDATE control frames + are sent many times where we don't want a new ssrc, including + whenever Asterisk has to send DTMF in a normal bridge. This is + also not the first time that this mistake has been made. The + initial implementation of the ast_rtp_new_source function also + changed the ssrc--and then it was removed because of this same + issue. Then, we put it back in again to fix a different issue. + This patch attempts to only change the ssrc when we see that the + other side of the conversation has changed the ssrc. It also + renames some functions to make their purpose more clear. Review: + https://reviewboard.asterisk.org/r/540/ + +2010-03-12 21:57 +0000 [r252088] Moises Silva + + * channels/chan_dahdi.c: add missing mfcr2_skip_category setting + +2010-03-12 19:43 +0000 [r251989] Tilghman Lesher + + * apps/app_voicemail.c: Don't override a user option with the + global option. (closes issue #16849) Reported by: ip-rob Patches: + 20100311__issue16849.diff.txt uploaded by tilghman (license 14) + Tested by: ip-rob + +2010-03-12 19:40 +0000 [r251946-251987] Richard Mudgett + + * /: Merged revisions 251986 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12 Mar 2010) + | 1 line Make chan_dahdi wakeup_sub() prototype not conditional. + ........ + + * channels/chan_dahdi.c: Doxegen this chan_dahdi lock. + +2010-03-11 21:07 +0000 [r251877-251884] Tilghman Lesher + + * apps/app_exec.c: Because ExecIf needs to reprocess arguments, + it's best if we don't remove quotes during parsing. (closes issue + #16905) Reported by: ip-rob Patches: + 20100303__issue16905.diff.txt uploaded by tilghman (license 14) + Tested by: ip-rob + + * tests/test_stringfields.c: Fix tests on 32-bit systems. + + * apps/app_system.c: If the argument to the system application is + quoted, ensure we remove the quotes before trying to execute. + (closes issue #16842) Reported by: ip-rob Patches: + 20100310__issue16842.diff.txt uploaded by tilghman (license 14) + Tested by: ip-rob + +2010-03-11 18:07 +0000 [r251821] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c: Minor tweaks and + comment updates to chan_dahdi. + +2010-03-11 07:03 +0000 [r251779] Alec L Davis + + * apps/app_directory.c: Add supporting code for app-directory pause + option. Since 1.6.1 CLI help reports that option p(n) 'initial + pause' is available. Supporting code was never implemented. + (closes issue #16751) Reported by: alecdavis Patches: + directory_pause.trunk.diff.txt uploaded by alecdavis (license + 585) Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/481/ + +2010-03-10 23:15 +0000 [r251736] Jeff Peeler + + * tests/test_stringfields.c (added), main/utils.c: Add new unit + test for stringfields. (Copied from reviewboard) Tests the + following: 1. Basic allocation and setting of string fields. 2. + Shrinking a string field and re-expanding it. 3. Growing the last + allocation in a string field pool. 4. Setting a string to a large + value such that a new string field pool must be allocated. In + each part, we make sure that the string field is accurate (has + the correct value in it), make sure that the 2 bytes before the + string field has the correct capacity for the field, and for + tests 2-4, we make sure that the string field is where we expect + it to be in memory. Also tested: 5. Shrinking a string field and + partially re-expanding it. 6. Setting strings in such a way as to + create three separate string field pools and then removing the + middle pool. There is a bug fix in the init function, which + ensures the embedded_pool is set to NULL which is important for + stack allocated structures. Review: + https://reviewboard.asterisk.org/r/185/ + +2010-03-10 20:54 +0000 [r251682] Tilghman Lesher + + * funcs/func_strings.c: Hmmm, apparently needed to be fixed in + trunk, too. (closes issue #16900) Reported by: bluecrow76 + Patches: asterisk-1.6.2.4-func_strings.diff uploaded by + bluecrow76 (license 270) + +2010-03-10 20:53 +0000 [r251680] Leif Madsen + + * apps/app_record.c: Be less ambiguous in Record() app docs. For + some reason the documentation for the 'k' application in trunk + and 1.6.2 is different than 1.6.0 and 1.6.1, so I'm setting them + all to match. The wording in 1.6.2 and trunk was ambiguous, so + you could interpret the wording the mean that recording would + continue upon hangup indefinitely, or you could interpret it to + mean that the recorded data would not be discarded upon hangup. + This change makes it clear we mean the latter, and not the + former. Came from a discussion in #asterisk on IRC. + +2010-03-10 20:51 +0000 [r251679] Jeff Peeler + + * main/features.c: Fix ParkAndAnnounce not respecting parking + options. The patch ensures that if a peer does not exist, parking + settings are read from the channel. A unit test has been written + to ensure proper operation for both standard parking and parking + using masquerades. (closes issue #16592) Reported by: mwyres + Patches: bug_16592.diff uploaded by snuffy (license 35) Review: + https://reviewboard.asterisk.org/r/539/ + +2010-03-10 20:30 +0000 [r251677] Tilghman Lesher + + * tests/test_substitution.c, funcs/func_strings.c: It's amazing + what writing a test will find. (issue #16900) Reported by: + bluecrow76 + +2010-03-10 18:25 +0000 [r251631] Jeff Peeler + + * main/abstract_jb.c: Fix jitterbuffer logging not creating + logfiles. Three changes made here: 1) Do not fail if a previous + log does not exist (in fact, this is probably expected). 2) + Ensure that the file descriptor to write to gets assigned + properly. I am at a loss as to why assigning safe_fd outside the + if fixes this, but it makes the if statement slightly less + complicated anyway. 3) Move up the failure message so that the + errno of the failure is not overwritten by fclose. (closes issue + #16917) Reported by: Artem + +2010-03-10 16:55 +0000 [r251538-251585] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h, channels/sig_pri.c: Simplified + dahdi_request() channel selection failed reason/cause code. Also + avoid potential crash because cause could be NULL. + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Reduce the amount of database access for + HAVE_PRI_SERVICE_MESSAGES. Rework HAVE_PRI_SERVICE_MESSAGES to + not use the active values directly from the database. Database + access is likely expensive. Database access now only happens on + initialization, destruction, and when the B channel is taken in + or out of service. This change is not related to call waiting but + it would cause the search for a call waiting interface to be very + expensive and slow down D channel message servicing. + +2010-03-09 20:30 +0000 [r251475] Tilghman Lesher + + * codecs/gsm/Makefile, Makefile.rules: Build system modifications + to ensure that Asterisk properly builds on Mac OS X 10.6. (closes + issue #16997) Reported by: jquinn Patches: + 20100309__issue16997__2.diff.txt uploaded by tilghman (license + 14) Tested by: tilghman, russell + +2010-03-08 18:08 +0000 [r251310] Leif Madsen + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 251309 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r251309 | lmadsen | 2010-03-08 12:07:44 -0600 (Mon, 08 Mar 2010) + | 13 lines Fix Debian init script to not use -c. When using the + init script as-is currently, it could cause issues on Debian such + as high CPU usage. This fix has worked for several people so I'm + implementing the change. (closes issue #16784) Reported by: + pabelanger Tested by: pabelanger, mnick, davidw, mutineer612 + (closes issue #16887) Reported by: jlpedrosa Tested by: + jlpedrosa, mutineer612 ........ + +2010-03-08 05:15 +0000 [r251262-251263] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/stdtime/localtime.c: Remove portions that weren't meant to + be committed for the OS X compat fix + + * funcs/func_pitchshift.c, configure, + include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, + main/stdtime/localtime.c: Change needed to make Mac OS X 10.6 + happy + +2010-03-07 14:53 +0000 [r251221-251222] Michiel van Baak + + * channels/chan_skinny.c: Clean transmit_* for start/stop media + transmission Small patch changing skinny_set_rtp_peer to use + transmit_stopmediatransmission and to use new + transmit_startmediatransmission. Basic testing on 30VIP's by + wedhorn Basic testing on 7960 by me (closes issue #16956) + Reported by: wedhorn Patches: skinny-clean05b.diff uploaded by + wedhorn (license 30) Tested by: wedhorn,mvanbaak + + * channels/chan_skinny.c: Cleanup transmit_callstate handling Broke + the various functions included in transmit_callstate to their own + functions. Transmit_callstate now just transmits callstate. + Generally left the functionality as it was, which highlight some + minor code issues (eg multiple transmit_callstate's). I did + however revise the hint code usage of the old transmit_callstate + as it it not appropriate to put a device on hook based on the + change of a hinted device. (closes issue #16939) Reported by: + wedhorn Patches: skinny-clean04.diff uploaded by wedhorn (license + 30) Tested by: mvanbaak,wedhorn + +2010-03-07 00:45 +0000 [r251181] Alexandr Anikin + + * addons/ooh323c/src/ooq931.c: small log issue from bug 0016664 + +2010-03-06 14:16 +0000 [r251137] Russell Bryant + + * channels/chan_sip.c: Fix a crash in SIP blind transfer handling + found by an automated external test. The first real test added to + the external test suite found a pretty nasty crash that occurred + in Asterisk trunk. The crash was due to a race condition between + the REFER handling and channel destruction in the channel thread. + After the transfer has been completed, we go back to the + transferrer channel and try to lock it so we can fire off a CEL + event. However, there was no guarantee that the channel was still + around at that point since it's racing against the channel + thread. Since ast_channel is a reference counted object, the fix + is simple. The code unlocks the transferrer channel before + finally completing the transfer with an async goto. At this point + the channel thread is going to start call tear down and the + channel will eventually be destroyed. To ensure that the channel + is valid when we want to fire off the CEL event, increase the + channel's reference count. + +2010-03-05 21:51 +0000 [r251038-251087] David Vossel + + * funcs/func_pitchshift.c: fixes xml error in func_pitchshift + + * funcs/func_pitchshift.c (added), CHANGES: PITCH_SHIFT dialplan + function The PITCH_SHIFT function can be used on a channel to + independently modify the pitch of both rx and tx audio streams. + Now you can improve your conference calls by assigning a random + pitch effect to everyone entering a meetme room, or just make + your day more interesting by making your co-workers sound funny. + These are just some of the numerious practical uses for this + function. Enjoy! https://reviewboard.asterisk.org/r/526/ + +2010-03-05 19:32 +0000 [r251022] Russell Bryant + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, + pbx/pbx_gtkconsole.c (removed): Remove pbx_gtkconsole and related + gtk1 checks. Review: https://reviewboard.asterisk.org/r/541/ + +2010-03-05 19:10 +0000 [r250979] Jeff Peeler + + * apps/app_followme.c: Fix app_followme playing wrong sound files. + Fixes regression introduced in 140167 that uses the wrong + variable names. (closes issue #16930) Reported by: ianc Patches: + fix_reload_followme.diff uploaded by ianc (license 998) + +2010-03-05 05:03 +0000 [r250917] Russell Bryant + + * channels/chan_sip.c: Fix up some of chan_sip's usage of the RTP + engine API. The get_local_address() function for an RTP instance + was used when building an SDP, but the results were not honored. + The RTP engine activate() function was not being used once we + have determined that media will now flow. + +2010-03-05 04:37 +0000 [r250913] Tilghman Lesher + + * apps/app_voicemail.c: Missing quote in ODBC query. (closes issue + #16953) Reported by: elguero Patches: + app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license + 37) + +2010-03-05 02:07 +0000 [r250871] Russell Bryant + + * include/asterisk/rtp_engine.h: Fix up the ast_rtp_property enum. + The mis-placement of the latest entry meant that when it was set, + it was writing one index past the end of the properties array in + the ast_rtp_instance (which happened to be the local_address + field). + +2010-03-05 01:05 +0000 [r250787] Jeff Peeler + + * /, res/res_musiconhold.c: Merged revisions 250786 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r250786 | jpeeler | 2010-03-04 19:02:58 -0600 (Thu, 04 + Mar 2010) | 9 lines Fix not being able to specify a URL in MOH + class directory. Don't attempt to chdir on a URL! (closes issue + #16875) Reported by: raarts Patches: moh-http.patch uploaded by + raarts (license 937) ........ + +2010-03-04 20:12 +0000 [r250730] Mark Michelson + + * funcs/func_channel.c: Adjust XML for func_channel to indicate + that rtpdest can take a "text" argument. + +2010-03-03 21:28 +0000 [r250609-250614] Leif Madsen + + * /: Recorded merge of revisions 250613 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r250613 | lmadsen | 2010-03-03 16:28:02 -0500 (Wed, 03 Mar 2010) + | 11 lines Update existing Local channel documentation. A + complete re-write of the Local channel documentation has been + performed, with the existing information from localchannel.txt + and localchannel.tex merged in. (issue #16637) Reported by: kobaz + Patches: localchannel.tex uploaded by lmadsen (license 10) + localchannel.txt uploaded by lmadsen (license 10) Tested by: + lmadsen, jsmith, mmichelson ........ + + * doc/tex/localchannel.tex: Update existing Local channel + documentation. A complete re-write of the Local channel + documentation has been performed, with the existing information + from localchannel.txt and localchannel.tex merged in. (closes + issue #16637) Reported by: kobaz Patches: localchannel.tex + uploaded by lmadsen (license 10) localchannel.txt uploaded by + lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson + +2010-03-03 19:38 +0000 [r250565] Richard Mudgett + + * apps/app_dial.c, channels/chan_dahdi.c, main/dial.c, + channels/chan_local.c, include/asterisk/channel.h, + apps/app_queue.c: Removed cdrflags from ast_channel structure. + Only chan_dahdi set a value in cdrflags. Everyone else just + copied it around the system. Noone cared about any value it may + have contained. + +2010-03-03 19:06 +0000 [r250481] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 250480 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) + | 15 lines Make sure to clear red alarm after polarity reversal. + From the issue: The automatic overnight line tests (or manual + ones) used on UK (BT) lines causes a red alarm on a dahdi / + TDM400P connected channel. This is because the line uses voltage + tests (battery loss) and polarity reversal. The polarity reversal + causes chan_dahdi to initiate v23 CallerID processing but during + this the event DAHDI_EVENT_NOALARM is ignored so that the alarm + is never cleared. (closes issue #14163) Reported by: jedi98 + Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license + 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ + +2010-03-03 19:02 +0000 [r250395-250478] David Vossel + + * main/test.c: Changes 0ms to <1ms in cli END results during 'test + execute' + + * /, channels/chan_iax2.c: Merged revisions 250394 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 + Mar 2010) | 16 lines fixes problem with duplicate TXREQ packets + When Asterisk receives an IAX2 TXREQ packet, try_transfer() will + call store_by_transfercallno() to link the chan_iax2_pvt struct + into iax_transfercallno_pvts. If a duplicate TXREQ packet is + received for the same call, the pvt struct will be linked into + iax_transfercallno_pvts multiple times. This patch fixes this. + Thanks rain for debugging this and providing a patch! (closes + issue #16904) Reported by: rain Patches: + iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested + by: rain, dvossel ........ + +2010-03-03 17:37 +0000 [r250392] Jeff Peeler + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES: + Add new config option to control AMI alarm event reporting in + chan_dahdi. New config parameter "reportalarms" added in + chan_dahdi.conf which supports the following possible values: + "channels": report each channel alarms (current behavior, default + for backward compatibility) "spans": report an "SpanAlarm" event + when the span of any configured channel is alarmed "all": report + channel and span alarms (aggregated behavior) "none": do not + report any alarms (closes issue #16709) Reported by: nahuelgreco + Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco + (license 162) + +2010-03-03 16:43 +0000 [r250303-250346] Tilghman Lesher + + * main/editline/configure: One more fix to editline + + * main/editline/configure, main/editline/Makefile.in, + main/editline/sys.h, main/editline/configure.in: Eliminate + remaining libedit warnings (shown in bamboo) + +2010-03-03 15:39 +0000 [r250302] Matthew Nicholson + + * res/res_fax.c, apps/app_fax.c, CHANGES, res/res_fax_spandsp.c: + Updated CHANGES file to mention res_fax and res_fax_spandsp. Also + fixed MODULEINFO depends and conflicts for app_fax, res_fax, and + res_fax_spandsp. + +2010-03-03 00:18 +0000 [r250235-250246] David Vossel + + * channels/chan_sip.c: fixes signed to unsigned int comparision + issue for FaxMaxDatagram value. + + * main/test.c: fixes assumption that test failed if it did not pass + when generating results + + * tests/test_utils.c: base64 unit test + +2010-03-02 23:22 +0000 [r250190-250213] Matthew Nicholson + + * configs/res_fax.conf.sample (added), include/asterisk/res_fax.h + (added): Merge missed files from res_fax/res_fax_spandsp merge. + + * res/res_fax.c (added), res/res_fax.exports (added), + include/asterisk/frame.h, res/res_fax_spandsp.c (added): Merge + res_fax and res_fax_spandsp. + +2010-03-02 21:58 +0000 [r250141] David Vossel + + * apps/app_directed_pickup.c, CHANGES: adds 'p' option to + PickupChan The 'p' option allows the PickupChan app to pickup a + ringing phone by looking for the first match to a partial channel + name rather than requiring a full match. (closes issue #16613) + Reported by: syspert Patches: pickipbycallid.patch uploaded by + syspert (license 938) pickupbycallerid_v2.patch uploaded by + dvossel (license 671) Tested by: dvossel, syspert + +2010-03-02 21:09 +0000 [r249950-250051] Leif Madsen + + * doc/tex/imapstorage.tex: Update IMAP documentation. Update the + IMAP documentation to make it clear that storing voicemails in + the same folder as a large number of emails could potentially + cause significant slow downs when writing or retrieving + voicemails. (issue #16704) Reported by: TimeHider Tested by: + lmadsen, TimeHider + + * /, configs/cdr.conf.sample: Merged revisions 250043 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 + Mar 2010) | 7 lines Update documentation to clarify purpose of + unanswered option. (closes issue #16267) Reported by: elsto + Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license + 10) Tested by: davidw, elsto ........ + + * /: Recorded merge of revisions 250041 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r250041 | lmadsen | 2010-03-02 15:45:37 -0500 (Tue, 02 Mar 2010) + | 4 lines Update documentation to not imply we support overriding + options. (issue #16855) Reported by: davidw ........ + + * doc/tex/configuration.tex: Update documentation to not imply we + support overriding options. (closes issue #16855) Reported by: + davidw + + * apps/app_directory.c: Fix literal values wrapped in + documentation. (closes issue #16145) Reported by: tilghman + +2010-03-02 19:39 +0000 [r249947] Alec L Davis + + * apps/app_echo.c: revert ability to exit echo app caused a + regression, as only supported VOICE, not VIDEO etc. (issue + #16880) + +2010-03-02 19:24 +0000 [r249912-249925] Leif Madsen + + * main/features.c: Add missing description of the PARKINGLOT + variable in XML documentation. (closes issue #16743) Reported by: + snuffy Patches: parkingdoc.diff uploaded by snuffy (license 35) + + * pbx/pbx_dundi.c: Convert some DUNDI functions to XML + documentation. (closes issue #16798) Reported by: snuffy Patches: + xml_dundi.diff uploaded by snuffy (license 35) + +2010-03-02 19:08 +0000 [r249893] David Vossel + + * channels/chan_unistim.c, configs/chan_dahdi.conf.sample, + configs/console.conf.sample, channels/chan_local.c, + channels/chan_sip.c, configs/oss.conf.sample, + configs/usbradio.conf.sample, configs/misdn.conf.sample, + channels/chan_console.c, channels/chan_gtalk.c, + channels/chan_oss.c, channels/misdn_config.c, + include/asterisk/abstract_jb.h, configs/alsa.conf.sample, + channels/chan_jingle.c, channels/chan_usbradio.c, + channels/chan_dahdi.c, channels/chan_skinny.c, + configs/mgcp.conf.sample, main/abstract_jb.c, + channels/chan_h323.c, channels/chan_alsa.c, + configs/sip.conf.sample, channels/chan_mgcp.c: fixes adaptive + jitterbuffer configuration When configuring the adaptive + jitterbuffer, the target_extra value not only could not be set + from the configuration, but was not even being set to its proper + default. This value is required in order for the adaptive + jitterbuffer to work correctly. To resolve this a config option + has been added to expose this value to the conf files, and a + default value is provided when no config specific value is + present. + +2010-03-02 19:02 +0000 [r249892] Leif Madsen + + * apps/app_osplookup.c, apps/app_confbridge.c, res/res_jabber.c: + Fix several XML documentation validate errors. + +2010-03-02 18:31 +0000 [r249889-249891] Jeff Peeler + + * apps/app_voicemail.c: fix build by checking result of symlink in + test_voicemail_vmsayname + + * CHANGES, apps/app_voicemail.c: Add new application VMSayName for + use with voicemail. VMSayName that will play the recorded name of + the voicemail user if it exists, otherwise will play the mailbox + number. A unit test has been written to verify correct + functionality called test_voicemail_vmsayname. (closes issue + #14973) Reported by: ghjm Review: + https://reviewboard.asterisk.org/r/530/ + +2010-03-02 07:38 +0000 [r249759-249801] Alec L Davis + + * apps/app_echo.c: fixes ability to exit echo app when called from + a ISDN channel, null frames prevent '#' exit. Now only echo back + VOICE and DTMF frames (issue #16880) Reported by: alecdavis + Patches: echo_exit.diff.txt uploaded by alecdavis (license 585) + Tested by: alecdavis + + * channels/chan_dahdi.c: fix asterisk setting of pritimers from + chan_dahdi.conf regression since sig_pri split. (issue #16909) + Reported by: alecdavis Patches: pritimer.asterisk.diff.txt + uploaded by alecdavis (license 585) Tested by: alecdavis + +2010-03-01 19:36 +0000 [r249672] Sean Bright + + * /, apps/app_voicemail.c: Merged revisions 249671 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, + 01 Mar 2010) | 11 lines Fix crash in app_voicemail related to + message counting. We were passing a 'struct inprocess **' and + treating it like a 'struct inprocess *' causing a segfault. + (closes issue #16921) Reported by: whardier Patches: + 20100301_issue16921.patch uploaded by seanbright (license 71) + Tested by: whardier ........ + +2010-03-01 19:33 +0000 [r249669-249670] Michiel van Baak + + * channels/chan_skinny.c: Cleanup display_*message functions. This + patch splits transmit_displaymessage into + transmit_clear_display_message and transmit_display_message which + better aligns with the skinny protocol. The new + transmit_display_message is not used in the current code, but + will be and so it is commented. Moved handle_datetime from this + function to onhook and offhook functions (display now properly + cleared at the end of a call on 30VIP). Removed skinny debug + messages from inline code as there's an ast_verb in + transmit_clear_display_message. Also, removed commentary that it + was a clear display as it is now apparent from the function name. + Split transmit_displaypromptmessage into display and clear. + (closes issue #16878) Reported by: wedhorn Patches: + skinny-clean02.diff uploaded by wedhorn (license 30) + skinny-clean03.diff uploaded by wedhorn (license 30) + + * channels/chan_skinny.c: fix endianes issues in chan_skinny + (closes issue #16826) Reported by: PipoCanaja Patches: + chan_skinny.c_bigendianPatch_20100218.diff uploaded by PipoCanaja + (license 994) Tested by: wedhorn + +2010-03-01 18:36 +0000 [r249623] Tilghman Lesher + + * apps/app_voicemail.c: Constify a bit of app_voicemail, to make + ODBC and IMAP compile once again. + +2010-03-01 17:11 +0000 [r249538] Jeff Peeler + + * channels/chan_local.c, /: Merged revisions 249536 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 + Mar 2010) | 11 lines Modify queued frames from local channels to + not set the other side to up In this case, attended transfers + were broken due to ast_feature_request_and_dial detecting the + channel being set to up before the answer frame could be read and + therefore failing to mark the channel as ready. This fix is a + regression fix for 244785, which should continue to work properly + as well. (closes issue #16816) Reported by: jamhed Tested by: + jamhed, corruptor ........ + +2010-02-28 20:50 +0000 [r249491] Tilghman Lesher + + * apps/app_voicemail.c: Fix unit test that Alec Davis broke. + (closes issue #16927) Reported by: alecdavis + +2010-02-28 16:36 +0000 [r249449] Alec L Davis + + * apps/app_voicemail.c: make unit test check for NULL folder, which + then defaults to INBOX previous test, gave false level of + assurance that code was healthy. (issue #16927) Reported by: + alecdavis Patches: based on app_voicemail_test.diff.txt uploaded + by alecdavis (license 585) Tested by: alecdavis + +2010-02-28 07:10 +0000 [r249405] Tilghman Lesher + + * include/asterisk/app.h, apps/app_voicemail.c: Properly document + voicemail API documents. Also fix a crash reported via the -dev + list. + +2010-02-27 22:49 +0000 [r249320] Alec L Davis + + * channels/sig_pri.c: overlap receiving: automatically send CALL + PROCEEDING when dialplan starts Following Q.931 5.2.4 When the + user has determined that sufficient call information has been + received the user shall stop T302 and send CALL PROCEEDING to the + network. Previously timeouts were possible if the dialplan took a + long time to issue any response back to the network. Verified + that our local TELCO also does the same. (issue #16789) Reported + by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded + by alecdavis (license 585) Tested by: alecdavis + +2010-02-27 14:08 +0000 [r249235] Kevin P. Fleming + + * /, channels/chan_iax2.c: Merged revisions 249234 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 + Feb 2010) | 1 line add a reference to the now-published IAX2 RFC + ........ + +2010-02-26 18:41 +0000 [r249187] Tilghman Lesher + + * apps/app_voicemail.c: Cleanups to fix bugs in the VM count API + functions. - Urgent voicemails were not attached, because the + attachment code looked in the wrong folder. - Urgent voicemails + were sometimes counted twice when displaying the count of new + messages. - Backends were inconsistent as to which voicemails + each API counted. - Unit tests added to verify behavior in the + future. (closes issue #15654) Reported by: tomo1657 Patches: + 20100225__issue15654.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman (closes issue #16448) Reported by: hevad + Review: https://reviewboard.asterisk.org/r/525/ + +2010-02-26 18:41 +0000 [r249186] David Vossel + + * main/test.c: adds Time field to "test show results" cli command + +2010-02-26 17:13 +0000 [r249101-249105] Mark Michelson + + * main/features.c: Send a manager event when the manager + BridgeAction command is used. (closes issue #16769) Reported by: + syspert Patches: bridgeaction.patch uploaded by syspert (license + 938) + + * /, channels/chan_sip.c: Merged revisions 249100 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb + 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488. + (closes issue #16792) Reported by: vrban Patches: t38_606.patch + uploaded by vrban (license 756) ........ + +2010-02-26 08:45 +0000 [r249009-249058] Russell Bryant + + * cdr/cdr_sqlite3_custom.c, cdr/cdr_syslog.c, cdr/cdr_sqlite.c, + cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_odbc.c, + cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c, + cdr/cdr_tds.c, cdr/cdr_csv.c: formatting tweaks and + constification + + * main/cdr.c: Trim trailing whitespace (to help reduce diff against + cdr-q branch) + + * include/asterisk/cdr.h: Trim trailing whitespace, convert lists + of defines to enums + + * cdr/cdr_sqlite.c: trivial formatting tweak (working on reducing + diff against trunk for cdr-q) + + * cdr/cdr_sqlite3_custom.c: remove include + + * cdr/cdr_csv.c: constification, remove include + + * cdr/cdr_tds.c: Remove unnecessary includes, formatting tweak + + * cdr/cdr_pgsql.c: constification and remove unnecessary include + +2010-02-25 23:09 +0000 [r248952] Jeff Peeler + + * /, res/res_monitor.c: Merged revisions 248860 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) + | 18 lines Ensure that monitor recordings are written to the + correct location (again) This is an extension to 248757. As such + the dialplan test has been extended: exten => 5040, 1, + monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, + dial(sip/5001) exten => 5041, 1, + monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, + dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) + exten => 5042, n, dial(sip/5001) exten => 5043, 1, + monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n, + changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001) + exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n, + changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by + design and emits a warning exten => 5044, n, dial(sip/5001) + ........ + +2010-02-25 22:41 +0000 [r248946] Mark Michelson + + * main/acl.c: Fix incorrect ACL behavior when CIDR notation of "/0" + is used. AST-2010-003 + +2010-02-25 21:22 +0000 [r248861] Tilghman Lesher + + * /, main/asterisk.c: Merged revisions 248859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) + | 15 lines Some platforms clear /var/run at boot, which makes + connecting a remote console... difficult. Previously, we only + created the default /var/run/asterisk directory at install time. + While we could create it in the init script, that would not work + for those who start asterisk manually from the command line. So + the safest thing to do is to create it as part of the Asterisk + boot process. This also changes the ownership of the directory, + because the pid and ctl files are created after we setuid/setgid. + (closes issue #16802) Reported by: Brian Patches: + 20100224__issue16802.diff.txt uploaded by tilghman (license 14) + Tested by: tzafrir ........ + +2010-02-25 18:37 +0000 [r248793] Jeff Peeler + + * /, res/res_monitor.c: Merged revisions 248757 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) + | 15 lines Ensure that monitor recordings are written to the + correct location. Recordings should be placed in the monitor + directory when a non-absolute path is used. Exact dialplan used + for testing: exten => 5040, 1, + monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, + dial(sip/5001) exten => 5041, 1, + monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, + dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) + exten => 5042, n, dial(sip/5001) ABE-2101 ........ + +2010-02-24 22:44 +0000 [r248584-248667] Tilghman Lesher + + * channels/Makefile: Also kill the .i files, or else the build + process will not recreate them, when we change flags. Fixes a + weird symbol problem mmichelson was having in a group branch, but + also applies to trunk. + + * /, main/logger.c, include/asterisk/term.h, main/term.c: Merged + revisions 248582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) + | 7 lines Remove color code sequences from verbose messages that + go to logfiles. (closes issue #16786) Reported by: dodo Patches: + logger2.patch uploaded by dodo (license 989) Tested by: tilghman + ........ + +2010-02-24 06:39 +0000 [r248533-248534] Russell Bryant + + * funcs/func_strings.c: Remove unnecessary warning message, make a + couple of formatting tweaks + + * tests/test_strings.c: Add ASTERISK_FILE_VERSION macro. + +2010-02-23 22:29 +0000 [r248489] Mark Michelson + + * tests/test_strings.c (added): Unit test for ast_str API. Review: + https://reviewboard.asterisk.org/r/517 + +2010-02-23 16:34 +0000 [r248397] David Vossel + + * /, channels/chan_sip.c: Merged revisions 248396 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) + | 9 lines fixes invite with replaces deadlock (closes issue + #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 + uploaded by dvossel (license 671) Tested by: pwalker, dvossel + ........ + +2010-02-22 20:19 +0000 [r248347] Mark Michelson + + * channels/chan_sip.c: Move the REF_DEBUG comment higher in the + include list. Uncommenting the REF_DEBUG definition where it was + in the source resulted in only a small part of the astobj2 + references being logged to a file. Moving this up higher in the + include list causes all references to be logged as they should + be. + +2010-02-22 06:45 +0000 [r248225-248226] Russell Bryant + + * include/asterisk/taskprocessor.h, main/taskprocessor.c: Minor + tweaks to comment blocks and includes. Fix the copyright lines, + tweak doxygen formatting, and remove some unnecessary includes. + + * tests/test_devicestate.c: Tweak copyright and author lines. + +2010-02-21 12:09 +0000 [r248184] Michiel van Baak + + * channels/chan_skinny.c: Cleanup transmit_* functions, part 1 + Break transmit_tone into transmit_start_tone and + transmit_stop_tone as per the skinny protocol. (closes issue + #16874) Reported by: wedhorn Patches: skinny-clean01.diff + uploaded by wedhorn (license 30) + +2010-02-20 22:37 +0000 [r248108] Olle Johansson + + * res/res_rtp_asterisk.c: Improve support for RTCP reports without + report blocks + +2010-02-19 18:38 +0000 [r248003] Moises Silva + + * channels/chan_dahdi.c: mfcr2 issue 0016844 - Fix portability bit + fields and make mfcr2_immediate_accept work again, reported and + patched by korihor + +2010-02-19 17:40 +0000 [r247915] David Vossel + + * channels/chan_sip.c: handle_request_invite revise comment, fix + coding guideline issues I'm working with this code right now + trying to analyze a deadlock. This change is just to clean up a + few things before I make a more complex patch. + +2010-02-19 17:33 +0000 [r247914] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 247910 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 + (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... + .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, + 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing + consistent with other channel technologies. The processing of + DTMF tones on the receiving side of an ISDN channel is + inconsistent with the way it is handled in other channels, + especially DAHDI analog. This causes DTMF tones sent from an ISDN + phone to be doubled at the connected party. We are using the + following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes + Option one is necessary because the asterisk DSP DTMF detection + is better than mISDN's internal DSP. Not as many false positives. + Option two is necessary to transmit DTMF tones end to end when + mISDN channels are connected to SIP channels with out of band + DTMF for example. The symptom is that DTMF tones sent by an ISDN + phone are doubled on the way through asterisk when two mISDN + channels are connected with a Local channel in between or if it + is bridged to an analog channel. The doubling of DTMF tones is + because DTMF is passed inband to asterisk by the mISDN channel + and passed out of band once again after the release of the DTMF + tone. Passing it inband is wrong. Neither an analog channel nor + SIP channel passes DTMF inband if configured to inband DTMF. + Analog and SIP channels filter out the DTMF tones because they + use the voice frames returned by ast_dsp_process. But chan_misdn + passes the unfiltered input voice frames instead. To overcome one + aspect of the problem, the doubling of DTMF tones when two mISDN + channels are directly bridged, someone made an 'optimization', + where in that case the DTMF tone passed out-of-band to the peer + channel is not translated to an inband tone at the transmit side. + This optimization is bad because it does not work in general. For + example, analog channels or mISDN channels when bridged through + an intermediary local channel will generate DTMF tones from + out-of-band information. Also, of course, it must not be done + when there is no inband DTMF available. This patch fixes the + issue. Now chan_misdn will filter the received inband DTMF signal + the same as other channel types. Another change included: No need + to build an extra translation path because ast_process_dsp does + it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 + ................ + +2010-02-18 23:13 +0000 [r247787-247841] Tilghman Lesher + + * res/res_speech.c: Revert an errant part of a previous cleanup, to + fix a memory corruption issue. (closes issue #16368) Reported by: + thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf + (license 955) + + * channels/chan_sip.c: If the peer record is from realtime, it + could be set to 0, due to MySQL not representing NULL well in + integer columns. NULL means the value is not specified for the + column, which normally means the driver uses whatever is the + default value. However, on MySQL, placing a NULL in either a + float or integer column results in a retrieval of the 0 value. + Hence, users get an errant error on load. This patch suppresses + that error and makes the value as if it was not there. Note that + this cannot be done in the realtime driver, because the lack of + difference between NULL and 0 can only be intepreted correctly by + the driver itself. If we did it in the realtime driver, then it + would be effectively impossible to set any realtime field to 0, + because it would act as if the field were unspecified and + possibly take on a different value. (closes issue #16683) + Reported by: wdoekes + +2010-02-18 21:23 +0000 [r247736-247770] David Vossel + + * bridges/bridge_softmix.c: fixes confbridge crash when no timing + module is loaded. (closes issue #16471) Reported by: kjotte + Patches: M16471.diff uploaded by junky (license 177) Tested by: + kjotte, junky + + * apps/app_queue.c: fixes Queue with C option crash (closes issue + #16475) Reported by: okrief Patches: queue_crash.diff uploaded by + dvossel (license 671) + +2010-02-18 19:39 +0000 [r247652] Matthew Nicholson + + * /, main/features.c: Merged revisions 247651 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb + 2010) | 6 lines Copy the calling party's account code to the + called party if they don't already have one. (closes issue + #16331) Reported by: bluefox Tested by: mnicholson ........ + +2010-02-18 18:31 +0000 [r247609] Richard Mudgett + + * main/channel.c: Fix placing ISDN calls on hold preventing native + bridging from being reexamined after a transfer. Consider the + following scenario: /-- B A == * == Network \-- C Party B calls + party A (EuroISDN BRI phone) Party A puts B on hold using the + HOLD/RETRIEVE messages. Party A calls party C. Party A puts C on + hold to talk with party B again. Party A transfers B to C by + hanging up. The call does not get the opportunity to get + re-transferred into the ISDN network by the native bridge because + native bridging is not being reexamined after the initial + transfer. + +2010-02-18 16:54 +0000 [r247503-247509] Leif Madsen + + * /, README-SERIOUSLY.bestpractices.txt: Merged revisions 247508 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010) + | 1 line Add additional link to best practices document per + jsmith. ........ + + * /, README-SERIOUSLY.bestpractices.txt (added): Merged revisions + 247502 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010) + | 10 lines Add best practices documentation. (issue #16808) + Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis + Tested by: lmadsen Review: + https://reviewboard.asterisk.org/r/507/ ........ + +2010-02-18 16:34 +0000 [r247500] Philippe Sultan + + * CHANGES, res/res_jabber.c: Add a new manager event for our + buddies status. The new JabberStatus event gives a concise view + of the status change to the AMI clients. Thanks fiddur! (closes + issue #16760) Reported by: fiddur Patches: 244498.2.diff uploaded + by fiddur (license 678) Tested by: fiddur, phsultan + +2010-02-18 04:20 +0000 [r247423] Russell Bryant + + * Makefile, /, sounds/Makefile: Merged revisions 247422 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010) + | 10 lines Tweak argument handling for wget in the sounds + Makefile. 1) Fix the check to see if we are using wget to not be + full of fail. The configure script populates this variable with + the absolute path to wget if it is found, so it didn't work. 2) + Allow some extra arguments to be passed in for wget. This is just + a simple change to allow our Bamboo build script to tell wget to + be quiet and not fill up our logs with download status output. + ........ + +2010-02-17 22:44 +0000 [r247335-247381] Mark Michelson + + * main/test.c: Fix a couple of bugs in test tab completion. 1. Add + missing unlock of lists. 2. Swap order of arguments to + test_cat_cmp in complete_test_name. + + * main/test.c: Tab completion for test categories and names for + "test show registered" and "test execute" CLI commands. + + * main/strings.c, include/asterisk/strings.h: Fix two problems in + ast_str functions found while writing a unit test. 1. The + documentation for ast_str_set and ast_str_append state that the + max_len parameter may be -1 in order to limit the size of the + ast_str to its current allocated size. The problem was that the + max_len parameter in all cases was a size_t, which is unsigned. + Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the + max_len parameter to be ssize_t fixed this issue. 2. Once issue 1 + was fixed, there was an off-by-one error in the case where we + attempted to write a string larger than the current allotted size + to a string when -1 was passed as the max_len parameter. When + trying to write more than the allotted size, the ast_str's + __AST_STR_USED was set to 1 higher than it should have been. + Thanks to Tilghman for quickly spotting the offending line of + code. Oh, and the unit test that I referenced in the top line of + this commit will be added to reviewboard shortly. Sit tight... + +2010-02-17 19:51 +0000 [r247295] Jeff Peeler + + * funcs/func_groupcount.c, tests/test_app.c (added), main/app.c, + CHANGES: Add support for GROUP_MATCH_COUNT regex matching on + category Current support for regex matching was previously only + available on the group. Also, error reporting for regex failures + has been added. In addition to this feature enhancement a unit + test has been written to check the regular expression logic to + ensure the count operation is working as expected. (closes issue + #16642) Reported by: kobaz Patches: groupmatch2.patch uploaded by + kobaz (license 834) Review: + https://reviewboard.asterisk.org/r/503/ + +2010-02-17 19:23 +0000 [r247248-247282] David Vossel + + * tests/test_devicestate.c: modified device2extension_test's + category + + * tests/test_devicestate.c (added): unit test for combined device + state mapping and device to exten state mapping Review: + https://reviewboard.asterisk.org/r/516/ + + * main/features.c, CHANGES, configs/features.conf.sample: addition + of dynamic parkinglots feature This feature allows for + parkinglots to be created dynamically within the dialplan. Thanks + to all who were involved with getting this patch written and + tested! (closes issue #15135) Reported by: IgorG Patches: + features.dynamic_park.v3.diff uploaded by IgorG (license 20) + 2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7) + dynamic_parkinglot.diff uploaded by dvossel (license 671) Tested + by: eliel, IgorG, acunningham, mvanbaak, zktech Review: + https://reviewboard.asterisk.org/r/352/ + +2010-02-17 16:24 +0000 [r247169] Mark Michelson + + * /, apps/app_queue.c: Merged revisions 247168 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb + 2010) | 3 lines Make sure that when autofill is disabled that + callers not in the front of the queue cannot place calls. + ........ + +2010-02-17 07:01 +0000 [r247124-247125] Tilghman Lesher + + * main/loader.c: RTP documentation states that you can pass NULL as + the module, so make sure that's really the case. + + * channels/sip/include/dialog.h (added), channels/chan_sip.c, + channels/sip/include/config_parser.h, + channels/sip/include/globals.h (added), + channels/sip/dialplan_functions.c (added), channels/Makefile, + channels/sip/include/sip_utils.h, + channels/sip/include/dialplan_functions.h (added): Make all of + the various rtpqos parameters in this branch available from the + CHANNEL function. Also includes a test for retrieving rtpqos + parameters, including a NULL RTP driver. Additionally, some + further separation of the SIP internal API into headers was + necessary. (closes issue #16652) Reported by: kkm Patches: + 20100204__issue16652.diff.txt uploaded by tilghman (license 14) + Review: https://reviewboard.asterisk.org/r/501/ + +2010-02-16 23:44 +0000 [r247076] Mark Michelson + + * main/strings.c: Add va_end calls to __ast_str_helper. According + to the man page for stdarg(3), "Each invocation of va_copy() must + be matched by a corresponding invocation of va_end() in the same + function." There were several cases in __ast_str_helper where + va_copy was not matched with a corresponding call to va_end. + +2010-02-16 22:58 +0000 [r247035] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c: generate + connected line info update from info in h.323 packets Tested by: + benngard + +2010-02-16 21:15 +0000 [r246985] Mark Michelson + + * include/asterisk/strings.h: Add some clarifying documentation to + the ast_str_set and ast_str_append functions. + +2010-02-16 21:03 +0000 [r246980-246981] David Vossel + + * main/tcptls.c: swap openssl with OpenSSL in warning message. + (issue #16673) + + * main/tcptls.c: warning message if openssl support is missing + while attempting tls connection (closes issue #16673) Reported + by: michaesc Patches: tls_error_msg.diff uploaded by dvossel + (license 671) + +2010-02-16 18:29 +0000 [r246942] Mark Michelson + + * tests/test_pbx.c (added): Add unit test for dialplan pattern + matching. This test works by reading input from arrays to build a + sample dialplan. From there, patterns are attempted to be matched + against said dialplan, with the expected match given. We then + search in our example dialplan to see if we find a match and if + what we find matches what we expected it to match. (closes issue + #16809) Reported by: lmadsen Tested by: mmichelson Review: + https://reviewboard.asterisk.org/r/504/ + +2010-02-16 17:07 +0000 [r246899] David Vossel + + * main/channel.c: fixes sample rate conversion issue with Monitor + application When using ast_seekstream with the read/write streams + of a monitor, the number of samples we are seeking must be of the + same rate as the stream or the jump calculation will be + incorrect. This patch adds logic to correctly convert the number + of samples to jump to the sample rate the read/write stream is + using. For example, if the call is G722 (16khz) and the + read/write stream is recording a 8khz wav, seeking 320 samples of + 16khz audio is not the same as seeking 320 samples of 8khz audio + when performing the ast_seekstream on the stream. ABE-2044 + +2010-02-16 15:36 +0000 [r246710-246863] Tilghman Lesher + + * build_tools/cflags.xml, build_tools/cflags-devmode.xml: Revert + changes for now, pending discussion + + * build_tools/cflags-devmode.xml: Add a few more targets for + DEBUG_THREADLOCALS + + * build_tools/cflags.xml, channels/chan_usbradio.c, + build_tools/cflags-devmode.xml, main/strings.c, + apps/app_voicemail.c: Change the blanket rules to delete + .lastclean on all CFLAGS menuselect targets to be more + particular. This change builds upon the recent change to + menuselect to add 'touch_on_change' as an attribute of both + categories and members. This should allow only the most invasive + defines to cause a complete rebuild, while defines which only + affect a subset of modules will only cause a rebuild of that + smaller set. + + * channels/chan_sip.c: Allow Timer B to be set on the peer, and + ensure SIP rules are followed (or warn) in comparison to Timer + T1. (closes issue #16643) Reported by: nahuelgreco Patches: + 20100204__issue16643.diff.txt uploaded by tilghman (license 14) + Tested by: oej + + * Makefile, /: Merged revisions 246709 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010) + | 5 lines Make the menuselect instructions correct by allowing + 'make menuselect' to actually solve dependency problems. + (Previously, it would fail out again with the same message about + running 'make menuselect', which was NOT at all helpful.) + ........ + +2010-02-15 22:08 +0000 [r246669] Richard Mudgett + + * channels/chan_dahdi.c: Restore triedtopribridge flag code removed + in -r211197. Ooops. Failed to note that we were inside a for loop + and pri_channel_bridge() needs to be executed only once. + +2010-02-15 21:37 +0000 [r246667] Tilghman Lesher + + * utils/utils.xml: Instead of just automatically filtering out in + the Makefile, give an indication of dependencies in menuselect. + +2010-02-15 15:45 +0000 [r246627] David Vossel + + * channels/chan_sip.c, channels/sip/reqresp_parser.c, + channels/sip/include/sip_utils.h, + channels/sip/include/reqresp_parser.h: chan_sip parse code + refactoring plus two new unit tests Code Refactoring Changes - + read_to_parts() moved to reqresp_parser.c and has been renamed as + get_name_and_number() - get_in_brackets() moved to + reqresp_parser.c - find_closing_quotes() added to sip_utils.h + Logic Changes - get_name_and_number() now uses parse_uri() and + get_calleridname() for parsing. Before this change only names + within quotes were found, when names not within quotes are + possible. New Unit Tests -sip_get_name_and_number_test + -sip_get_in_brackets_test (closes issue #16707) Reported by: + Nick_Lewis Patches: issue16706.diff uploaded by dvossel (license + 671) Review: https://reviewboard.asterisk.org/r/499/ + +2010-02-12 23:32 +0000 [r246420-246546] David Vossel + + * main/channel.c, /: Merged revisions 246545 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010) + | 16 lines lock channel during datastore removal On channel + destruction the channel's datastores are removed and destroyed. + Since there are public API calls to find and remove datastores on + a channel, a lock should be held whenever datastores are removed + and destroyed. This resolves a crash caused by a race condition + in app_chanspy.c. (closes issue #16678) Reported by: + tim_ringenbach Patches: datastore_destroy_race.diff uploaded by + tim ringenbach (license 540) Tested by: dvossel ........ + + * channels/chan_sip.c: fixes areas where port should be removed + from domain during parsing A patch was committed recently that + converted duplicate uri parsing code to use the parse_uri + function. There were two instances where this conversion did not + mimic previous behavior exactly because the port was not being + parsed off the end of the domain. In order to do this, a dummy + pointer argument needs to be passed into parse_uri so it will + know it must parse out the port from the domain. If a port output + paramenter is not present, the domain is returned with the port + still attached. + +2010-02-12 08:30 +0000 [r246382] TransNexus OSP Development + + * apps/app_osplookup.c, UPGRADE.txt, CHANGES: Updated doc for OSP + lookup application. + +2010-02-11 21:57 +0000 [r246299-246338] David Vossel + + * tests/test_heap.c, tests/test_event.c, + channels/sip/reqresp_parser.c, channels/sip/config_parser.c: + fixes some test description formatting inconsistencies so log + file looks nice + + * tests/test_astobj2.c (added), main/astobj2.c: astobj2 unit test + and bug fix A bug was discovered during the creation of the + astobj2 unit test. When OBJ_MULTIPLE | OBJ_UNLINK is used, the + objects being returned had a ref count issue. This patch resolves + that. Review: https://reviewboard.asterisk.org/r/496/ + +2010-02-10 23:19 +0000 [r246260] Russell Bryant + + * include/asterisk/event.h, tests/test_event.c (added), + main/event.c: Add a test module for the event API, test_event.c. + This module includes a single test so far that creates events + using two different methods and does some verification on the + result to make sure the correct data can be retrieved from the + event that was created. One bug was found in the event API while + developing this test, which makes me happy. :-) Review: + https://reviewboard.asterisk.org/r/495/ + +2010-02-10 23:13 +0000 [r246249] David Vossel + + * channels/sip/reqresp_parser.c, + channels/sip/include/reqresp_parser.h: additional parse_uri test + and documentation + +2010-02-10 21:55 +0000 [r246200-246208] Tilghman Lesher + + * res/res_pktccops.exports (added): res_pktccops needs to be able + to export a symbol for chan_mgcp (closes issue #16782) Reported + by: nahuelgreco Patches: res_pktccops.exports uploaded by + nahuelgreco (license 162) + + * funcs/func_strings.c: Fussy compiler on another machine... + + * funcs/func_strings.c: Fix weird issue with unit tests on + optimized build - turned out to be a signing issue. + +2010-02-10 17:49 +0000 [r246116] David Vossel + + * /, apps/app_queue.c: Merged revisions 246115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) + | 8 lines fixes random deadlock in app_queue with use_weight + during reload (closes issue #16677) Reported by: tim_ringenbach + Patches: app_queue_use_weight_deadlock.diff uploaded by tim + ringenbach (license 540) ........ + +2010-02-10 16:47 +0000 [r246070] Jeff Peeler + + * channels/chan_local.c: Change channel state on local channels for + busy,answer,ring. Previously local channels channel state never + changed. This became problematic when the state of the other side + of the local channel was lost, for example during a masquerade. + Changing the state of the local channel allows for the scenario + to be detected when the channel state is set to ringing, but the + peer isn't ringing. The specific problem scenario is described in + 164201. Although this was noted on one of the issues, here is the + tested dialplan verified to work: exten => + 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten => + *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) + exten => *9700,n,wait(3) ;3 works, 1 did not exten => + *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did + not exten => + 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes + issue #14992) Reported by: davidw + +2010-02-10 16:01 +0000 [r245945-246030] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + res/res_agi.c: Solaris doesn't like outputting a NULL to a %s in + format strings. Detect all platforms that don't like that, + either, and ensure that when documentation is missing, we pass a + non-NULL pointer when outputting the corresponding documentation. + (closes issue #16689) Reported by: bklang Patches: + 20100209__issue16689__with_tests.diff.txt uploaded by tilghman + (license 14) Review: https://reviewboard.asterisk.org/r/497/ + + * funcs/func_strings.c: Enable warnings on atypical conditions for + the FILTER function (suggested by mmichelson on the -dev list). + + * /, funcs/func_strings.c, configs/extensions.conf.sample: Merged + revisions 245944 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) + | 2 lines Include examples of FILTER usage in extension patterns + where a "." may be a risk. ........ + +2010-02-09 23:32 +0000 [r245864] Russell Bryant + + * include/asterisk/test.h, tests/test_sha1.c (removed), + include/asterisk/utils.h, tests/test_substitution.c, + tests/test_heap.c, tests/test_ast_format_str_reduce.c, + tests/test_skel.c, tests/test_utils.c, funcs/func_math.c, + channels/sip/reqresp_parser.c, main/test.c, tests/test_md5.c + (removed), channels/sip/config_parser.c, tests/test_sched.c: + Various updates to the unit test API. 1) It occurred to me that + the difference in usage between the error ast_str and the + ast_test_update_status() usage has turned out to be a bit + ambiguous in practice. In a lot of cases, the same message was + being sent to both. In other cases, it was only sent to one or + the other. My opinion now is that in every case, I think it makes + sense to do both; we should output it to the CLI as well as save + it off for logging purposes. This change results in most of the + changes in this diff, since it required changes to all existing + unit tests. It also allowed for some simplifications of unit test + API implementation code. 2) Update ast_test_status_update() to + include the file, function, and line number for the code + providing the update. 3) There are some formatting tweaks here + and there. Hopefully they aren't too distracting for code review + purposes. Reviewboard's diff viewer seems to do a pretty good job + of pointing out when something is a whitespace change. 4) I moved + the md5_test and sha1_test into the test_utils module. It seemed + like a better approach since these tests are so tiny. 5) I + changed the number of nodes used in heap_test_2 from 1 million to + 100 thousand. The only reason for this was to reduce the time it + took for this test to run. 6) Remove an unused function prototype + that was at the bottom of utils.h. 7) Simplify test_insert() + using the LIST_INSERT_SORTALPHA() macro. The one minor difference + in behavior is that it no longer checks for a test registered + with the same name. 8) Expand the code in test_alloc() to provide + specific error messages for each failure case, to clearly inform + developers if they forget to set the name, summary, description, + etc. 9) Tweak the output of the "test show registered" CLI + command. I swapped the name and category to have the category + first. It seemed more natural since that is the sort key. 10) + Don't output the status ast_str in the "test show results" CLI + command. This is going to tend to be pretty verbose, so just + leave that for the detailed test logs (test generate results). + Review: https://reviewboard.asterisk.org/r/493/ + +2010-02-09 23:18 +0000 [r245793-245804] David Vossel + + * channels/chan_iax2.c: fixes a merging error for the iaxs and + iaxsl off by one fix + + * /, channels/chan_iax2.c: Merged revisions 245792 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 + Feb 2010) | 12 lines Fixes iaxs and iaxsl size off by one issue. + 2^15 = 32768 which is the maximum allowed iax2 callnumber. + Creating the iaxs and iaxsl array of size 32768 means the maximum + callnumber is actually out of bounds. This causes a nasty crash. + (closes issue #15997) Reported by: exarv Patches: iax_fix.diff + uploaded by dvossel (license 671) ........ + +2010-02-09 18:06 +0000 [r245729] Tilghman Lesher + + * apps/app_fax.c: Ensure frames are only freed once. (closes issue + #16361) Reported by: vlad Patches: 20100208__issue16361.diff.txt + uploaded by tilghman (license 14) Tested by: kenny, bloodoff, + misaksen + +2010-02-09 17:40 +0000 [r245727] Matthew Nicholson + + * channels/chan_sip.c: This commit removes an extra newline in T.38 + generated SDP packets. This bug was caused by the fix introduced + in r243860. (closes issue #16766) Reported by: raivisr Patches: + t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96) + Tested by: raivisr + +2010-02-09 16:24 +0000 [r245680] Kevin P. Fleming + + * apps/app_fax.c: Don't offer MMR or JBIG transcoding during T.38 + negotiation. After further discussion with Steve Underwood, we + should not (yet) be offering to receive MMR or JBIG transcoded + streams from T.38 endpoints. A future spandsp release will + support those features, and then they can be enabled during + negotiation + +2010-02-08 23:43 +0000 [r245597-245624] Russell Bryant + + * main/event.c: Fix return value of get_ie_str() and + get_ie_str_hash() for non-existent IE. I found this bug while + developing a unit test for event allocation. Testing is awesome. + + * tests/test_utils.c: UNREGISTER instead of REGISTER in + unload_module(). + + * main/pbx.c: Use memmove() instead of memcpy() for a case where + the buffers overlap. Once again, valgrind is freaking awesome. + That is all. + + * channels/Makefile: Remove object files from the channels/sip/ + directory on make clean. + +2010-02-08 22:31 +0000 [r245578] Tilghman Lesher + + * main/Makefile, channels/Makefile: Actually use _ASTLDFLAGS in the + main/ and channels/ Makefiles. They were previously passed + correctly, but they simply weren't used. This caused issues with + various platforms whose builds needed to pass special linker + flags via the configure script. (closes issue #16596) Reported + by: pprindeville Patches: asterisk-1.6-astldflags.patch uploaded + by pprindeville (license 347) Tested by: tilghman + +2010-02-08 20:41 +0000 [r245497] Jason Parker + + * /, main/ast_expr2f.c, main/ast_expr2.fl: Merged revisions 245496 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) | + 4 lines Remove reference of documentation in source directory. + People don't always build Asterisk from source (distro packages, + anybody?). ........ + +2010-02-08 04:51 +0000 [r245268-245385] Russell Bryant + + * contrib/scripts/install_prereq: Add the libvpb-dev package as a + dependency. + + * pbx/pbx_gtkconsole.c: Add a todo for pbx_gtkconsole for updating + to gtk2. This module needs to be converted to gtk2, or we will + eventually have to just remove it from the tree. gtk1 isn't even + packaged anymore in the distro I'm using. I suspect nobody uses + this and that nobody would notice if we removed it. + + * contrib/scripts/install_prereq: Add more packages required for + building Asterisk modules. + + * channels/chan_usbradio.c: Make chan_usbradio compile. + + * tests/test_sha1.c (added): Add a SHA1 test module. Review: + https://reviewboard.asterisk.org/r/492/ + + * tests/test_md5.c: Remove unnecessary include, ast_md5_hash() + comes from utils.h. + + * tests/test_md5.c (added): Add an MD5 test module. Review: + https://reviewboard.asterisk.org/r/491/ + + * tests/test_ast_format_str_reduce.c: Fix a couple of spelling + errors, and add format module dependencies. + + * channels/sip/include/config_parser.h, channels/sip/include/sip.h, + channels/sip/include/sip_utils.h, + channels/sip/include/reqresp_parser.h: Tweak formatting and add + minor updates to some comments. + + * main/test.c: Remove an extra space. + +2010-02-07 19:51 +0000 [r245230] Mark Michelson + + * channels/chan_sip.c: Remove parsing of constantssrc from + reload_config. This config option is already handled by the + function handle_common_options and it is unnecessary to parse the + value again. + +2010-02-06 14:43 +0000 [r245192] Mark Michelson + + * channels/chan_sip.c, configs/sip.conf.sample: Remove useless sip + options related to hash table size. First off, these options + weren't actually doing anything. By the time the options were + parsed, the peer and dialog containers had already been allocated + with their default values. Second, hash table size is something + that doesn't really make sense to change in a config file. If a + user is that interested in changing the hashtable size, he can + modify the source itself. I have removed the parsing of the + hash_peer, hash_user, and hash_dialog options. I have removed the + hash_user_size variable altogether since it is not used at all. I + also changed hash_peer_size and hash_dialog_size to be constant, + and have changed the symbols to be in all caps as constants + typically are. I have also removed the entire section in + sip.conf.sample regarding configurable hashtable sizes. + +2010-02-05 21:21 +0000 [r245147] David Vossel + + * include/asterisk/astobj2.h, main/astobj2.c: fixes astobj2 + unlinking of multiple objects when OBJ_MULTIPLE was disabled When + OBJ_MULTIPLE was off but OBJ_UNLINK was on, all the items in a + bucket were being unlinked instead of just the first match. This + fixes that. Review: https://reviewboard.asterisk.org/r/490/ + +2010-02-05 19:26 +0000 [r245090] Jeff Peeler + + * /, LICENSE, contrib/firmware (removed): Merged revisions 245044 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb + 2010) | 5 lines Remove contrib/firmware directory as it is empty + Remove explicit license for IAXy firmware as it is no longer + included in the tree ........ + +2010-02-05 19:07 +0000 [r245046] Tilghman Lesher + + * tests/test_ast_format_str_reduce.c, main/file.c: Merge tests that + verify the same thing. (Oops.) + +2010-02-05 18:12 +0000 [r245006] David Vossel + + * channels/chan_iax2.c: adds total call numbers available to 'iax2 + show callnumber usage' cli output + +2010-02-05 17:20 +0000 [r244945] Terry Wilson + + * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, + res/res_calendar_caldav.c: Fix crash on 32-bit for users not + using https (closes issue #16778) Reported by: pitel Patches: + diff.txt uploaded by twilson (license 396) Tested by: twilson, + pitel + +2010-02-05 17:05 +0000 [r244927] Sean Bright + + * /, main/asterisk.c: Merged revisions 244926 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb + 2010) | 1 line Update main copyright date. ........ + +2010-02-05 16:59 +0000 [r244769-244924] David Vossel + + * channels/chan_sip.c, channels/sip/include/config_parser.h, + channels/sip/config_parser.c: fixes issue with sip registry not + having correct default expiry default expiry was not being set + correctly for a registry object. Thanks to ebroad for reporting + the issue and testing the patch. + + * main/astobj2.c: fixes memory leak in astobj2 test + ao2_iterator_destroy was not being used on the iterator during + the test. This resulted in the container never actually being + destroyed. + + * channels/chan_sip.c: parse_moved_contact tries to parse + contact_name twice parse_moved_contact attempts to remove a + quoted string twice, and the first try wasn't even being done + correctly. + +2010-02-04 22:43 +0000 [r244728-244768] Tilghman Lesher + + * main/file.c: Try to make ast_format_str_reduce fail... + + * include/asterisk/manager.h: Oops + + * include/asterisk/manager.h: Define a small set of constant return + values + +2010-02-04 15:36 +0000 [r244688] David Vossel + + * main/test.c: fix truncated format string in 'test show + registered' When using the 'test show registered' cli command the + 'Test Results' category was truncating the last few characters + making it look like 'Test Resul'. I also expanded other parts of + the format to better represent how long function names and + categories will likely be. + +2010-02-04 00:12 +0000 [r244647] Richard Mudgett + + * channels/sip: Add ignore *.i files property to the new + channels/sip directory. + +2010-02-03 20:48 +0000 [r244598] Jeff Peeler + + * main/features.c, CHANGES: Add some additional option support for + non-default parking lots. The options are: parkedcallparking, + parkedcallhangup, parkedcallrecording, and parkedcalltransfers. + Previously these options were only available for the default + parking lot. (closes issue #16641) Reported by: bluecrow76 + Patches: asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76 + (license 270) + +2010-02-03 20:33 +0000 [r244597] David Vossel + + * channels/chan_sip.c, channels/sip/include/config_parser.h + (added), channels/sip/reqresp_parser.c (added), channels/sip + (added), channels/Makefile, channels/sip/config_parser.c (added), + channels/sip/include (added), channels/sip/include/sip.h (added), + channels/sip/include/sip_utils.h (added), + channels/sip/include/reqresp_parser.h (added): -----Changes ----- + New files - channels/sip/sip.h – A new header for shared #define, + enum, and struct definitions. - channels/sip/include/sip_utils.h + – sip util functions shared among the all the sip APIs - + channels/sip/include/config_parser.h – sip config-parser API - + channels/sip/config_parser.c – Contains sip.conf parsing helper + functions with unit tests. - + channels/sip/include/reqresp_parser.h – sip request response + parser API - channels/sip/reqresp_parser.c – Contains sip request + and response parsing helper functions with unit tests. New Unit + Tests - sip_parse_uri_test - sip_parse_host_test - + sip_parse_register_line_test Code Refactoring - All reusable + #define, enum, and struct definitions were moved out of + chan_sip.c into sip.h. During this process formatting changes + were made to comments in both sip.h and chan_sip.c in order to + better adhere to the coding guidelines. - The beginnings of three + new sip APIs, sip-utils.h, config-parser.h, reqresp-parser.h + using existing chan_sip.c functions. - parse_uri() and + get_calleridname() were moved from chan_sip.c to request-parser.c + along with unit tests for both functions. - sip_parse_host() and + sip_parse_register_line() were moved from chan_sip.c to + config-parser.c along with unit tests for both functions. Changes + to parse_uri() -removal of the options parameter. It was never + used and did not behave correctly. -additional check for + [?header] field. When this field was present, the transport type + was not being set correctly. ----- Overview ----- This patch is + introduced with the hope that unit tests for all our sip parsing + functions will be written soon. chan_sip is a huge file, and with + the addition of each unit test chan_sip is going to grow larger + and harder to maintain. I'm proposing we begin refactoring + chan_sip, starting with the parsing functions. With each parsing + function we move into a separate helper file, a unit test should + accompany it. I've attempted to lay down the ground work for this + change by creating two new parser helper files (config-parser.c + and reqresp-parser.c) and moving all shared structs, enums, and + defines from chan_sip.c into a shared sip.h file. We can't verify + everything in Asterisk using unit tests, but string parsing is + one area where unit tests make the most sense. By beginning to + restructure the code in this way, chan_sip not only becomes less + bloated, but Asterisk as a whole will become more stable. Review: + https://reviewboard.asterisk.org/r/477/ + +2010-02-03 19:26 +0000 [r244547] Mark Michelson + + * main/sched.c: Initialize counters in ast_sched_report so that + resulting data is not bogus. + +2010-02-03 18:34 +0000 [r244505] Tilghman Lesher + + * channels/chan_dahdi.c: The chanvar= setting should inherit the + entire list of variables, not just the first one. (closes issue + #16359) Reported by: raarts Patches: dahdi-setvars.diff uploaded + by raarts (license 937) Tested by: raarts + +2010-02-02 22:27 +0000 [r244443] David Vossel + + * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h: + fixes crash during T.38 negotiation caused by invalid or missing + FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported + by: krn (closes issue #16724) Reported by: barthpbx (closes issue + #16517) Reported by: bklang (closes issue #16485) Reported by: + elsto + +2010-02-02 20:32 +0000 [r244071-244393] Tilghman Lesher + + * apps/app_dial.c, CHANGES: Properly respect GOSUB_RESULT as to + what to do with the master channel. Previously, we would parse + GOSUB_RESULT, but not actually do anything with it. Also, allow + GOSUB_RETVAL to be inherited back across a peer/master channel. + (closes issue #16687) Reported by: bklang Patches: + app_dial-preserve-gosub_retval.patch uploaded by bklang (license + 919) (with modifications) (closes issue #16686) Reported by: + bklang Patches: app_dial-respect-gosub_result.patch uploaded by + bklang (license 919) (with modifications) + + * funcs/func_math.c: Correct some off-by-one errors, especially + when expressions don't contain expected spaces. Also include the + tests provided by the reporter, as regression tests. (closes + issue #16667) Reported by: wdoekes Patches: + astsvn-func_match-off-by-one.diff uploaded by wdoekes (license + 717) + + * /, apps/app_voicemail.c: Merged revisions 244242 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r244242 | tilghman | 2010-02-01 17:13:44 -0600 (Mon, 01 + Feb 2010) | 11 lines Backup and restore original textfile, for + prosthesis (gerund of prepend). Also, fix menuselect such that + changing voicemail build options correctly causes rebuild. + (closes issue #16415) Reported by: tomo1657 Patches: + prepention.patch uploaded by tomo1657 (license 484) (with + modifications by me to backport to 1.4) ........ + + * main/channel.c, channels/chan_local.c, /: Merged revisions 244070 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r244070 | tilghman | 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010) + | 16 lines Revert previous chan_local fix (r236981) and fix + instead by destroying expired frames in the queue. (closes issue + #16525) Reported by: kobaz Patches: 20100126__issue16525.diff.txt + uploaded by tilghman (license 14) + 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman + (license 14) Tested by: kobaz, atis (closes issue #16581) + Reported by: ZX81 (closes issue #16681) Reported by: alexr1 + ........ + +2010-01-28 22:37 +0000 [r243986] Jeff Peeler + + * main/manager.c: Optimization to manager events. When potentially + sending manager events, return immediately if there are no + sessions or hooks. Also, avoid locking the hooks list if it is + empty. (issue #16455) Reported by: atis Patches: + manager_hooks_trunk.patch uploaded by atis (license 242) + +2010-01-28 20:00 +0000 [r243943] Tilghman Lesher + + * channels/iax2-parser.c: Informational message, not an error. + +2010-01-28 18:35 +0000 [r243780-243860] Russell Bryant + + * channels/chan_sip.c: Add a missing line terminator for T.38 SDP. + + * /, channels/chan_sip.c: Merged revisions 243779 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r243779 | russell | 2010-01-28 09:03:17 -0600 (Thu, 28 Jan 2010) + | 2 lines Fix a bogus third argument to ast_copy_string(). + ........ + +2010-01-27 20:37 +0000 [r243551-243693] Jeff Peeler + + * /, apps/app_queue.c: Merged revisions 243691 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010) + | 5 lines Revert 243570, I should have looked at this closer. + Will reopen the issue, but am leaving the review closed as the + change was pointless. (issue #16488) ........ + + * CHANGES: expand code based appreviation of AST_CONFIG_DIR to + configuration directory + + * /, apps/app_queue.c: Merged revisions 243570 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010) + | 9 lines Extend announcement URL used with Queue from 80 chars + to PATH_MAX. (closes issue #16488) Reported by: syspert Patches: + soundfilelen.pacth-2 uploaded by syspert (license 938) Review: + https://reviewboard.asterisk.org/r/475/ ........ + + * Makefile, CHANGES, include/asterisk/options.h, main/asterisk.c, + main/loader.c: Add new option to asterisk.conf (lockconfdir) to + protect conf dir during reloads (closes issue #16358) Reported + by: raarts Patches: lockconfdir.diff uploaded by raarts (license + 937) modified by me + +2010-01-27 18:08 +0000 [r243487] Mark Michelson + + * main/pbx.c, /: Merged revisions 243486 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r243486 | mmichelson | 2010-01-27 12:06:43 -0600 (Wed, 27 Jan + 2010) | 3 lines Use a safe list traversal while checking for + duplicate vars in pbx_builtin_setvar_helper. ........ + +2010-01-27 17:32 +0000 [r243482] Russell Bryant + + * funcs/func_channel.c, channels/chan_iax2.c: Fix the ability to + specify an OSP token for an outbound IAX2 call. When this patch + was originally submitted, the code allowed for the token to be + set via a channel variable. I decided that a cleaner approach + would be to integrate it into the CHANNEL() function. + Unfortunately, that is not a suitable approach. It's not possible + to get the value set on the channel soon enough using that + method. So, go back to the simple channel variable method. + (closes issue #16711) Reported by: homesick Patches: iax-svn.diff + uploaded by homesick (license 91) + +2010-01-26 23:56 +0000 [r243391] David Vossel + + * /, main/features.c: Merged revisions 243390 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r243390 | dvossel | 2010-01-26 17:55:49 -0600 (Tue, 26 Jan 2010) + | 9 lines fixes bug with channel receiving wrong privileges after + call parking (closes issue #16429) Reported by: Yasuhiro Konishi + Patches: features.c.diff uploaded by Yasuhiro Konishi (license + 947) Tested by: dvossel ........ + +2010-01-26 20:49 +0000 [r243346] David Ruggles + + * apps/app_senddtmf.c: Code clean up in app_senddtmf Pushes code + clean up done in app_externalivr back into app_senddtmf Review: + https://reviewboard.asterisk.org/r/473/ + +2010-01-26 18:20 +0000 [r243244-243266] Jeff Peeler + + * main/channel.c, /: Merged revisions 243258 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r243258 | jpeeler | 2010-01-26 12:19:10 -0600 (Tue, 26 Jan 2010) + | 2 lines Remove unnecessary code in ast_read as issue 16058 has + been fully solved now. ........ + + * main/frame.c: Fix crash resulting from frames with invalid data + pointers. In ast_frdup the frame data union does not get set to + point to malloced memory if the datalen is zero, so make sure to + handle the same case in ast_frisolate appropriately. (closes + issue #16058) Reported by: atis Patches: bug16058-fix.patch + uploaded by jpeeler (license 325) Tested by: atis + +2010-01-26 17:40 +0000 [r243200-243242] David Vossel + + * main/test.c: modify 'test show registered' cli output format In + order to improve readability, the output from 'test show + registered' has been modified to truncate fields to fit within + the format output if they are over a certain length. + + * include/asterisk/utils.h, channels/chan_sip.c, tests/test_utils.c + (added), main/test.c, main/utils.c: RFC compliant uri and + display-name encode/decode 1. URI Encoding This patch changes + ast_uri_encode()'s behavior when doreserved is enabled. + Previously when doreserved was enabled only a small set of + reserved characters were encoded. This set was comprised + primarily of the reserved characters defined in RFC3261 section + 25.1, but contained other characters as well. Rather than only + escaping the reserved set, doreserved now escapes all characters + not within the unreserved set as defined by RFC 3261 and RFC + 2396. Also, the 'doreserved' variable has been renamed to + 'do_special_char' in attempts to avoid confusion. When doreserve + is not enabled, the previous logic of only encoding the + characters <= 0X1F and > 0X7f remains, except for the '%' + character, which must always be encoded as it signifies a HEX + escaped character during the decode process. 2. URI Decoding: + Break up URI before decode. In chan_sip.c ast_uri_decode is + called on the entire URI instead of it's individual parts after + it is parsed. This is not good as ast_uri_decode can introduce + special characters back into the URI which can mess up parsing. + This patch resolves this by not decoding a URI until parsing is + completely done. There are many instances where we check to see + if pedantic checking is enabled before we decode a URI. In these + cases a new macro, SIP_PEDANTIC_DECODE, is used on the individual + parsed segments of the URI rather than constantly putting if + (pedantic) { decode() } checks everywhere in the code. In the + areas where ast_uri_decode is not dependent upon pedantic + checking this macro is not used, but decoding is still moved to + each individual part of the URI. The only behavior that should + change from this patch is the time at which decoding occurs. + Since I had to look over every place URI parsing occurs to create + this patch, I found several places where we use duplicate code + for parsing. To consolidate the code, those areas have updated to + use the parse_uri() function where possible. 3. SIP display-name + decoding according to RFC3261 section 25. To properly decode the + display-name portion of a FROM header, chan_sip's + get_calleridname() function required a complete re-write. More + information about this change can be found in the comments at the + beginning of this function. 4. Unit Tests. Unit tests for + ast_uri_encode, ast_uri_decode, and get_calleridname() have been + written. This involved the addition of the test_utils.c file for + testing the utils api. (closes issue #16299) Reported by: wdoekes + Patches: astsvn-16299-get_calleridname.diff uploaded by wdoekes + (license 717) get_calleridname_rewrite.diff uploaded by dvossel + (license 671) Tested by: wdoekes, dvossel, Nick_Lewis Review: + https://reviewboard.asterisk.org/r/469/ + +2010-01-26 15:46 +0000 [r243118-243158] Russell Bryant + + * tests/test_substitution.c: Log the variable name being tested. + + * tests/test_substitution.c: Update test_substitution to show + failures in the test log. + + * funcs/func_aes.c: Update func_aes to its pre-ast_str_substitution + state. This change makes the AES tests in test_substitution.c + pass. We still need to work through what's going wrong in the + ast_str version. + +2010-01-26 01:56 +0000 [r242967-243077] Tilghman Lesher + + * tests/test_substitution.c: Fixing last errors in the conversion, + though it appears that the AES_* functions are still broken. + + * tests/test_substitution.c: Using a dummy channel causes CDR() + testing to fail. + + * tests/test_substitution.c: Wish I had gotten to the review before + this got submitted, because there's failures we need to address. + + * /, main/Makefile, res/Makefile: Merged revisions 242969 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r242969 | tilghman | 2010-01-25 15:50:22 -0600 (Mon, 25 Jan 2010) + | 2 lines Err, and use the new menuselect define, too. ........ + + * build_tools/cflags.xml, /, build_tools/menuselect-deps.in, + configure, configure.ac: Merged revisions 242966 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r242966 | tilghman | 2010-01-25 15:36:33 -0600 (Mon, 25 + Jan 2010) | 2 lines Only rebuild parsers by an option in + menuselect ........ + +2010-01-25 21:32 +0000 [r242954-242965] Russell Bryant + + * tests/test_substitution.c, tests/test_heap.c, + tests/test_ast_format_str_reduce.c, tests/test_skel.c, + tests/test_sched.c: Make unit test modules depend on + TEST_FRAMEWORK instead of off by default. + + * tests/test_substitution.c: Convert test_substitution module to + the unit test API. Review: + https://reviewboard.asterisk.org/r/474/ + +2010-01-25 21:20 +0000 [r242933] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/oochannels.c, + addons/ooh323c/src/ooCalls.c: small corrections in call clearing + +2010-01-25 21:13 +0000 [r242904-242919] Olle Johansson + + * main/pbx.c, main/manager.c, include/asterisk/pbx.h: Change api + for pbx_builtin_setvar to actually return error code if a + function can't be written to. This patch removes code that was + duplicated from pbx.c to manager.c in order to prevent API change + in released versions of Asterisk. There are propably also other + places that would benefit from reading the return code and react + if a function returns error codes on writing a value into it. + + * main/manager.c, /: Merged revisions 242850 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r242850 | oej | 2010-01-25 21:03:38 +0100 (Mån, 25 Jan 2010) | 2 + lines Report error when writing to functions returns error in AMI + setvar action ........ + +2010-01-25 20:18 +0000 [r242857] Tilghman Lesher + + * /, configure, main/Makefile, configure.ac, res/Makefile: Merged + revisions 242852 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r242852 | tilghman | 2010-01-25 14:15:45 -0600 (Mon, 25 Jan 2010) + | 2 lines Restore FreeBSD to able-to-compile-ish-mode ........ + +2010-01-25 18:01 +0000 [r242812] Terry Wilson + + * res/res_calendar.c: Fix INTERNAL_OBJ error on stop when + calendars.conf missing Initialize the calendars container before + calling load_config and return FAILURE on allocation failure. + Also, use the AST_MODULE_LOAD_* values for return values. Thanks + to rmudgett for pointing out the error and the need to use the + defined values for return + +2010-01-25 05:45 +0000 [r242719-242729] Tilghman Lesher + + * /, main/Makefile, res/Makefile: Merged revisions 242728 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r242728 | tilghman | 2010-01-24 23:42:22 -0600 (Sun, 24 Jan 2010) + | 2 lines Buildbot pointed out an error (thanks, buildbot!) + ........ + + * /, res/Makefile: Merged revisions 242723 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r242723 | tilghman | 2010-01-24 23:33:37 -0600 (Sun, 24 Jan 2010) + | 2 lines Oops, should have used CMD_PREFIX, not ECHO_PREFIX, for + the commands. ........ + + * /, main/Makefile: Merged revisions 242683 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r242683 | tilghman | 2010-01-24 23:13:28 -0600 (Sun, 24 Jan 2010) + | 2 lines Make the build of the Asterisk expression parser match + that of the AEL parser. ........ + +2010-01-24 22:42 +0000 [r242645] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c, + addons/ooh323c/src/ooStackCmds.h, + addons/ooh323c/src/oochannels.c, + addons/ooh323c/src/ooCmdChannel.c, + addons/ooh323c/src/ooStackCmds.c: AST_CONTROL_CONNECTED_LINE + frame type processing added to setup DisplayIE field incorrect + q.931 message order filtered on incoming calls (first msg must be + setup, next must be not setup) + +2010-01-24 21:49 +0000 [r242607] Sean Bright + + * res/res_phoneprov.c: Instead of crashing, allocate our header + ast_str before we try to use it. (closes issue #16680) Reported + by: lmadsen Patches: issue16680_20100122.patch uploaded by + seanbright (license 71) Tested by: lmadsen + +2010-01-24 06:40 +0000 [r242521] Tilghman Lesher + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + pbx/Makefile, res/Makefile, makeopts.in: Merged revisions 242520 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) + | 8 lines Only rebuild bison and flex source files on demand, if + bison and flex are detected by the configure script. Changed + after discussion on the -dev list about possible unnecessary + build failures, due to checkouts/untars causing these special + source files to possibly be newer than their resulting C files. + This should additionally ensure that nobody need learn about + extra Makefile arguments to ensure the proper files get rebuilt + when changes are made to these special source files. ........ + +2010-01-22 21:45 +0000 [r242424] Tilghman Lesher + + * /, res/Makefile: Merged revisions 242423 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r242423 | tilghman | 2010-01-22 15:44:18 -0600 (Fri, 22 Jan 2010) + | 7 lines Rebuild from flex, bison sources when necessary. (issue + #14629) Reported by: Marquis Patches: + 20100121__issue14629.diff.txt uploaded by tilghman (license 14) + ........ + +2010-01-22 16:20 +0000 [r242357] David Ruggles + + * apps/app_externalivr.c: Add send DTMF feature to ExternalIVR app + Implemented a new command 'D' that allows client IVRs to send + DTMF digits to the channel. (closes issue #16615) Reported by: + thedavidfactor Review: https://reviewboard.asterisk.org/r/465/ + +2010-01-22 15:09 +0000 [r242317] Tilghman Lesher + + * tests/test_sched.c: The irony of not compile-testing a test + program before committing is killing me. + +2010-01-22 09:28 +0000 [r242227] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 242226 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3 + lines Initialize notify_types to NULL ........ + +2010-01-22 04:57 +0000 [r242184-242186] Russell Bryant + + * main/test.c: Update the doxygenification of some comments. + + * tests/test_sched.c: Convert scheduler API entry order test to the + test API. Review: https://reviewboard.asterisk.org/r/470/ + + * tests/test_skel.c: Add test API usage example to test_skel.c. + Review: https://reviewboard.asterisk.org/r/471/ + +2010-01-21 22:37 +0000 [r242092] Mark Michelson + + * main/acl.c: Add missing argument to ast_calloc calls. + +2010-01-21 21:05 +0000 [r242043] Olle Johansson + + * main/acl.c: Make sure we initialize the ast_ha structure with + ast_calloc + +2010-01-21 15:27 +0000 [r241938] Sean Bright + + * /, configure, configure.ac: Merged revisions 241932 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r241932 | seanbright | 2010-01-21 10:25:46 -0500 (Thu, + 21 Jan 2010) | 5 lines Fix configure check for PTHREAD_ONCE_INIT + when manually adding -Wall to CFLAGS. (closes issue #16666) + Reported by: romain_proformatique ........ + +2010-01-21 15:14 +0000 [r241896] Tilghman Lesher + + * channels/chan_vpb.cc: Formats are inconsistent between even + 32-bit and 64-bit Linux. Use casts to ensure both compile. + +2010-01-21 14:10 +0000 [r241855-241856] Russell Bryant + + * main/test.c: Point to a useful reference on the XML output + format. + + * main/test.c: Modify test results XML format to match the JUnit + format. When this code was developed, we came up with our own XML + format for the test output. I have since started looking at + integration with other tools, namely continuous integration + frameworks, and this format seems to be supported across a number + of applications. With these changes in place, I was able to get + Atlassian Bamboo to interpret the test results. + +2010-01-21 05:54 +0000 [r241766] Tilghman Lesher + + * /, funcs/func_math.c: Merged revisions 241765 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r241765 | tilghman | 2010-01-20 23:53:17 -0600 (Wed, 20 Jan 2010) + | 2 lines Guard against division by zero. ........ + +2010-01-20 21:14 +0000 [r241627-241714] David Vossel + + * res/res_rtp_asterisk.c: rtp timestamp to timeval calculation fix + The rtp timestamp to timeval calculation was only accurate for + 8kHz audio. This patch corrects this. Review: + https://reviewboard.asterisk.org/r/468/ SWP-648 + + * Makefile, /: Merged revisions 241626 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r241626 | dvossel | 2010-01-20 14:00:04 -0600 (Wed, 20 Jan 2010) + | 6 lines fixes parsing error in Makefile. Some echo lines were + missing "; . Thanks to jparker for pointing out the problem. + ........ + +2010-01-20 17:49 +0000 [r241581] Alec L Davis + + * main/cdr.c: Add Calling and Called Subaddress to CDR record + Requires 'callingsubaddr' and 'calledsubaddr' fields in backend + cdr. (closes issue #16600) Reported by: alecdavis Patches: + cdr_subaddr.diff.txt uploaded by alecdavis (license 585) Tested + by: alecdavis Review: https://reviewboard.asterisk.org/r/460/ + +2010-01-20 13:01 +0000 [r241503] Kevin P. Fleming + + * channels/chan_vpb.cc: Fix up compile breakage from + ast_tvdiff_ms() API change. + +2010-01-20 08:18 +0000 [r241416] Alec L Davis + + * main/pbx.c, channels/sig_pri.c: Update CDR variables as pbx + starts Allows CDR variables added in cdr.c:set_one_cid to become + visable during the call, by executing ast_cdr_update() early in + __ast_pbx run. Reverts sig_pri changes in trunk that are specific + to isdn technology only. (closes issue #16638) Reported by: + alecdavis Patches: cdr_update.diff3.txt uploaded by alecdavis + (license 585) Tested by: alecdavis + +2010-01-19 22:59 +0000 [r241366] Jeff Peeler + + * main/pbx.c: Initialize data on the stack so that Park doesn't + interpret random arguments. passdata was only being set in + pbx_substitue_variables when arguments were passed. (closes issue + #16406) (closes issue #16586) Reported by: DLNoah Patches: + bug16586v2.patch uploaded by jpeeler (license 325) Tested by: + DLNoah + +2010-01-19 22:41 +0000 [r241364] Tilghman Lesher + + * doc/janitor-projects.txt, apps/app_sendtext.c: Enable SendText to + send strings in encoded format. See + http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html + +2010-01-19 18:51 +0000 [r241314-241315] Jeff Peeler + + * channels/chan_agent.c: small correction from 241314 + + * /, channels/chan_agent.c: Merged revisions 241227 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19 + Jan 2010) | 13 lines Fix deadlock in agent_read by removing call + to agent_logoff. One must always lock the agents list lock before + the agent private. agent_read locks the private immediately, so + locking the agents list lock is not an option (which is what + agent_logoff requires). Because agent_read already has access to + the agent private all that is necessary is to do the required + hanging up that agent_logoff performed. (closes issue #16321) + Reported by: valon24 Patches: bug16321.patch uploaded by jpeeler + (license 325) ........ + +2010-01-19 17:42 +0000 [r241230] Jason Parker + + * Makefile: Allow parallel make (-j) to work properly. After some + back and forth with the reporter, we came up with the necessary + changes. (closes issue #16489) Reported by: Chainsaw Patches: + asterisk-1.6.2.1-parallel-make-minimal.patch uploaded by Chainsaw + (license 723) Tested by: Chainsaw, qwell + +2010-01-19 00:28 +0000 [r241188] Tilghman Lesher + + * main/srv.c, res/res_agi.c, CHANGES, include/asterisk/srv.h: + Create iterative method for querying SRV results, and use that + for finding AGI servers. (closes issue #14775) Reported by: + _brent_ Patches: 20091215__issue14775.diff.txt uploaded by + tilghman (license 14) hagi-5.patch uploaded by brent (license + 388) Tested by: _brent_ Reviewboard: + https://reviewboard.asterisk.org/r/378/ + +2010-01-19 00:24 +0000 [r241187] Alec L Davis + + * channels/sig_pri.c: Update CDR variables before pbx starts + (overlap dial) Allows CDR variables added in cdr.c:set_one_cid to + become visable during the call. (issue #16638) Reported by: + alecdavis Patches: cdr_update.diff2.txt uploaded by alecdavis + (license 585) Tested by: alecdavis + +2010-01-18 22:31 +0000 [r241143] Jeff Peeler + + * main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c, + main/features.c, pbx/pbx_dundi.c, main/enum.c, + include/asterisk/time.h, main/timing.c: Extend max call limit + duration from 24.8 days to 292+ million years. If the limit was + set past MAX_INT upon answering, the call was immediately hung up + due to overflow from the return of ast_tvdiff_ms (in + ast_check_hangup). The time calculation functions ast_tvdiff_sec + and ast_tvdiff_ms have been changed to return an int64_t to + prevent overflow. Also the reporter suggested adding a message + indicating the reason for the call hanging up. Given that the new + limit is so much higher, the message (which would only really be + useful in the overflow scenario) has been made a debug message + only. (closes issue #16006) Reported by: viraptor + +2010-01-18 22:03 +0000 [r241098] Jason Parker + + * main/rtp_engine.c: Fix an RTP instance allocation failure on + Solaris. (closes issue #16543) Reported by: crjw Patches: + rtp_sin_family.patch uploaded by crjw (license 963) Tested by: + crjw, qwell + +2010-01-18 22:00 +0000 [r241097] Alec L Davis + + * channels/sig_pri.c: Update CDR variables before pbx starts Allows + CDR variables added in cdr.c:set_one_cid to become visable during + the call. (closes issue #16638) Reported by: alecdavis Patches: + cdr_update.diff.txt uploaded by alecdavis (license 585) + +2010-01-18 19:57 +0000 [r241016] Sean Bright + + * /, main/config.c: Merged revisions 241015 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r241015 | seanbright | 2010-01-18 14:54:19 -0500 (Mon, 18 Jan + 2010) | 12 lines Plug a memory leak when reading configs with + their comments. While reading through configuration files with + the intent of returning their full contents (comments + specifically) we allocated some memory and then forgot to free + it. This doesn't fix 16554 but clears up a leak I had in the lab. + (issue #16554) Reported by: mav3rick Patches: + issue16554_20100118.patch uploaded by seanbright (license 71) + Tested by: seanbright ........ + +2010-01-18 19:26 +0000 [r241012] Tilghman Lesher + + * funcs/func_strings.c, CHANGES: Make HASHes inheritable across + channel creation. + +2010-01-18 18:00 +0000 [r240973-240974] David Ruggles + + * UPGRADE.txt: ExternalIVR information for UPGRADE.txt added a + paragraph about the fixes and changes to the ExternalIVR + application. + + * doc/externalivr.txt: Updated ExternalIVR documentation Rewrote a + large portion of the existing documentation and added information + about the TCP/IP socket interface + +2010-01-18 17:45 +0000 [r240971] David Vossel + + * Makefile, CHANGES: transmit_silence_during_record replaced by + transmit_silence In asterisk.conf, transmit_silence_during_record + has been removed in favor of using only the transmit_silence + option. The transmit_silence_during_record option remains a valid + option in asterisk.conf, but has been removed from the sample + config and noted in CHANGES. + +2010-01-18 17:41 +0000 [r240969] David Ruggles + + * apps/app_externalivr.c: Add notification of interrupted file Add + file information to data element of T event so the file + information is sent to the client when it is interrupted. + Previously only notification of pending files that were dropped + was sent (closes issue #16147) Reported by: thedavidfactor Tested + by: thedavidfactor Review: + https://reviewboard.asterisk.org/r/449/ + +2010-01-18 16:45 +0000 [r240842-240887] David Vossel + + * Makefile: updated transmit_silence option documentation in + asterisk.conf This patch updates the transmit_silence option to + better document why the option exists, and what it affects. + Thanks to russell for providing the verbage for this update. + + * apps/app_queue.c: fixes spelling error. s/memeber/member + +2010-01-17 19:45 +0000 [r240717] Sean Bright + + * main/pbx.c: Avoid a crash on Solaris when running 'core show + functions.' (closes issue #16309) Reported by: asgaroth + +2010-01-16 00:54 +0000 [r240667] Sean Bright + + * res/res_musiconhold.c: Get MoH building on OpenSolaris. + +2010-01-15 23:50 +0000 [r240629] Tilghman Lesher + + * Makefile, main/asterisk.c: Err, oops, it was already the way I + intended. + +2010-01-15 23:09 +0000 [r240548-240552] Russell Bryant + + * include/asterisk/doxygen/commits.h: Note where empty lines should + reside in commit messages. + + * Makefile, /: Merged revisions 240547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r240547 | russell | 2010-01-15 17:06:11 -0600 (Fri, 15 Jan 2010) + | 2 lines Fix a spelling error in the asterisk.conf sample. + ........ + +2010-01-15 22:07 +0000 [r240505] Sean Bright + + * res/res_timing_timerfd.c: Clarify error message in + res_timing_timerfd. + +2010-01-15 21:42 +0000 [r240421-240500] Tilghman Lesher + + * utils/astcanary.c: Oops, missed an include + + * utils/astcanary.c, main/asterisk.c: The previous attempt at using + a pipe to guarantee astcanary shutdown did not work. We're + revisiting the previous patch, albeit with a method that + overcomes the prior criticism that it was not POSIX-compliant. + (closes issue #16602) Reported by: frawd Patches: + 20100114__issue16602.diff.txt uploaded by tilghman (license 14) + Tested by: frawd + + * apps/app_directed_pickup.c, main/features.c, + include/asterisk/manager.h: Add pickup event to AMI. Also, fix + AMI documentation. (closes issue #16431) Reported by: syspert + Patches: 20100112__issue16431.diff.txt uploaded by tilghman + (license 14) + +2010-01-15 20:58 +0000 [r240420] Mark Michelson + + * main/utils.c: Make sure to set owner_line, ownder_func, and + owner_file in ast_calloc_with_stringfields. Asterisk would crash + on startup if MALLOC_DEBUG were set in menuselect. This is + because the manager action UpdateConfig had to resize its string + field allocation to set the description. When the resize + occurred, ast_copy_string would crash because we were attempting + to copy a string from a NULL pointer. Setting the strings + initially makes the code much less crashy. + +2010-01-15 20:58 +0000 [r240415-240419] Tilghman Lesher + + * apps/app_voicemail.c: Make sure that the limit is N, not N - 1. + + * /, apps/app_voicemail.c: Merged revisions 240414 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r240414 | tilghman | 2010-01-15 14:52:27 -0600 (Fri, 15 + Jan 2010) | 15 lines Disallow leaving more than maxmsg + voicemails. This is a possibility because our previous method + assumed that no messages are left in parallel, which is not a + safe assumption. Due to the vmu structure duplication, it was + necessary to track in-process messages via a separate structure. + If at some point, we switch vmu to an ao2-reference-counted + structure, which would eliminate the prior noted duplication of + structures, then we could incorporate this new in-process + structure directly into vmu. (closes issue #16271) Reported by: + sohosys Patches: 20100108__issue16271.diff.txt uploaded by + tilghman (license 14) 20100108__issue16271__trunk.diff.txt + uploaded by tilghman (license 14) + 20100108__issue16271__1.6.0.diff.txt uploaded by tilghman + (license 14) Tested by: jsutton ........ + +2010-01-15 20:41 +0000 [r240411] Russell Bryant + + * main/event.c: Ensure payload type is properly checked when + comparing against cached events. (closes issue #16607) Reported + by: ddv2005 Patches: event.patch uploaded by ddv2005 (license + 769) + +2010-01-15 18:21 +0000 [r240368] Sean Bright + + * main/pbx.c, main/manager.c, res/res_smdi.c, apps/app_meetme.c, + channels/chan_sip.c, cel/cel_tds.c, main/features.c, + res/res_phoneprov.c, cdr/cdr_tds.c, apps/app_jack.c: Convert a + few places to use ast_calloc_with_stringfields where applicable. + +2010-01-15 16:51 +0000 [r240329] Russell Bryant + + * configure: Update configure script for an OSP toolkit related + change. + +2010-01-15 16:28 +0000 [r240328] Kevin P. Fleming + + * configs/sip.conf.sample: Clarify RTP NAT handling a bit. + +2010-01-14 23:13 +0000 [r240226-240271] Sean Bright + + * res/res_config_ldap.c: Plug a memory leak in res_config_ldap. + (closes issue #16257) Reported by: nito Patches: + issue16257_20100111.diff uploaded by seanbright (license 71) + + * res/res_timing_timerfd.c: If we aren't running on a machine that + support CLOCK_MONOTONIC, don't load. Group developed and tested + by seanbright, Corydon76, Kobaz, and Amorsen. + +2010-01-14 18:03 +0000 [r240179] Jeff Peeler + + * main/channel.c: Fix broken call pickup The problem was the + OUTGOING flag was not getting set properly on the channel, + resulting in pickup failing as ast_read thought the call was + inbound. Refer to 170393 for a more verbose description as this + is the same exact change. (closes issue #16539) Reported by: + syspert Patches: bug16539.patch uploaded by jpeeler (license 325) + Tested by: syspert + +2010-01-14 17:34 +0000 [r240129-240175] Tilghman Lesher + + * main/pbx.c: Similarly, ensure that matchcid is duplicated + correctly when merging contexts. + + * main/pbx.c: Ensure that the callerid is NULL when the parent is + effectively NULL. This applies only to pattern-match hints, which + create exact-match hints on the fly. + +2010-01-14 16:14 +0000 [r240078] Matthew Nicholson + + * main/udptl.c: This change fixes a few bugs in the way the far max + IFP was calculated that were introduced in r231692. (closes issue + #16497) Reported by: globalnetinc Patches: + udptl-max-ifp-fix1.diff uploaded by mnicholson (license 96) + Tested by: globalnetinc + +2010-01-14 14:38 +0000 [r240039] Leif Madsen + + * doc/building_queues.txt (added): Add documentation about how to + build queues. Add a how-to set of documentation about building + queues with Asterisk. This documentation is based on Asterisk + 1.6.2 but should work on most versions with minor modifications. + (closes issue #16237) Reported by: lmadsen Patches: Building + Queues (FINAL).txt uploaded by lmadsen (license 10) Tested by: + pdhales, lmadsen, cmdrwalrus + +2010-01-13 23:22 +0000 [r239920-239997] Tilghman Lesher + + * main/pbx.c: Oops, another tag error + + * main/pbx.c: Oops, missed a closing tag + + * main/pbx.c, include/asterisk/pbx.h: Add the TESTTIME() dialplan + function, which permits testing GotoIfTime. Specifically, by + setting TESTTIME() to a particular date and time, you can test + whether a dialplan correctly branches as was intended. This was + developed after recent questions on the -users list on how to + test their holiday dialplan logic. (closes issue #16464) Reported + by: tilghman Patches: 20100112__issue16464.diff.txt uploaded by + tilghman (license 14) Review: + https://reviewboard.asterisk.org/r/458/ + + * main/ast_expr2f.c, main/ast_expr2.fl: Flex uses fwrite + incorrectly, which breaks the build. Providing a workaround. + +2010-01-13 19:48 +0000 [r239839] Jeff Peeler + + * /, main/features.c: Merged revisions 239838 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r239838 | jpeeler | 2010-01-13 13:43:33 -0600 (Wed, 13 Jan 2010) + | 11 lines Fix regression for timed out parked call returning to + caller This issue seems to have been exposed by the fix in 160390 + whereby using a masquerade prevented a crash. The new channel + used in the masquerade was not copying the macro information from + the old channel. (closes issue #15459) Reported by: djrodman + Patches: patch_15459.txt uploaded by mnick (license ) ........ + +2010-01-13 19:31 +0000 [r239834] Leif Madsen + + * configs/extensions.conf.sample: Add more examples to + extensions.conf showing how to use various functionality and + provide commonly useful features. (closes issue #16090) Reported + by: pprindeville Patches: extensions.conf-bugid16090.patch#3 + uploaded by pprindeville (license 347) Tested by: tzafrir, + pprindeville, lmadsen + +2010-01-13 18:16 +0000 [r239797] Tilghman Lesher + + * main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Code + previously added to ast_expr2f.c warranted a change in the source + file ast_expr2.fl. Also, made a Makefile change to ensure that + the expression parser C source files get regenerated correctly, + when we need that to happen. + +2010-01-13 16:31 +0000 [r239712] David Vossel + + * Makefile, main/channel.c, apps/app_waitforring.c, + apps/app_waitforsilence.c: add silence gen to wait apps + asterisk.conf's 'transmit_silence' option existed before this + patch, but was limited to only generating silence while recording + and sending DTMF. Now enabling the transmit_silence option + generates silence during wait times as well. To achieve this, + ast_safe_sleep has been modified to generate silence anytime no + other generators are present and transmit_silence is enabled. + Wait apps not using ast_safe_sleep now generate silence when + transmit_silence is enabled as well. (closes issue #16524) + Reported by: kobaz (closes issue #16523) Reported by: kobaz + Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/456/ + +2010-01-13 10:45 +0000 [r239663-239665] Olle Johansson + + * main/poll.c: MAX() moved to utils.h + + * channels/chan_sip.c: SIP Show channelstats fix - use float + division to show proper stats (closes issue #15819) Reported by: + klaus3000 Patches: asterisk-sip-show-channelstats-trunk.txt + uploaded by klaus3000 (license 65) Tested by: klaus3000, oej This + patch is for trunk only and will be blocked in 1.6.2 + +2010-01-13 07:02 +0000 [r239624-239625] TransNexus OSP Development + + * doc/tex/channelvariables.tex: Updated channel variable list of + osplookup application. + + * apps/app_osplookup.c: Updated XML doc for OSP. + +2010-01-12 19:58 +0000 [r239571] Tilghman Lesher + + * main/pbx.c: Blank callerid and NULL callerid should not compare + equal. The second is the default state for matching CID in the + dialplan (no matching) while the first matches one particular + CallerID. This is a regression. (fixes AST-314, SWP-611) + +2010-01-12 18:55 +0000 [r239525] Alec L Davis + + * main/cdr.c: add Dialed Number Identifier (DNID) field to cdr + records. reviewboard link: + https://reviewboard.asterisk.org/r/455/ Reported by: alecdavis + Tested by: alecdavis Patch cdr_dnid.diff2.txt uploaded by + alecdavis (license 585) + +2010-01-12 18:22 +0000 [r239520] Leif Madsen + + * configs/sip.conf.sample: Note that direct T.38 is not supported. + (closes issue #16411) Reported by: stanusr Patches: + __20091210-sip.conf.sample-documentation.txt uploaded by lmadsen + (license 10) + +2010-01-12 17:09 +0000 [r239473] Sean Bright + + * res/res_config_ldap.c: Fix crash in res_config_ldap. We need to + allocate enough room for 2 pointers, not 2 characters. (closes + issue #16397) Reported by: bklang Patches: res_config_ldap.patch + uploaded by applsplatz (license 949) Tested by: applsplatz + +2010-01-12 16:14 +0000 [r239427] David Vossel + + * channels/chan_sip.c: fixes text support in sdp answer The code + that handled setting 'm=text' in the sdp was not executing in the + correct order. The check to see if text was needed came after the + check to add 'm=text' to the sdp, this resulted in 'm=text' + always being set to 0 because it looked like text was never + required. (closes issue #16457) Reported by: peterj Patches: + textportinsdp.diff uploaded by peterj (license 951) + issue16457.diff uploaded by dvossel (license 671) Tested by: + peterj + +2010-01-12 07:48 +0000 [r239389] Olle Johansson + + * include/asterisk/astmm.h: Adding Tilghman's documentation from + asterisk-dev to the actual file. + +2010-01-12 03:21 +0000 [r239152-239308] Tilghman Lesher + + * /, contrib/scripts/safe_asterisk: Merged revisions 239307 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r239307 | tilghman | 2010-01-11 21:18:36 -0600 (Mon, 11 Jan 2010) + | 8 lines Portability and other fixes for the safe_asterisk + script (closes issue #16416) Reported by: bklang Patches: + safe_asterisk-compat-1.patch uploaded by bklang (license 919) + 20100106__issue16416__trunk.diff.txt uploaded by tilghman + (license 14) Tested by: bklang ........ + + * contrib/init.d/rc.mandriva.asterisk, + contrib/init.d/rc.debian.asterisk, + contrib/init.d/rc.redhat.asterisk, + contrib/init.d/rc.gentoo.asterisk, + contrib/init.d/rc.slackware.asterisk, + contrib/init.d/rc.archlinux.asterisk, + contrib/init.d/rc.suse.asterisk: Add LSB headers to init scripts. + (closes issue #14864) Reported by: lathama Patches: + lsb-init-info-debian.diff uploaded by pkempgen (license 169) + + * res/res_pktccops.c: Socket level option is SOL_SOCKET, not + SO_SOCKET. (issue #16580) + + * Makefile, contrib/init.d/rc.mandriva.asterisk, + contrib/init.d/rc.debian.asterisk, + contrib/init.d/rc.redhat.asterisk, + contrib/init.d/rc.suse.asterisk: Permit more options in the + Makefile as to startup options (closes issue #16454) Reported by: + syspert Patches: 20091228__issue16454__3.diff.txt uploaded by + tilghman (license 14) Tested by: syspert + + * Makefile: Including bundle1.o breaks Tiger and Leopard (issue + #16449) + + * addons/cdr_mysql.c, configs/cdr_mysql.conf.sample: Permit dates + and times to be stored in timezones other than the default + (typically, UTC) (closes issue #16401) Reported by: lordmortis + +2010-01-11 16:41 +0000 [r239111-239114] Sean Bright + + * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, + res/res_calendar_caldav.c, res/res_clialiases.c: Pass NULL for + the ao2_callback function pointer instead of duplicating cb_true. + + * main/astobj2.c: Fix ao2_callback when both OBJ_MULTIPLE and + OBJ_NODATA are passed. There is an issue which only affects trunk + and the new ao2_callback OBJ_MULTIPLE implementation. When both + OBJ_MULTIPLE and OBJ_NODATA are passed, only the first object is + visited, regardless of what is returned by the specified + callback. This causes a problem when we are clearing a container, + i.e.: ao2_callback(container, OBJ_UNLINK | OBJ_NODATA | + OBJ_MULTIPLE, NULL, NULL); Only unlinks the first object. This + patch resolves this. (closes issue #16564) Reported by: pj + Patches: issue16564_20100111.diff uploaded by seanbright (license + 71) Tested by: pj, seanbright Review: + https://reviewboard.asterisk.org/r/457/ + + * main/test.c: Fix spelling of 'category.' + +2010-01-10 19:37 +0000 [r239074] Tilghman Lesher + + * addons/chan_ooh323.c, main/frame.c, channels/chan_iax2.c: + According to POSIX, the capital L modifier applies only to + floating point types. Fixes a crash on Solaris. (closes issue + #16572) Reported by: crjw Patches: frame_changes.patch uploaded + by crjw (license 963) Plus several others found and fixed by me + +2010-01-10 17:53 +0000 [r239037] Alexandr Anikin + + * addons/ooh323c/src/ooq931.h, addons/ooh323c/src/oochannels.c, + addons/ooh323c/src/ooq931.c: add docallbacks flag in q931decode + function because when we decode received q931 packet we must do + callbacks and when we print sended q931 packet we must not. + +2010-01-10 06:56 +0000 [r239000] Tilghman Lesher + + * Makefile, main/asterisk.c: It's been long enough -- make the + behavior introduced in 1.6 the default. + +2010-01-09 01:08 +0000 [r238916] Tilghman Lesher + + * main/manager.c, /: Merged revisions 238915 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238915 | tilghman | 2010-01-08 18:57:58 -0600 (Fri, 08 Jan 2010) + | 6 lines -1 is interpreted as an error, intead of the maximum + mask. (closes issue #16241) Reported by: vnovy Patches: + manager.c.patch uploaded by vnovy (license 922) ........ + +2010-01-08 23:30 +0000 [r238835] Jeff Peeler + + * /, main/features.c: Merged revisions 238834 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238834 | jpeeler | 2010-01-08 17:28:37 -0600 (Fri, 08 Jan 2010) + | 4 lines Stop a crash when no peer is passed to masq_park_call. + (distantly related to issue #16406) ........ + +2010-01-08 22:54 +0000 [r238754-238795] Tilghman Lesher + + * res/res_musiconhold.c: Add the class actually used in the + MusicOnHold start event. (closes issue #16499) Reported by: + syspert Patches: mohclass.patch uploaded by syspert (license 938) + + * res/res_agi.c: Initialize variables that we attempt to free + later. (closes issue #16302) Reported by: yahsyn Patches: + 20091124__issue16302.diff.txt uploaded by tilghman (license 14) + Tested by: yahsyn + +2010-01-08 21:04 +0000 [r238716] Matthew Nicholson + + * tests/test_ast_format_str_reduce.c (added): Added a test for + ast_format_reduce_str(). (related to issue #16560) + +2010-01-08 19:39 +0000 [r238635] David Vossel + + * include/asterisk/audiohook.h, main/audiohook.c: fixes + AUDIOHOOK_INHERIT regression During the process of removing an + audiohook from one channel and attaching it to another the + audiohook's status is updated to DONE and then back to whatever + it was previously. Typically updating the status after setting it + to DONE is not a good idea because DONE can trigger unrecoverable + audiohook destruction events... because of this a conditional + check was added to audiohook_update_status to explicitly prevent + the audiohook from ever changing after being set to DONE. It was + this check that prevented audiohook inherit from work properly + though. Now ast_audiohook_move_by_source is treated as a special + exception, as the audiohook must be returned to its previous + status after attaching it to the new channel. This is only a safe + operation because the audiohook's lock is held the entire time, + otherwise this could cause trouble. (closes issue #16522) + Reported by: corruptor + +2010-01-08 19:32 +0000 [r238630] Matthew Nicholson + + * /, main/file.c: Merged revisions 238629 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238629 | mnicholson | 2010-01-08 13:20:44 -0600 (Fri, 08 Jan + 2010) | 5 lines Properly calculate the remaining space in the + output string when reducing format strings. (closes issue #16560) + Reported by: goldwein ........ + +2010-01-08 17:18 +0000 [r238583] Jeff Peeler + + * main/features.c: Stop trying to find a parking space after + traversing the parkinglot one time. (closes issue #16428) + Reported by: Yasuhiro Konishi + +2010-01-07 21:24 +0000 [r238527] Richard Mudgett + + * channels/sig_pri.c: Fix using the wrong pointer type in + do_idle_thread(). + +2010-01-07 20:42 +0000 [r238361-238492] David Vossel + + * main/channel.c: fixes ast_transfer stall until hangup if called + with a channel that doesn't support transfers ast_transfer sets + res to 0 if there is no technology transfer function, but then + tests for it to be negative before deciding to do an early exit. + As a result, it will will wait for an AST_CONTROL_TRANSFER + message that will never come. (closes issue #16424) Reported by: + davidw Patches: Issue_16424_trunk_234134.patch uploaded by davidw + (license 780) + + * /, channels/chan_iax2.c: Merged revisions 238411 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 + Jan 2010) | 10 lines fixes crash in "scheduled_destroy" in + chan_iax A signed short was used to represent a callnumber. This + is makes it possible to attempt to access the iaxs array with a + negative index. (closes issue #16565) Reported by: jensvb + ........ + + * channels/chan_sip.c: Change in sip show channels display format + allowing more digits for CID (closes issue #16459) Reported by: + Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins + (license 953) + + * apps/app_queue.c: cli 'queue show' formatting fix. queue name was + truncated over 12 characters (closes issue #16078) Reported by: + RoadKill Patches: quequename_limit.patch uploaded by ppyy + (license 906) Tested by: dvossel + +2010-01-07 09:14 +0000 [r238313] Tzafrir Cohen + + * configs/sip.conf.sample: Document the usefulness of explicit + udp:// in the register string + +2010-01-06 21:45 +0000 [r238231] Tilghman Lesher + + * /, funcs/func_cdr.c: Merged revisions 238230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010) + | 4 lines Revise documentation on disposition values to the + actual values used. (closes issue #16289) Reported by: wdoekes + ........ + +2010-01-06 20:37 +0000 [r238134-238181] Jeff Peeler + + * apps/app_meetme.c: Fix misreverting from 177158. (closes issue + #15725) Reported by: shanermn Patches: v1-15725.patch uploaded by + dimas (license 88) Tested by: shanermn + + * main/features.c: Fix channel name comparison for bridge + application. The channel name comparison was not comparing the + whole string and therefore if one channel name was a substring of + the other, the bridge would fail. (closes issue #16528) Reported + by: telecos82 Patches: res_features_r236843.diff uploaded by + telecos82 (license 687) + +2010-01-06 16:36 +0000 [r238091] David Vossel + + * include/asterisk/test.h: fixes test.c compile issue when + TEST_FRAMEWORK is not enabled The ast_test_status_update() + function is defined in test.h. When TEST_FRAMEWORK is not enabled + a macro is defined as a no-op place holder for this function. The + macro did not contain the correct number of arguments. This + caused a compile error. Much thanks to wdoekes for reporting the + issue and supplying the patch! + +2010-01-06 15:35 +0000 [r238014] Sean Bright + + * addons/format_mp3.c: Fix reading samples from format_mp3 after + ast_seekstream/ast_tellstream. There is a bug when using + ast_seekstream/ast_tellstream with format_mp3 in that the file + read position is not reset before attempting to read samples. So + when we seek to determine the maximum size of the file (as in + res_agi's STREAM FILE) we weren't then resetting the file pointer + so that we could properly read samples. This patch addresses that + (in a similar manner to format_wav.c). (closes issue #15224) + Reported by: rbd Patches: 20091230_addons_1.4_issue15224.diff + uploaded by seanbright (license 71) Tested by: rbd, seanbright + Review: https://reviewboard.asterisk.org/r/453 + +2010-01-06 15:19 +0000 [r238010] Russell Bryant + + * /, apps/app_mp3.c: Merged revisions 238009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) + | 7 lines Resolve a crash due to an ast_frame not being fully + initialized. (closes issue #16531) Reported by: john8675309 + (closes SWP-615) ........ + +2010-01-06 06:53 +0000 [r237968] Tilghman Lesher + + * channels/chan_sip.c: Whoa, duplicate setting (dead code). + +2010-01-05 23:08 +0000 [r237920] David Vossel + + * apps/app_queue.c: fixes holdtime playback issue in app_queue When + reporting hold time, the number of seconds should be mod 60. + Otherwise audio playback could be something like "2 minutes 123 + seconds" rather than "2 minutes 3 seconds". Also, the "minute" + sound file is missing, so for the moment until that file can be + created the "minutes" file is used instead. (closes issue #16168) + Reported by: nickilo Patches: patch-unified-trunk-rev-222176 + uploaded by nickilo (license ) Tested by: nickilo, wonderg + +2010-01-05 20:56 +0000 [r237882] Mark Michelson + + * apps/app_dial.c: Mismerged a bit. + +2010-01-05 19:29 +0000 [r237839] David Vossel + + * main/pbx.c: fixes subscriptions being lost after 'module reload' + During a module reload if multiple extension configs are present, + such as both extensions.conf and extensions.ael, watchers for one + config's hints will be lost during the merging of the other + config. This happens because hint watchers are only preserved for + the current config being merged. The old context list is + destroyed after the merging takes place, meaning any watchers + that were not perserved will be removed. Now all hints are + preserved during merging regardless of what config file is being + merged. These hints are only restored if they are present within + the new context list. (closes issue #16093) Reported by: jlaroff + +2010-01-05 18:57 +0000 [r237804] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h, channels/sig_pri.c: Removed unused + parameters from analog_available() and sig_pri_available(). + +2010-01-05 18:46 +0000 [r237802-237803] Mark Michelson + + * apps/app_dial.c, CHANGES: Add a missing part of the connected + line work into trunk. Part of the work done for connected line + was to add an optional argument to the 'f' option to allow for + the connected party information of the outgoing channel to be set + to the argument provided. This was overlooked during the merge of + the work to trunk and is being added back now. The CHANGES file + has also been updated to note this change. + + * CHANGES: Spell "aficionado" like someone who isn't stupid. + +2010-01-05 17:26 +0000 [r237699-237749] Russell Bryant + + * main/utils.c: Fix build of utility apps that include utils.c. + + * /, main/utils.c: Merged revisions 237697 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010) + | 7 lines Change a NOTICE log message to DEBUG where it belongs. + (closes issue #16479) Reported by: alexrecarey (closes SWP-577) + ........ + +2010-01-05 16:08 +0000 [r237656] Michiel van Baak + + * apps/app_mixmonitor.c: Make CLI command 'mixmonitor start|stop + work again. (closes issue #16534) Reported by: + jlaguilar Fix as suggested by jlaguilar in the bugreport + +2010-01-04 21:48 +0000 [r237406-237574] Tilghman Lesher + + * /, main/say.c: Merged revisions 237573 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010) + | 6 lines Bounds checking for input string (closes issue #16407) + Reported by: qwell Patches: 20100104__issue16407.diff.txt + uploaded by tilghman (license 14) ........ + + * main/pbx.c, /: Merged revisions 237493 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010) + | 8 lines Regression in issue #15421 - Pattern matching (closes + issue #16482) Reported by: wdoekes Patches: + astsvn-16482-betterfix.diff uploaded by wdoekes (license 717) + 20091223__issue16482.diff.txt uploaded by tilghman (license 14) + Tested by: wdoekes, tilghman ........ + + * main/config.c: Oops, didn't compile (thanks, kpfleming) + + * main/config.c: Further reduce the encoded blank values back to + blank in the realtime API. (closes issue #16533) Reported by: + sergee Patches: 200100104__issue16533.diff.txt uploaded by + tilghman (license 14) Tested by: sergee + + * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged + revisions 237405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) + | 16 lines Add a flag to disable the Background behavior, for AGI + users. This is in a section of code that relates to two other + issues, namely issue #14011 and issue #14940), one of which was + the behavior of Background when called with a context argument + that matched the current context. This fix broke FreePBX, + however, in a post-Dial situation. Needless to say, this is an + extremely difficult collision of several different issues. While + the use of an exception flag is ugly, fixing all of the issues + linked is rather difficult (although if someone would like to + propose a better solution, we're happy to entertain that + suggestion). (closes issue #16434) Reported by: rickead2000 + Patches: 20091217__issue16434.diff.txt uploaded by tilghman + (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by + tilghman (license 14) Tested by: rickead2000 ........ + +2010-01-04 16:39 +0000 [r237327] David Vossel + + * apps/app_queue.c: app_queue segfaults if realtime field uniqueid + is NULL (closes issue #16385) Reported by: haakon Patches: + app_queue.c.patch uploaded by haakon (license 880) + app_queue.c.patch_v2 uploaded by dvossel (license 671) Tested by: + haakon + +2010-01-04 16:24 +0000 [r237323] Jeff Peeler + + * res/res_agi.c: Fix timeout for AGI command speech recognize. + (closes issue #16297) Reported by: semond + +2010-01-04 16:20 +0000 [r237319] Tilghman Lesher + + * channels/chan_local.c, /: Merged revisions 237318 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 + Jan 2010) | 3 lines It's also possible for the Local channel to + directly execute an Application. Reviewboard: + https://reviewboard.asterisk.org/r/452/ ........ + +2010-01-04 07:55 +0000 [r237284] Olle Johansson + + * res/res_pktccops.c, channels/chan_mgcp.c: - Disable res_pktccops + by default - Add dependency in chan_mgcp that was missing - Add a + small amount of doc to the source code + +2010-01-04 03:38 +0000 [r237250] TransNexus OSP Development + + * apps/app_osplookup.c: 1. Added reporting operator names in + AuthReq. 2. Added retrieving operator names from AuthRsp and + exporting them. + +2010-01-02 16:35 +0000 [r237213] Tilghman Lesher + + * channels/chan_sip.c: global_contact_ha was renamed in trunk + +2010-01-02 09:54 +0000 [r237136] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 237135 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 + lines Release memory of the contact acl before unloading module + ........ + +2009-12-30 23:51 +0000 [r237098] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooq931.c, + addons/ooh323c/src/ooCalls.c: small q931 processing and + signalling corrections don't decode UUIE from Q931StatusMessage + clean call without callIdentifier data don't start tcs/msd + exchange procedure after call proceeding received (closes issue + #16365) Reported by: benngard2 Tested by: may213, benngard2 + +2009-12-30 22:30 +0000 [r237050] Jason Parker + + * main/say.c, doc/lang/vietnamese.ods (added), + apps/app_voicemail.c: Add app_voicemail and say.c support for + Vietnamese. Also add an XXX comment that I'm baffled nobody has + ever complained about. We say "first message", and then we go + into language-specific stuff where we proceed to say..."first + message". (closes issue #15053) Reported by: dinhtrung Patches: + vietnamese.ods uploaded by dinhtrung (license 776) + app_voicemail.c.diff uploaded by dinhtrung (license 776) (closes + issue #15626) Reported by: dinhtrung Patches: say.c.diff uploaded + by dinhtrung (license 776) + +2009-12-30 21:59 +0000 [r236982] Tilghman Lesher + + * channels/chan_local.c, /: Merged revisions 236981 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 + Dec 2009) | 9 lines Don't queue frames to channels that have no + means to process them. (closes issue #15609) Reported by: aragon + Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt + uploaded by tilghman (license 14) Tested by: aragon Review: + https://reviewboard.asterisk.org/r/452/ ........ + +2009-12-30 21:09 +0000 [r236893-236902] Jeff Peeler + + * utils/ael_main.c: One more LOW_MEMORY compile fix. + + * channels/chan_sip.c, main/cli.c: Fix compiling with LOW_MEMORY. + Modified handle_verbose to be LOW_MEMORY aware, removed old RTP + related code in chan_sip. (closes issue #16381) Reported by: + michael_iedema Patches: ast_complete_source_filename.patch + uploaded by michael iedema (license 942) modified by me + +2009-12-30 17:53 +0000 [r236802-236847] Tilghman Lesher + + * cdr/cdr_adaptive_odbc.c, cel/cel_adaptive_odbc.c: When the field + is blank, don't warn about the field being unable to be coerced, + just skip the column. (closes + http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html) + Reported by Nic Colledge on the -dev list, fixed by me. + + * channels/chan_sip.c: Shut down the SIP session timers more + gracefully, in order to prevent a possible crash. (closes issue + #16452) Reported by: corruptor Patches: + 20091221__issue16452.diff.txt uploaded by tilghman (license 14) + Tested by: corruptor + +2009-12-29 10:59 +0000 [r236756] TransNexus OSP Development + + * configs/osp.conf.sample, apps/app_osplookup.c, configure.ac: 1. + Updated for OSP Toolkit 3.6.0. 2. Added service type ported + number query. 3. Formated code. + +2009-12-28 22:09 +0000 [r236713] Jason Parker + + * main/ast_expr2.y, main/ast_expr2.c: Allow "REMAINDER" to function + properly in expressions. (closes issue #16427) Reported by: + wdoekes Patches: ast16-reminder-remainder.patch uploaded by + wdoekes (license 717) Tested by: wdoekes + +2009-12-28 17:37 +0000 [r236667] Tilghman Lesher + + * apps/app_voicemail.c: Use recommended option, not deprecated + option. (closes issue #16515) Reported by: ManChicken + +2009-12-28 15:22 +0000 [r236510-236613] Sean Bright + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/threadstorage.h: Merged revisions 236585 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec + 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT + requires extra braces. There was conditional code (based on build + platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that + was removed since it is fixed in newer versions of + Solaris/OpenSolaris, but I am still running into it on Solaris 10 + x86 so add a configure-time check for it. ........ + + * /, apps/app_meetme.c: Merged revisions 236509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec + 2009) | 12 lines Avoid a crash with large numbers of MeetMe + conferences. Similar to changes made to Queue(), when we have + large numbers of conferences in meetme.conf (1000s) and we use + alloca()/strdupa(), we can blow out the stack and crash, so + instead just use a single fixed buffer. (closes issue #16509) + Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded + by seanbright (license 71) Tested by: seanbright ........ + +2009-12-27 18:20 +0000 [r236434] Tilghman Lesher + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236433 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 Dec 2009) + | 2 lines Turn on colors in the daemon, since there's many + requests for it on Ubuntu. ........ + +2009-12-26 15:27 +0000 [r236358] Kevin P. Fleming + + * /, sounds/Makefile: Merged revisions 236357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec + 2009) | 1 line update to latest releases with zero uid/gid + ........ + +2009-12-23 19:17 +0000 [r236304-236312] David Vossel + + * CHANGES: Update CHANGES to reflect new QUEUE_MEMBER option, + "ready" + + * apps/app_queue.c: QUEUE_MEMBER(..., ready) counts only ready + agents, not free agents wrapping up The QUEUE_MEMBER dialplan + function can return total members, logged-in members and "free" + members count. A member is counted as "free" immediately after + his call ends, even though its wrap-up time, if specified in + queues.conf, has not yet expired, and the queue will not actually + route a call to it. This Patch introduces a new "ready" option + that only counts free agents no longer in the wrap up time + period. (closes issue #16240) Reported by: kkm Patches: + appqueue-memberfun-readyoption-trunk.diff uploaded by kkm + (license 888) Tested by: kkm, dvossel + + * CHANGES, apps/app_queue.c: update CHANGES to reflect new 'R' + app_queue option plus a minor optimization to the feature patch + (issue #16384) + + * apps/app_queue.c: new parameter 'R' to the Queue application The + 'R' argument stops moh and indicates ringing once the agent is + ringing. This allows the person in the queue to know their call + is potentially about to be answered. (closes issue #16384) + Reported by: haakon Patches: new_app_queue.c.patch uploaded by + haakon (license 880) Tested by: haakon, loloski, dvossel + +2009-12-23 18:25 +0000 [r236183-236300] Tilghman Lesher + + * apps/app_stack.c: AGI may be invoked from outside the dialplan + (closes issue #16510) Reported by: atis Patches: + 20091223__issue16510.diff.txt uploaded by tilghman (license 14) + Tested by: atis + + * /, res/res_agi.c: Merged revisions 236184 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009) + | 4 lines If EXEC only gets a single argument, don't crash when + the second is used. (closes issue #16504) Reported by: bklang + ........ + + * include/asterisk/test.h: Allow test_heap.c to compile when + AST_DEVMODE is true, but TEST_FRAMEWORK is false + + * apps/app_voicemail.c: Actually use tmp for something (brings + trunk back into sync with 1.6 branches). + +2009-12-22 21:53 +0000 [r236027-236144] David Vossel + + * channels/chan_iax2.c: fixes iax "can't compress subclass + 4294967295" error (closes issue #16456) Reported by: dvossel + Tested by: dvossel + + * /, channels/chan_sip.c: Merged revisions 236062 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) + | 11 lines fixes issue with p->method incorrectly set to ACK It + is possible for a second ACK to come in for a retransmitted + message. If an ack does not match an unacked message in our + queue, restore the previous p->method as this ACK is completely + ignored. (closes issue #16295) Reported by: omolenkamp Patches: + issue16295_v2.diff uploaded by dvossel (license 671) ........ + + * CHANGES: update CHANGES to reflect the addition of the test + framework + + * include/asterisk/test.h (added), build_tools/cflags-devmode.xml, + tests/test_heap.c, main/test.c (added), + include/asterisk/_private.h, main/asterisk.c: Unit Test Framework + API The Unit Test Framework is a new API that manages + registration and execution of unit tests in Asterisk with the + purpose of verifying the operation of C functions. The Framework + consists of a single test manager accompanied by a list of + registered test functions defined within the code. A test is + defined, registered, and unregistered from the framework using a + set of macros which allow the test code to only be compiled + within asterisk when the TEST_FRAMEWORK flag is enabled in + menuselect. This allows the test code to exist in the same file + as the C functions it intends to verify. Registered tests may be + viewed and executed via a set of new CLI commands. CLI commands + are also present for generating and exporting test results into + xml and txt formats. For more information and use cases please + refer to the documentation provided at the beginning of the + test.h file. Review: https://reviewboard.asterisk.org/r/447/ + +2009-12-21 19:54 +0000 [r235941] Jeff Peeler + + * /, res/res_monitor.c: Merged revisions 235940 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) + | 13 lines Change Monitor to not assume file to write to does not + contain pathing. 227944 changed the fname_base argument to always + append the configured monitor path. This change was necessary to + properly compare files for uniqueness. If a full path is given + though, nothing needs to be appended and that is handled + correctly now. (closes issue #16377) (closes issue #16376) + Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch + uploaded by dant (license 670) ........ + +2009-12-21 18:51 +0000 [r235904] Kevin P. Fleming + + * contrib/upstart/asterisk.upstart-0.3.9, include/asterisk/cel.h, + main/say.c, include/asterisk/channel.h, + include/asterisk/manager.h, channels/sig_pri.c, + include/asterisk/logger.h, include/asterisk/http.h, + include/asterisk/callerid.h, include/asterisk/syslog.h, + channels/chan_dahdi.c, include/asterisk/app.h, + include/asterisk/doxyref.h, include/asterisk/event.h, + channels/sig_analog.c, channels/chan_misdn.c, + contrib/upstart/asterisk.user.conf, + include/asterisk/rtp_engine.h, + include/asterisk/security_events.h, + include/asterisk/stringfields.h: Change all refererences to 1.6.3 + to be 1.8, since that will be the next feature release + +2009-12-21 17:00 +0000 [r235822] Tilghman Lesher + + * /, main/features.c: Merged revisions 235821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009) + | 8 lines Send parking lot announcement to the channel which + parked the call, not the park-ee. (closes issue #16234) Reported + by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded + by tilghman (license 14) 20091221__issue16234__1.4.diff.txt + uploaded by tilghman (license 14) Tested by: yeshuawatso ........ + +2009-12-20 08:22 +0000 [r235740-235774] Alec L Davis + + * main/dsp.c: restarts busydetector (if enabled) when DTMF is + received after call is bridged. (closes issue 0016389) Reported + by: alecdavis Tested by: alecdavis Patch + dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585) + + * apps/app_dial.c, CHANGES: app_dial optional parameter to option + 'r' to allow play indication from indications.conf (closes issue + #14504) Reported by: alecdavis Tested by: alecdavis,jsmith Patch + app_dial.play_ring_indications.diff7.txt uploaded by alecdavis + (license 585) + +2009-12-18 22:51 +0000 [r235660] Jeff Peeler + + * main/channel.c, /, include/asterisk/cdr.h: Merged revisions + 235635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) + | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is + simple in that it reorders the disposition defines so that the + fix for issue 12946 works properly (the default CDR disposition + was changed to AST_CDR_NOANSWER). Also, the + AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all + CDR records are written. The side effects of CDR changes are + scary, so I'm documenting the test cases performed to attempt to + catch any regressions. The following tests were all performed + using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls + B (busy) Hangup C Hangup A (Both SIP and features) A calls B A + blind transfers to C Hangup C (Both SIP and features) A calls B A + attended transfers to C Hangup C A calls B A attended transfers + to C (SIP) C blind transfers to A (features) Hangup A All of the + test scenario CDRs matched. The following tests were performed + just with the patch to ensure proper operation (with + unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten + =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) + (closes issue #16180) Reported by: aatef Patches: bug16180.patch + uploaded by jpeeler (license 325) ........ + +2009-12-18 22:40 +0000 [r235573-235656] Tilghman Lesher + + * /, configure, configure.ac: Merged revisions 235652 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18 + Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion + ........ + + * /, configure, configure.ac: Merged revisions 235572 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 + Dec 2009) | 2 lines Point to the typical missing package, not the + cryptic "termcap support". ........ + +2009-12-17 23:21 +0000 [r235521] Joshua Colp + + * channels/chan_sip.c: Remove some old code for going to the 'fax' + extension when a T.38 switchover occurs. This would have already + happened when we detected the CNG tone so this was basically a + noop. + +2009-12-17 17:19 +0000 [r235422] Tilghman Lesher + + * main/pbx.c, /: Merged revisions 235421 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235421 | tilghman | 2009-12-17 11:17:51 -0600 (Thu, 17 Dec 2009) + | 8 lines Use context from which Macro is executed, not macro + context, if applicable. Also, ensure that the extension COULD + match, not just that it won't match more. (closes issue #16113) + Reported by: OrNix Patches: 20091216__issue16113.diff.txt + uploaded by tilghman (license 14) Tested by: OrNix ........ + +2009-12-17 00:52 +0000 [r235342-235382] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_analog.c: Fix call forwarding + for analog phones. (closes issue #16440) Reported by: mmichelson + + * configs/jabber.conf.sample, include/asterisk/jabber.h, CHANGES, + res/res_jabber.c: Add auth_policy option to jabber.conf for auto + user registration. The option is global and currently the + acceptable values as noted in the sample config are accept or + deny. (closes issue #15228) Reported by: lp0 + +2009-12-16 05:24 +0000 [r235298] Jared Smith + + * /, configs/sip.conf.sample: Merged revisions 235181 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15 + Dec 2009) | 4 lines Add a line showing that we can use CIDR + notation. patch by jsmith, after discussion with jtodd ........ + +2009-12-16 00:31 +0000 [r235265] Jeff Peeler + + * main/manager.c, CHANGES: Enhance AMI redirect to allow channels + to be redirected to different places. New parameters + ExtraContext, ExtraExtension, and ExtraPriority have been added + to redirect the second channel to a different location. + Previously, it was only possible to redirect both channels to the + same place. (closes issue #15853) Reported by: haakon Patches: + trunk-manager.c.patch uploaded by haakon (license 880) Tested by: + jpeeler + +2009-12-15 23:51 +0000 [r235229] Tilghman Lesher + + * include/asterisk/strings.h: Is it Friday yet? + +2009-12-15 23:41 +0000 [r235226] Jeff Peeler + + * main/channel.c: Change match criteria existence in + ast_channel_cmp_cb to use ast_strlen_zero. (closes issue #16161) + Reported by: may213 Patches: core-show-channel.patch uploaded by + may213 (license 454) + +2009-12-15 18:43 +0000 [r235132] David Vossel + + * channels/chan_sip.c: reverse minor sip registration regression A + registration regression caused by a code tweak in (issue #14331) + and a bug fix in (issue #15539) caused some sip registration + config entries to be constructed incorrectly. Origially issue + #14331 contained the code tweak as well as a bug fix, but since + the issue was reported as a tweak the bug fix portion was moved + into issue #15539. Both the tweak and the bug fix contained minor + incorrect logic that resulted in some SIP registrations to fail. + (issue #14331) (issue #15539) + +2009-12-15 15:33 +0000 [r235053] Tilghman Lesher + + * /, res/res_agi.c: Merged revisions 235052 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235052 | tilghman | 2009-12-15 09:29:24 -0600 (Tue, 15 Dec 2009) + | 4 lines Mandatory argument checking (closes issue #16446) + Reported by: nicchap ........ + +2009-12-15 14:35 +0000 [r235010] Kevin P. Fleming + + * apps/app_fax.c: spandsp does in fact support V.17 modulation at + 14.4 kilobits per second, so we should generate T38MaxBitRate of + 14400 (even though that doesn't really affect the FAX + transmission much at all) + +2009-12-15 07:18 +0000 [r234855-234976] Alec L Davis + + * apps/app_directory.c: Support option 'n', as applications like + Playback, Background etc. Suggested on asterisk-dev as trivial + application change. Reported by: alecdavis Tested by: alecdavis + + * main/dsp.c: Whitespace. + + * main/dsp.c: restarts busydetector (if enabled) when DTMF is + received. (closes issue #16389) Reported by: alecdavis Tested by: + alecdavis Patch dtmf_busydetector.diff.txt uploaded by alecdavis + (license 585) + + * apps/app_directory.c: fixes escape to extensions 'o' and 'a', for + digits '0' and '*' (closes issue #16437) Reported by: alecdavis + Tested by: alecdavis Patch extension_o_a_fix.diff.txt uploaded by + alecdavis (license 585) + + * apps/app_directory.c: ast_stream_and_wait(chan,dir-usingkeypad) + didn't capture the dialled DTMF. (closes issue #16409) Reported + by: alecdavis Tested by: alecdavis Patch bug_16409.diff.txt + uploaded by alecdavis (license 585) + +2009-12-14 23:16 +0000 [r234820] Tilghman Lesher + + * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: + Allow greetings-only mailboxes for Voicemail. (closes issue + #15132) Reported by: floletarmo Patches: voicemail_changes.patch + uploaded by floletarmo (license 784) (with some additional + changes by me) + +2009-12-14 21:32 +0000 [r234776] Jason Parker + + * apps/app_readexten.c: Allow tonelist as argument to ReadExten. + ReadExten already supported playing a tonezone from + indications.conf. It now has the ability to use a tonelist like + 440+480/2000|0/4000 (closes issue #15185) Reported by: jcovert + Patches: app_readexten.c.patch uploaded by jcovert (license 551) + Tested by: qwell Patch modified by me, to maintain backwards + compatibility. + +2009-12-14 21:13 +0000 [r234700] Tilghman Lesher + + * /, build_tools/make_version_c, build_tools/make_version_h: Merged + revisions 234699 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r234699 | tilghman | 2009-12-14 15:09:56 -0600 (Mon, 14 Dec 2009) + | 5 lines Deal with the situation where .flavor exists but + .version does not. Also make the script slightly more portable, + in keeping with autoconf syntax. (closes issue #14737) Reported + by: davidw ........ + +2009-12-14 17:19 +0000 [r234631] Leif Madsen + + * doc/tex/imapstorage.tex, /: Update IMAP build documentation. + Update the IMAP build documentation to show how to build on + 64-bit platforms. (issue #16433) Reported by: shrift Tested by: + lmadsen + +2009-12-14 16:08 +0000 [r234572] Sean Bright + + * main/timing.c: The default rate for 'timing test' is actually + 50/sec, not 100/sec as advertised. + +2009-12-14 10:46 +0000 [r234526] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 234492 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8 + lines Stop sending 183's after call hangup. There where still + cases where the 183 keep-alive mechanism would not stop sending + 183's even though the Asterisk server had sent a final reply to + the invite. EDVX-28 ........ + +2009-12-13 09:41 +0000 [r234458] Tilghman Lesher + + * main/pbx.c: Trim leading/trailing spaces from the filename, to + deal with common user error. + +2009-12-11 23:17 +0000 [r234380] Jeff Peeler + + * /, apps/app_meetme.c: Merged revisions 234379 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009) + | 11 lines Fix talking detection status after conference user is + muted. This patch ensures that when a conference user is muted + that the accompanying AMI Meetme talking off event is sent. Also, + the meetme list output is updated to show the muted user as + unmonitored. (closes issue #16247) Reported by: dimas Patches: + v3-16247.patch uploaded by dimas (license 88) ........ + +2009-12-10 21:01 +0000 [r234256] Jason Parker + + * Makefile, /: Merged revisions 234255 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r234255 | qwell | 2009-12-10 14:58:09 -0600 (Thu, 10 Dec 2009) | + 9 lines Fix unselecting of menuselect options via GLOBAL_MAKEOPTS + and USER_MAKEOPTS. (closes issue #16296) Reported by: abelbeck + Patches: issue16296-20091210.diff uploaded by qwell (license 4) + (abelbeck described a fix, which I expanded upon) Tested by: + abelbeck, qwell, lmadsen ........ + +2009-12-10 18:56 +0000 [r234210] Tilghman Lesher + + * res/res_musiconhold.c: Missed a case that emits a WARNING where + none is warranted. + +2009-12-10 17:31 +0000 [r234173] Jeff Peeler + + * apps/app_meetme.c, apps/app_page.c, main/app.c, CHANGES: Add + audio announcement option to app_page As described in the CHANGES + file: * MeetMe has a new option 'G' to play an announcement + before joining a conference. * Page has a new option 'A(x)' which + will playback an announcement simultaneously to all paged phones + (and optionally excluding the caller's one using the new option + 'n') before the call is bridged. To add the new option to meetme, + the conference flag options had to be extended to 64 bits. + (closes issue #14365) Reported by: dferrer Patches: + page_announce.patch uploaded by dferrer (license 525) modified by + me Review: https://reviewboard.asterisk.org/r/188/ + +2009-12-10 16:24 +0000 [r234129] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 234095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009) + | 9 lines When we receive no response at all to our INVITE, allow + the channel to be destroyed. (closes issue #15627) Reported by: + falves11 Patches: 20091209__issue15627__1.6.0.diff.txt uploaded + by tilghman (license 14) 20091209__issue15627__1.4.diff.txt + uploaded by tilghman (license 14) Tested by: falves11 Review: + https://reviewboard.asterisk.org/r/446/ (closes issue #15716) + Reported by: dant (closes issue #16270) Reported by: corruptor + (closes issue #15356) Reported by: falves11 (issue #16382) + Reported by: lftsy ........ + +2009-12-09 23:35 +0000 [r233967-234055] Russell Bryant + + * UPGRADE.txt, CHANGES: Move an entry from CHANGES to UPGRADE.txt. + + * UPGRADE.txt, CHANGES: Move an entry from CHANGES that should be + in UPGRADE.txt. + + * CHANGES: Provide a real description of LOCAL_PEEK(). + + * CHANGES: Remove a feature from CHANGES that was listed twice for + 1.6.2. + + * CHANGES: Fix up the faxdetect entry in CHANGES. This feature was + listed as a 1.6.2 feature, even though it's in all 1.6.X + versions. The description of the feature was also no longer + accurate. + + * CHANGES: Remove an entry from CHANGES that is already in + UPGRADE.txt (where it should be). + +2009-12-08 18:40 +0000 [r233718-233732] Tilghman Lesher + + * addons/res_config_mysql.c: Typo pointed out on #asterisk-dev (by + atis_work) + + * res/res_musiconhold.c: Find another ref leak and change how we + manage module references. (closes issue #16388, closes issue + #16279, closes issue #16390) Reported by: parisioa Patches: + 20091208__issue16388.diff.txt uploaded by tilghman (license 14) + Tested by: parisioa, tilghman Review: + https://reviewboard.asterisk.org/r/442/ + +2009-12-08 18:00 +0000 [r233692] Russell Bryant + + * formats/format_sln.c, formats/format_wav.c, + formats/format_ogg_vorbis.c, formats/format_sln16.c, + formats/format_wav_gsm.c, formats/format_siren7.c, + formats/format_ilbc.c, formats/format_vox.c, + formats/format_pcm.c, formats/format_h263.c, + formats/format_g723.c, formats/format_h264.c, + formats/format_g726.c, formats/format_siren14.c, + formats/format_jpeg.c, formats/format_gsm.c, + formats/format_g729.c: Set a module load priority for format + modules. A recent change to app_voicemail made it such that the + module now assumes that all format modules are available while + processing voicemail configuration. However, when autoloading + modules, it was possible that app_voicemail was loaded before the + format modules. Since format modules don't depend on anything, + set a module load priority on them to ensure that they get loaded + first when autoloading. This fix applies to trunk, 1.6.1, and + 1.6.2. The fix for 1.4 and 1.6.0 will require a different + approach since the module load priority functionality is not + present in the module API. (issue #16412) Reported by: jiddings + +2009-12-07 23:28 +0000 [r233611] David Vossel + + * main/utils.c: fixes incorrect logic in ast_uri_encode issue + #16299 + +2009-12-07 23:10 +0000 [r233577] Atis Lezdins + + * contrib/valgrind.supp: Fix compatibility with valgrind 3.3 and + older. (noticed in issue #16388) Reported by: parisioa Patches: + valgrind.supp uloaded by atis (license 242) Tested by: atis, + parisioa + +2009-12-07 19:48 +0000 [r233545] David Ruggles + + * apps/app_externalivr.c: Fix TCP Client interface Fix a couple of + very minor bugs that prevent the socket client from working. The + wrong set of properties were used in one place and the size of + the address variable isn't set if the host name is an ip address. + Also includes a fix for a bug that was introduced previously. + (closes issue #16121) Reported by: thedavidfactor Tested by: + thedavidfactor Review: https://reviewboard.asterisk.org/r/439/ + +2009-12-07 18:08 +0000 [r233472] David Vossel + + * /, channels/chan_sip.c: Merged revisions 233471 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) + | 9 lines fixes missing Contact header angle brackets (closes + issue #16298) Reported by: mgernoth Patches: + reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested + by: dvossel ........ + +2009-12-07 17:59 +0000 [r233468] Jeff Peeler + + * include/asterisk/jabber.h, CHANGES, res/res_jabber.c: Add + applications JabberJoin, JabberLeave, JabberSendGroup for XMPP + groupchat (closes issue #14352) Reported by: fiddur Patches: + trunk-14352-2.diff uploaded by phsultan (license 73) Tested by: + fiddur + +2009-12-07 16:14 +0000 [r233394] Matthew Nicholson + + * channels/chan_sip.c: Do not reject SDP packets describing only + non audio streams. (closes issue #16387) Reported by: zalex1953 + Patches: media-level-c-fix1.diff uploaded by mnicholson (license + 96) Tested by: mnicholson, zalex1953 + +2009-12-06 07:01 +0000 [r233358] Tilghman Lesher + + * include/asterisk/compat.h, main/strcompat.c, main/app.c: Move + implementation of closefrom(3) from app.c to strcompat.c + +2009-12-04 21:54 +0000 [r233280] David Vossel + + * configs/iax.conf.sample, /: Merged revisions 233279 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 + Dec 2009) | 7 lines clarify requirecalltoken option in + iax.sample.conf (closes issue #16223) Reported by: bklang + Patches: clarify-iax-requirecalltoken.patch uploaded by bklang + (license 919) ........ + +2009-12-04 21:06 +0000 [r233239] Tilghman Lesher + + * main/translate.c: Using the builtin function breaks OpenBSD 4.2 + (closes issue #16395) Reported by: jtodd + +2009-12-04 20:21 +0000 [r233121-233235] David Vossel + + * CHANGES: update CHANGES file for .m3u support in Mp3Player + application + + * apps/app_mp3.c: .m3u support for Mp3Player app (closes issue + #14823) Reported by: macli Patches: app_mp3.diff1 uploaded by + macli (license ) Tested by: macli, dvossel + + * CHANGES: update CHANGES for new queue option, + penaltymemberslimit. + + * apps/app_queue.c: changes penaltymemberslimit to use scanf for + config value parsing + + * configs/queues.conf.sample, apps/app_queue.c: new queue option, + penaltymemberslimit, disregards penalty on too few queue members + when enabled (closes issue #14559) Reported by: fiddur Patches: + trunk-199584-1.diff uploaded by fiddur (license 678) Tested by: + fiddur, dvossel + + * /, apps/app_voicemail.c: Merged revisions 233116 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 + Dec 2009) | 6 lines document and rename strip_control() in + app_voicemail (closes issue #16291) Reported by: wdoekes ........ + +2009-12-04 17:18 +0000 [r233100] Russell Bryant + + * main/channel.c, /: Merged revisions 233092 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009) + | 7 lines Only do frame payload check for HOLD frames. This code + was added for helping to debug the source of invalid HOLD frames. + However, a side effect of this is that it will incorrectly report + errors for frames that have an integer payload. Make the check + for this block specific to the HOLD frame case. ........ + +2009-12-04 17:15 +0000 [r233093] Matthias Nick + + * pbx/pbx_config.c: Parse global variables or expressions in hint + extensions Parse global variables or expressions in hint + extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)} + (closes issue #16166) Reported by: rmudgett Tested by: mnick, + rmudgett + +2009-12-04 16:55 +0000 [r233059-233089] Michiel van Baak + + * channels/chan_skinny.c: Let's unlock the lines list after the + AST_LIST_TRAVERSE instead of inside it. + + * channels/chan_skinny.c: Only assign line and device in + handle_transfer_button when we have a subchannel. (closes issue + #16040) Reported by: ebroad + +2009-12-04 16:08 +0000 [r233050] Tilghman Lesher + + * addons/res_config_mysql.c: Update the mysql driver to always + return NULL columns, as this is needed for the realtime API to + work correctly. (closes issue #16138) Reported by: sohosys + Patches: 20091029__issue16138.diff.txt uploaded by tilghman + (license 14) Tested by: sohosys + +2009-12-04 15:38 +0000 [r233046] Matthias Nick + + * /, main/dsp.c: Merged revisions 233014 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) | + 11 lines Warning message gets displayed only once Added + additional field 'int display_inband_dtmf_warning', which when + set to '1' displays the warning ('Inband DTMF is not supported on + codec %s. Use RFC2833'), and when set to '0' doesn't display the + warning. Otherwise you would get hundreds of warnings every + second. (closes issue #15769) Reported by: falves11 Patches: + patch_15769_14.txt uploaded by mnick (license 874) Tested by: + mnick, falves11 ........ + +2009-12-04 05:26 +0000 [r232854-232982] Tilghman Lesher + + * res/res_pktccops.c: Buildbot complained + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + res/res_pktccops.c: OS X does not define MSG_NOSIGNAL, but it + does have a socket option SO_NOSIGPIPE. (closes issue #16178) + Reported by: oej + + * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add + pagerdateformat, to allow shorter dates for SMS messages. (closes + issue #16263) Reported by: andrew Patches: pagerdate.patch + uploaded by andrew (license 240) (with a slight modification by + me) + + * /, apps/app_voicemail.c: Merged revisions 232820 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 + Dec 2009) | 8 lines Deprecate "cz" in favor of "cs". Also, change + the use of language codes so that language registers as a prefix, + rather than an exact match. (closes issue #16272) Reported by: + patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by + tilghman (license 14) ........ + +2009-12-03 20:26 +0000 [r232853] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c, + addons/ooh323c/src/ooh245.c: jitterbuffer setup correction + correction of double pointer references from previous rev + +2009-12-03 08:47 +0000 [r232738-232771] TransNexus OSP Development + + * apps/app_osplookup.c: Replaced two deprecated functions of OSP + Toolkit. + + * apps/app_osplookup.c: Added custom info support. + +2009-12-03 00:38 +0000 [r232700] Jeff Peeler + + * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: + Extend voicemail to allow IMAP folders to be specified per + mailbox. Previously only possible per context, new option called + imapfolder. (closes issue #14298) Reported by: jablko Patches: + patch-200906202 uploaded by jablko (license 675) + +2009-12-03 00:09 +0000 [r232660-232661] Tilghman Lesher + + * res/res_musiconhold.c: Remove debugging line + + * include/asterisk/astobj2.h, res/res_musiconhold.c: Fix multiple + issues with musiconhold, which led to classes not getting + destroyed properly. * Classes are now tracked past removal from + the core container, and module removal is actively prevented + until all references are freed. * A hanging reference stored in + the channel has been removed. This could have caused a mismatch + and the music state not properly cleared, if two or more reloads + occurred between MOH being stopped and MOH being restarted. * In + certain circumstances, duplicate classes were possible. * A race + existed at reload time between a process being killed and the + thread responsible for reading from the related pipe respawning + that process. * Several reference counts have also been + corrected. At least one could have caused deleted classes to + stick around forever, consuming resources. This originally + manifested as MOH external processes that were not killed at + reload time. (closes issue #16279, closes issue #16207) Reported + by: parisioa, dcabot Patches: 20091202__issue16279__2.diff.txt + uploaded by tilghman (license 14) Tested by: parisioa, tilghman + +2009-12-02 23:27 +0000 [r232657] David Vossel + + * UPGRADE.txt, CHANGES: update CHANGES and UPGRADE.txt for early + media behavior change between 1.6.1 and 1.6.2 (closes issue + #16212) Reported by: miki + +2009-12-02 22:17 +0000 [r232587] David Ruggles + + * apps/app_externalivr.c: Prevent double closing of FDs by EIVR + This caused a problem when asterisk was under heavy load and + running both AGI and EIVR applications. EIVR would close an FD at + which point it would be considered freed and be used by a new AGI + instance the second close would then close the FD now in use by + AGI. (closes issue #16305) Reported by: diLLec Tested by: + thedavidfactor, diLLec Review: + https://reviewboard.asterisk.org/r/436/ + +2009-12-02 22:02 +0000 [r232582] Jeff Peeler + + * main/manager.c, /: Merged revisions 232581 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009) + | 7 lines Send ack (response/message) after receiving manager + action userevent (closes issue #16264) Reported by: dimas + Patches: event-ack.patch uploaded by dimas (license 88) ........ + +2009-12-02 21:37 +0000 [r232580] Matthew Nicholson + + * addons/chan_mobile.c: Fix support for multiline SMS messages in + chan_mobile. (closes issue #16278) Reported by: Artem Patches: + multiline-sms-fix2.diff uploaded by mnicholson (license 96) + Tested by: Artem + +2009-12-02 21:32 +0000 [r232576] Jeff Peeler + + * main/manager.c: Make manager response to "Action: events" finish + with empty line (closes issue #16275) Reported by: vnovy Patches: + manager.c.diff uploaded by vnovy (license 922) + +2009-12-02 21:13 +0000 [r232544] Matthew Nicholson + + * addons/chan_mobile.c: Do something with the service indicator so + that asterisk does not attempt to use a chan_mobile endpoint that + does not have service. (closes issue #16132) Reported by: nikkk + Patches: service-indicator2.diff uploaded by mnicholson (license + 96) Tested by: nikkk + +2009-12-02 20:10 +0000 [r232442-232510] Joshua Colp + + * CHANGES, main/asterisk.c, doc/asterisk.sgml: Add an 'X' option to + the asterisk application which enables #exec for configuration + files. This option can be used to enable #exec support in the + asterisk.conf configuration file. (closes issue #16260) Reported + by: atis Patches: exec_includes.patch uploaded by atis (license + 242) + + * apps/app_record.c, CHANGES: Add an option to Record which enables + a mode where any DTMF digit will terminate recording. (closes + issue #15436) Reported by: Vince Patches: app_record.diff + uploaded by Vince (license 823) Tested by: dbrooks + +2009-12-02 17:18 +0000 [r232365] Mark Michelson + + * channels/chan_sip.c: Do not change the exten string field or + rebuild the contact header on an inbound sip_pvt if the outbound + call is redirected. + +2009-12-02 17:06 +0000 [r232356] Joshua Colp + + * /, apps/app_amd.c: Merged revisions 232355 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 + lines Fix a bug where if you hung up very quickly after calling + AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG. + (closes issue #16239) Reported by: CGMChris ........ + +2009-12-02 17:00 +0000 [r232351] David Vossel + + * /, main/acl.c: Merged revisions 232350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009) + | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in + strace. (closes issue #16290) Reported by: wdoekes ........ + +2009-12-02 16:40 +0000 [r232345] Joshua Colp + + * channels/chan_sip.c: Add support for handling the 415 Unsupported + media type response like we do for a 488 Not acceptable here + response. (closes issue #16186) Reported by: atis Patches: + sip_t38_response_415.patch uploaded by atis (license 242) + +2009-12-02 15:42 +0000 [r232269] David Vossel + + * funcs/func_groupcount.c, /: Merged revisions 232268 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 + Dec 2009) | 9 lines fixes segfault in func_groupcount closes + issue #16337) Reported by: Parantido Patches: issue_16337.diff + uploaded by dvossel (license 671) Tested by: Parantido, dvossel + ........ + +2009-12-02 14:54 +0000 [r232230] Joshua Colp + + * channels/chan_sip.c: Fix a bug where a scheduled item ID would + get retained on registrations in a certain scenario causing code + to execute during reload that should not. (issue AST-263) + +2009-12-02 03:26 +0000 [r232164] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, + include/asterisk/compat.h, main/strcompat.c, configure.ac: So + apparently, some platforms don't have ffsll(3). The manpage lies; + it says that the function is in POSIX, but that's only for + ffs(3), not ffsll(3). + +2009-12-02 00:45 +0000 [r232091] Jeff Peeler + + * channels/chan_dahdi.c, /: Merged revisions 232090 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 + Dec 2009) | 10 lines Do not modify the gain settings on data + calls. (The digital flag actually represents a data call.) + (closes issue #15972) Reported by: udosw Patches: + transcap_digital_fix.diff.txt uploaded by alecdavis (license 585) + Tested by: alecdavis ........ + +2009-12-01 23:56 +0000 [r232008-232017] Russell Bryant + + * main/translate.c: Use __builtin_ffsll() from gcc instead of + ffssll() to fix a FreeBSD build error. + + * funcs/func_lock.c: Fix a build error on FreeBSD. + + * /, main/file.c: Merged revisions 232007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) + | 2 lines Fix a warning pointed out by buildbot. ........ + +2009-12-01 21:54 +0000 [r231927] Jeff Peeler + + * main/channel.c, /: Merged revisions 231911 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009) + | 12 lines Fix crash with invalid frame data The crash was + happening as a result of a frame containing an invalid data + pointer, but was set with data length of zero. The few times the + issue was reproduced it _seemed_ that the frame was queued + properly, that is the data pointer was set to NULL. I never could + reproduce the crash so as a last resort the crash has been fixed, + but a check in __ast_read has been added to give as much + information about the source of problematic frames in the future. + (closes issue #16058) Reported by: atis ........ + +2009-12-01 21:20 +0000 [r231867] David Vossel + + * main/pbx.c, /: Merged revisions 231853 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009) + | 3 lines WaitExten m option with no parameters generates frame + with zero datalen but non-null data ptr ........ + +2009-12-01 20:27 +0000 [r231814-231850] Tilghman Lesher + + * res/res_rtp_asterisk.c, channels/chan_unistim.c, + main/rtp_engine.c, addons/chan_ooh323.c, channels/chan_sip.c, + res/res_adsi.c, addons/chan_ooh323.h, + include/asterisk/callerid.h, channels/chan_phone.c, + channels/chan_dahdi.c, channels/chan_skinny.c, main/callerid.c, + channels/chan_h323.c, addons/ooh323cDriver.c, + include/asterisk/rtp_engine.h, addons/ooh323cDriver.h: More + 32->64 bit codec conversions. In the process of swapping ULAW to + a place in the extended codec space, we found several unhandled + cases, where a 32-bit integer was still being used to handle a + codec field. Most of these have been fixed with this commit, + although there is at least one case (codec_dahdi) which depends + upon outside headers to be altered before a conversion can be + made. (Fixes AST-278, SWP-459) + + * include/asterisk/mod_format.h: Formats need to be able to + represent all 64 codec bits. + +2009-12-01 15:47 +0000 [r231741] Matthew Nicholson + + * /, main/file.c: Merged revisions 231740 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec + 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce() + and return an error if no know formats are found. ........ + +2009-11-30 21:47 +0000 [r231692] Kevin P. Fleming + + * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h: + Another round of UDPTL stack fixes/improvements: 1) Allow users + of UDPTL stack to associate a character-string tag with a UDPTL + session, so that log/error/debug messages generated by the UDPTL + stack can be 'connected' to the endpoint that caused them to be + generated. 2) Improve comments (and process) of calculating the + far end's maximum IFP size when redundancy mode is in use for + error correction. 3) When an IFP larger than the calculated 'far + max IFP' size is presented for writing, truncate it rather than + putting in the buffer and allowing the buffer to overflow; this + will cause the ends to retrain to a lower bit rate that produces + IFPs of an appropriate size if possible, and if not possible, the + FAX transfer will fail completely. In these cases, it is due to + the one endpoint supplying a T38FaxMaxDatagram value that is + improperly calculated and is too low to be of use; we have + configuration options available to override this behavior. 4) + Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no + longer needed. + +2009-11-30 21:31 +0000 [r231616-231688] Matthew Nicholson + + * include/asterisk/file.h, /, main/file.c, main/app.c, + apps/app_voicemail.c: Merged revisions 231614 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov + 2009) | 8 lines Remove duplicate entries from voicemail format + lists. This prevents app_voicemail from entering an infinite loop + when the same format is specified twice in the format list. + (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson + Review: https://reviewboard.asterisk.org/r/429/ ........ + + * include/asterisk/file.h, /, main/app.c, apps/app_voicemail.c: + Reverted 231616 + + * include/asterisk/file.h, /, main/app.c, apps/app_voicemail.c: + Merged revisions 231614 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov + 2009) | 8 lines Remove duplicate entries from voicemail format + lists. This prevents app_voicemail from entering an infinite loop + when the same format is specified twice in the format list. + (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson + Review: https://reviewboard.asterisk.org/r/429/ ........ + +2009-11-30 20:44 +0000 [r231602] Joshua Colp + + * channels/chan_sip.c: When receiving SDP that matches the version + of the last one do not treat it as a fatal error. (closes issue + #16238) Reported by: seandarcy + +2009-11-30 18:55 +0000 [r231491-231556] David Vossel + + * apps/app_queue.c: app_queue crashes randomly, often during + call-transfers This patch adds a ref to the queue_ent object's + parent call_queue in queue_exec() so the call_queue won't be + destroyed while the the queue_ent still holds a pointer to it. + (closes issue 0015686) Tested by: dvossel, aragon + + * res/res_rtp_asterisk.c, /: Merged revisions 231441 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 + Nov 2009) | 11 lines fixes crash caused by RTP comfort noise + payload greater than 24 bytes AST-2009-010 (closes issue #16242) + Reported by: amorsen Patches: issue16242.diff uploaded by oej + (license 306) Tested by: amorsen, oej, dvossel ........ + +2009-11-30 16:53 +0000 [r231439] Tilghman Lesher + + * main/asterisk.dynamics (added), Makefile.rules: Export dynamic + (weak-linked) symbols correctly. (closes issue #15193) Reported + by: eliel Patches: 20091111__issue15193.diff.txt uploaded by + tilghman (license 14) + +2009-11-30 16:29 +0000 [r231436] Joshua Colp + + * channels/chan_sip.c: Fix a bug where an immediate masquerade + would cause a queued unhold frame to get lost. Now we just + indicate unhold directly after the masquerade is complete. (issue + ABE-2011) + +2009-11-27 08:47 +0000 [r231401] TransNexus OSP Development + + * apps/app_osplookup.c: 1. Modified exported variable names. 2. + Added destination port support. 3. Added new protocols. 4. Added + QoS. + +2009-11-26 02:09 +0000 [r231299-231369] Tilghman Lesher + + * doc/CODING-GUIDELINES, main/asterisk.c: Reorder option flags. + Change guidelines so that example code is consistent with + guidelines + + * main/channel.c, /: Merged revisions 231298 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009) + | 2 lines After a frame duplication failure, unlock the channel + before returning. ........ + +2009-11-25 15:42 +0000 [r231189] Matthew Nicholson + + * pbx/pbx_lua.c: Load pbx_lua with global symbols to allow linking + with other lua libraries. Found by Maxim Litnitskiy. + +2009-11-24 20:31 +0000 [r231134] Tilghman Lesher + + * apps/app_queue.c: Found a few places where queue refcounts were + counted incorrectly. Also add debug statements. (closes issue + #15982, closes issue #15984) Reported by: atis Patches: + 20091111__issue15982.diff.txt uploaded by tilghman (license 14) + Tested by: atis + +2009-11-24 18:50 +0000 [r231058-231095] Jeff Peeler + + * main/features.c: Fix erroneous hangup extension execution + ast_spawn_extension behaves differently from 1.4 in that hangups + and extensions that do not exist do not return an error, whereas + in 1.6 it does. This is now taken into account so that the + AST_FLAG_BRIDGE_HANGUP_RUN flag gets set properly. (closes issue + #16106) Reported by: ajohnson Tested by: ajohnson + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Fix problem on digital channels due to digital flag not getting + set Changed areas in sig_pri to set the digital flag using a + callback that will also set the corresponding flag in chan_dahdi. + Modified dahdi_request slightly so that if a bearer is marked as + digital, that information is available when creating the new + channel. (closes issue #16151) Reported by: alecdavis Patch based + on bug_16151.diff.txt uploaded by alecdavis (license 585) + +2009-11-24 13:52 +0000 [r231025] Matthew Nicholson + + * CHANGES: Updated CHANGES file to describe the new 'd' option to + app_followme added in r230964 (related to issue #14155) Reported + by: junky + +2009-11-24 04:58 +0000 [r230994] Tilghman Lesher + + * include/asterisk/app.h, funcs/func_strings.c, CHANGES: Add + REPLACE & PASSTHRU functions, overhaul of func_strings, fix API + docs for the ast_get_encoded_* functions. * Add REPLACE function, + which searches a given variable for a set of characters and + replaces each with a given character. * Add PASSTHRU function, + which passes a literal string back, like a NoOp for functions. + Intent is to be able to specify a literal string to another + function that takes a variable name as an argument. * Let the + array manipulation functions work with dialplan functions, in + addition to variables. This allows the array manipulation + functions to modify ASTDB and ODBC backends, assuming the + func_odbc configuration has both read and write functions. + (closes issue #15223) Reported by: ajohnson Patches: + 20091112__issue15223.diff.txt uploaded by tilghman (license 14) + Tested by: lmadsen, tilghman + +2009-11-23 22:37 +0000 [r230964] Matthew Nicholson + + * apps/app_followme.c: Add an option to app_followme to disable the + "please hold" announcement. (closes issue #14155) Reported by: + junky Patches: M14555-trunk.diff uploaded by junky (license 177) + (modified) Tested by: junky + +2009-11-23 15:45 +0000 [r230881] Joshua Colp + + * channels/chan_sip.c, configs/sip.conf.sample: Change fax + detection in chan_sip so it behaves as one would expect. + Internally the way T.38 is negotiated has changed and the option + no longer reflects a behavior that is valid. It will now look for + a CNG tone on received calls and if present send the call to the + 'fax' extension. It is then up to the application or channel to + request the switch over to T.38. + +2009-11-23 15:34 +0000 [r230773-230877] Kevin P. Fleming + + * /, channels/chan_sip.c: Merged revisions 230839 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov + 2009) | 1 line Correct fix for issue #16268... the reporter's + original patch was very close to correct. ........ + + * /, channels/chan_sip.c: Merged revisions 230772 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov + 2009) | 5 lines Ensure that SDP parsing does not ignore the last + line of the SDP. (closes issue #16268) Reported by: sgimeno + ........ + +2009-11-20 22:35 +0000 [r230726] David Vossel + + * channels/chan_iax2.c: fixes iax2 show cache locking error, thanks + alecdavis! (closes issue #16094) Reported by: alecdavis Patches: + bug16094.diff.txt uploaded by alecdavis (license 585) Tested by: + alecdavis, dvossel + +2009-11-20 21:47 +0000 [r230697] Tilghman Lesher + + * include/asterisk/unaligned.h: Revert code in error and include + the gcc suggested workaround for the original problem, while gcc + investigates. + +2009-11-20 21:01 +0000 [r230628] Matthew Nicholson + + * /, main/features.c: Merged revisions 230627 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov + 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR + if it exists. This is necessary for the recordagentcalls option + in chan_agent to store the recorded file name in the bridge CDR. + (closes issue #14590) Reported by: msetim Patches: + queue_agent_userfield.patch uploaded by Laureano (license 265) + Tested by: Laureano, mnicholson ........ + +2009-11-20 17:28 +0000 [r230584] David Ruggles + + * doc/externalivr.txt, apps/app_externalivr.c: Fix/Implement error + events for non-existing files also include a better cmd define + for S command Review: https://reviewboard.asterisk.org/r/430/ + +2009-11-20 17:26 +0000 [r230509-230583] David Vossel + + * include/asterisk/audiohook.h, main/audiohook.c: audiohook signal + trigger on every status change (issue #14618) Review: + https://reviewboard.asterisk.org/r/434/ + + * /, apps/app_mixmonitor.c: Merged revisions 230508 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 + Nov 2009) | 10 lines fixes MixMonitor thread not exiting when + StopMixMonitor is used (closes issue #16152) Reported by: AlexMS + Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license + 671) Tested by: dvossel, AlexMS Review: + https://reviewboard.asterisk.org/r/424/ ........ + +2009-11-19 14:53 +0000 [r230438] David Ruggles + + * apps/app_externalivr.c: Basic cleanup of ExternalIVR: cleaned up + argument parsing; implemented good coding practices where + applicable; replaced most notice level logging with verbose + logging; replaced warning messages that terminated with error + messages; fixed memory leak identified by russellb + +2009-11-16 16:40 +0000 [r230343-230381] Kevin P. Fleming + + * apps/app_fax.c: Fix another buglet in T.38 session teardown at + the end of FAX sessions. + + * apps/app_fax.c: Ensure that only one end of a T.38 session + initiates teardown at completion. + +2009-11-16 01:49 +0000 [r230314] TransNexus OSP Development + + * apps/app_osplookup.c: 1. Added SIP Diversion support. 2. Fixed + compile warning for UUID. + +2009-11-15 17:23 +0000 [r230247] Kevin P. Fleming + + * /, channels/chan_iax2.c: Merged revisions 230246 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 + Nov 2009) | 6 lines Correct mistaken option name in error + message. The configuration option for allowing hosts to make + non-token-based calls is 'calltokenoptional', not + 'calltokenignore'. (reported on asterisk-users) ........ + +2009-11-15 07:53 +0000 [r230217] Tilghman Lesher + + * include/asterisk/channel.h: Increase maximum length of language + buffers (closes issue #16217) Reported by: dsessions + +2009-11-13 22:00 +0000 [r230145] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 230144 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8 + lines Respect the maddr parameter in the Via header. (closes + issue #14446) Reported by: frawd Patches: via_maddr.patch + uploaded by frawd (license 610) Tested by: frawd ........ + +2009-11-13 20:42 +0000 [r230111] Tilghman Lesher + + * apps/app_dial.c, channels/chan_sip.c, apps/app_meetme.c, + apps/app_fax.c, configs/manager.conf.sample, + res/res_musiconhold.c, include/asterisk/manager.h, + channels/chan_iax2.c, apps/app_queue.c, CHANGES, + res/res_monitor.c, main/cdr.c, main/channel.c, main/manager.c, + main/features.c, apps/app_minivm.c, apps/app_chanspy.c, + apps/app_voicemail.c: Display a list of channel variables in each + channel-oriented event. (Closes AST-33) Reviewboard: + https://reviewboard.asterisk.org/r/368/ + +2009-11-13 19:44 +0000 [r229912-230039] Joshua Colp + + * channels/chan_local.c, /: Merged revisions 230038 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov + 2009) | 9 lines Fix a crash caused by two threads thinking they + should both free the chan_local private structure when only one + should. (closes issue #15314) Reported by: sroberts Patches: + Issue15314_Move_Nulling_owner.patch uploaded by davidw (license + 780) Tested by: davidw, lottc ........ + + * UPGRADE.txt, apps/app_chanisavail.c, CHANGES: Store the cause + code that is returned when trying to create a channel in + ChanIsAvail in the AVAILCAUSECODE dialplan variable instead of + overwriting the device state in AVAILSTATUS. (closes issue + #14426) Reported by: macli + + * /: Merged revisions 229965 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6 + lines Document a limitation in the AVAILSTATUS variable from + ChanIsAvail and provide a workaround for it that does not change + existing behavior. (closes issue #14426) Reported by: macli + ........ + + * channels/chan_sip.c: Fix T.38 negotiation regression introduced + with the SDP parser changes. + +2009-11-13 10:53 +0000 [r229819-229871] Olle Johansson + + * main/loader.c: Fixing trunk in a way so that it compiles again. + Thanks, Philippe :-) + + * addons/cdr_mysql.c: If CDR logging is disabled, it's considered a + FAILURE + + * configs/modules.conf.sample, CHANGES, main/asterisk.c, + main/loader.c: Add the capability to require a module to be + loaded, or else Asterisk exits. Review: + https://reviewboard.asterisk.org/r/426/ + +2009-11-13 03:16 +0000 [r229788] TransNexus OSP Development + + * apps/app_osplookup.c: Added full number portability parameter + support. + +2009-11-12 23:43 +0000 [r229750-229754] Jason Parker + + * configs/alsa.conf.sample: Update sample config for ALSA mute and + noaudiocapture + + * channels/chan_alsa.c: Add mute functionality. Add config option + to not try to open capture device. Adds "console {mute|unmute}" + CLI command. Adds mute and noaudiocapture config options (will + update sample configs shortly). (closes issue #14673) Reported + by: Nick_Lewis Patches: chan_alsa.c-oneway3.patch uploaded by + Nick Lewis (license 657) Tested by: qwell + + * channels/chan_oss.c: Fix mute toggling on OSS channels. + +2009-11-12 16:44 +0000 [r229670] David Vossel + + * funcs/func_audiohookinherit.c, /: Merged revisions 229669 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009) + | 6 lines fixes merging error, datastore was being freed in the + wrong function. (closes issue #16219) Reported by: aragon + ........ + +2009-11-12 13:54 +0000 [r229639] Leif Madsen + + * configs/sip.conf.sample: Update sip.conf.sample. Just updating a + spelling error and some capitalization in a documentation update + that Olle added. May the Swenglish be with you. + +2009-11-12 10:24 +0000 [r229606-229607] Olle Johansson + + * configs/sip.conf.sample: Clarification + + * configs/sip.conf.sample: Clarify some security issues early in + the sample configuration + +2009-11-11 20:47 +0000 [r229568] David Ruggles + + * doc/externalivr.txt: Remove non-functional feature from + ExternalIVR documentation Remove non-functional socket + implementation of ExternalIVR from documentation (closes issue + #16225) Reported by: thedavidfactor Patches: + externalivr.txt.20091111.1542.patch uploaded by thedavidfactor + (license 903) + +2009-11-11 19:48 +0000 [r229460-229499] David Brooks + + * main/pbx.c, /: Merged revisions 229498 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009) + | 8 lines Solaris doesn't like NULL going to ast_log Solaris will + crash if NULL is passed to ast_log. This simple patch simply uses + S_OR to get around this. (closes issue #15392) Reported by: + yrashk ........ + + * apps/app_softhangup.c: Flags not initialized in app_softhangup.c, + causing undefined behavior Trivial patch [kobaz] to initialize an + ast_flags = {0} (closes issue #16129) Reported by: kobaz + +2009-11-11 14:30 +0000 [r229431] Leif Madsen + + * CHANGES: Update CHANGES file. Updating the CHANGES file after + noticing an email on the asterisk-dev mailing list from Russell. + (issue #15874) + +2009-11-10 22:14 +0000 [r229361] Tilghman Lesher + + * main/pbx.c, /: Merged revisions 229360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009) + | 12 lines If two pattern classes start with the same digit and + have the same number of characters, they will compare equal. The + example given in the issue report is that of [234] and [246], + which have these characteristics, yet they are clearly not + equivalent. The code still uses these two characteristics, yet + when the two scores compare equal, an additional check will be + done to compare all characters within the class to verify + equality. (closes issue #15421) Reported by: jsmith Patches: + 20091109__issue15421__2.diff.txt uploaded by tilghman (license + 14) Tested by: jsmith, thedavidfactor ........ + +2009-11-10 22:01 +0000 [r229356] David Ruggles + + * doc/externalivr.txt: Merged revisions 229355 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov + 2009) | 9 lines Fix ExternalIVR Documentation Remove + documentation for event that doesn't function (closes issue + #16220) Reported by: thedavidfactor Patches: + externalivr.txt.20091110.1622.patch uploaded by thedavidfactor + (license 903) ........ + +2009-11-10 21:22 +0000 [r229351] Tilghman Lesher + + * apps/app_stack.c: When GOSUB is invoked within an AGI, it may not + exit correctly. (closes issue #16216) Reported by: atis Patches: + 20091110__atis_work.diff.txt uploaded by tilghman (license 14) + Tested by: atis + +2009-11-10 20:06 +0000 [r229282] Joshua Colp + + * /, codecs/codec_g726.c: Merged revisions 229281 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8 + lines Remove broken support for direct transcoding between G.726 + RFC3551 and G.726 AAL2. On some systems the translation core + would actually consider g726aal2 -> g726 -> signed linear to be a + quicker path then g726aal2 -> signed linear which exposed this + problem. (closes issue #15504) Reported by: globalnetinc ........ + +2009-11-10 17:33 +0000 [r229228] David Ruggles + + * /, doc/externalivr.txt: Merged revisions 229191 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov + 2009) | 11 lines Document ExternalIVR event tag collision + ExternalIVR uses the D tag for two different event types. This + documents that behavior and how to differentiate between the two + cases. Also includes a minor spelling fix and clarification + (closes issue #16211) Reported by: thedavidfactor Patches: + externalivr.txt.20091109.1507.patch uploaded by thedavidfactor + (license 903) ........ + +2009-11-10 17:16 +0000 [r229168] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 229167 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 + Nov 2009) | 9 lines don't crash on log message in solaris + AST-2009-006 (closes issue #16206) Reported by: bklang Tested by: + bklang ........ + +2009-11-10 15:53 +0000 [r229102] Matthew Nicholson + + * channels/chan_sip.c: Reverted revision 201717. (closes issue + 0016175) Reported by: paul-tg + +2009-11-10 15:27 +0000 [r229093] David Vossel + + * res/res_config_pgsql.c: fixes pgsql double free of threadstorage + A thread storage variable was being freed incorrectly, which + resulted in a double free if two queries were made in the same + thread. (closes issue #16011) Reported by: cristiandimache + Patches: issue16011.diff uploaded by dvossel (license 671) + +2009-11-10 11:16 +0000 [r229050] Gavin Henry + + * contrib/scripts/asterisk.ldap-schema: Schema file additions * + Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox + objectClasses to allow standalone dialplan, account and mailbox + entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage, + AstAccountTransport, AstAccountPromiscRedir, - + AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap, + - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed + redundant IPaddr (there's already IPAddress) - Gives more + configuration Flags for SIP-Users available (tested) - Allows to + create Asterisk Attributes in defined Asterisk ObjectClasses + without extensibleObject (which really should be the last + resort); gives also additional possibilities for LDAP-filter + (closes issue #15874) Reported by: Medozas Patches: + asterisk.ldap-schema.patch uploaded by Medozas (license 41) + Tested by: Medozas, suretec + +2009-11-09 22:50 +0000 [r229015] Terry Wilson + + * channels/chan_local.c: Don't crash when bridge->tech_pvt == NULL + This is a similar solution to what is in place for chan_agent + (closes issue #16003) Reported by: atis Tested by: twilson + +2009-11-09 17:17 +0000 [r228979] Tilghman Lesher + + * channels/iax2-parser.c: Don't try to convert a 64-bit integer, + where only a 32-bit integer is stored. (closes issue #16194) + Reported by: habile + +2009-11-09 16:28 +0000 [r228947] Matthew Nicholson + + * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add the + 'relative-periodic-announce' option to app_queue to allow for + calculating the time of announcments from the end of the previous + announcment rather than from the beginning. (closes issue #15260) + Reported by: tonils + +2009-11-09 15:38 +0000 [r228897] Leif Madsen + + * main/channel.c, /: Merged revisions 228896 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009) + | 6 lines Update WARNING message. Update a WARNING message to + give a suggested fix when encountered. (closes issue #16198) + Reported by: atis Tested by: atis ........ + +2009-11-09 14:37 +0000 [r228858] Matthew Nicholson + + * /, include/asterisk/lock.h: Merged revisions 228827 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, + 09 Nov 2009) | 8 lines Perform limited bounds checking when + destroying ast_mutex_t structures to make sure we don't try to + use negative indices. (closes issue #15588) Reported by: zerohalo + Patches: 20090820__issue15588.diff.txt uploaded by tilghman + (license 14) Tested by: zerohalo ........ + +2009-11-09 07:37 +0000 [r228798] Tilghman Lesher + + * addons/cdr_mysql.c, main/event.c, channels/chan_console.c, + res/res_pktccops.c, main/loader.c: Fix various problems detected + with Valgrind. * chan_console accessed pvts after deallocation. * + cdr_mysql stored a pointer that was freed by realloc() * The + module loader did not check usecount on shutdown, which led to + chan_iax2 reading a timer that was already unloaded. * The event + subsystem sometimes creates an event with no IEs. Due to a corner + condition, the code would read beyond the memory boundary. * + res_pktccops did not correctly check whether its monitor thread + was started. (closes issue #16062) Reported by: alexanderheinz + Patches: 20091109__issue16062.diff.txt uploaded by tilghman + (license 14) Tested by: tilghman + +2009-11-07 17:02 +0000 [r228766] Tzafrir Cohen + + * contrib/init.d/rc.debian.asterisk: Add LSB headers to the Debian + init.d script See also issue #14864 . + +2009-11-06 22:35 +0000 [r228693] David Vossel + + * main/channel.c, /: Merged revisions 228692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009) + | 9 lines fixes audiohook write crash occuring in chan_spy + whisper mode. After writing to the audiohook list in ast_write(), + frames were being freed incorrectly. Under certain conditions + this resulted in a double free crash. (closes issue #16133) + Reported by: wetwired (closes issue #16045) Reported by: + bluecrow76 Patches: issue16045.diff uploaded by dvossel (license + 671) Tested by: bluecrow76, dvossel, habile ........ + +2009-11-06 22:32 +0000 [r228691] Richard Mudgett + + * channels/chan_dahdi.c, CHANGES, channels/sig_pri.c: Created + standard location to add options to chan_dahdi for ISDN dialing. + Dial(DAHDI/g1[/extension[/options]]) Current options: + K() R Reverse charging indication (Collect calls) + The earlier Dial(DAHDI/g1[/K][/extension] format + was variable and did not allow for the easy addition of more + options. The earlier 'C' prefix character for reverse charge + indiation would conflict with the a-d DTMF digits if ISDN uses + them. + +2009-11-06 22:07 +0000 [r228661] David Brooks + + * tests/test_amihooks.c: ami_testhooks.c automatically registers + hook ami_testhooks.c was registering for AMI events upon module + load. Moved the registration to its own CLI command. Added CLI + command for unregistering the hook. Changed some of the wording, + removed unnecessary arguments/parameters. Reported by: rmudgett + +2009-11-06 22:02 +0000 [r228658-228659] Mark Michelson + + * addons/chan_ooh323.c: Make compilation of chan_ooh323 disabled by + default. All addons modules should be disabled by default, + requiring the user to turn them on if desired. After all, these + are addons we're talking about here. + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c: Get + chan_ooh323 to compile with gcc 4.2. For some reason, the code + compiles just fine with later versions of GCC, but this one + requires some weird double casting in order to get rid of all + warnings. Whatever. + +2009-11-06 19:53 +0000 [r228621] Richard Mudgett + + * main/frame.c: Fix compiler warning gcc 4.2.4 found + +2009-11-06 19:47 +0000 [r228620] Matthew Nicholson + + * funcs/func_base64.c, /, main/utils.c: Merged revisions 228378 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov + 2009) | 8 lines Properly handle '=' while decoding base64 + messages and null terminate strings returned from BASE64_DECODE. + (closes issue #15271) Reported by: chappell Patches: + base64_fix.patch uploaded by chappell (license 8) Tested by: + kobaz ........ + +2009-11-06 19:38 +0000 [r228616] Tilghman Lesher + + * channels/chan_nbs.c, addons/chan_mobile.c: Missed these two + channel drivers on the codec_bits merge + +2009-11-06 18:37 +0000 [r228499-228548] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 228547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4 + lines Don't overwrite caller ID name on a trunk with the + configured fullname when using users.conf (issue ABE-1989) + ........ + + * doc/tex/localchannel.tex: Fix the localchannel.tex file. + +2009-11-06 17:22 +0000 [r228420-228441] David Vossel + + * codecs/codec_ilbc.c: Fixes merging issue from 1.4, frame data is + held in data.ptr in trunk + + * /, codecs/codec_ilbc.c: Merged revisions 228418 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009) + | 13 lines fixes segfault in iLBC For reasons not yet known, it + appears possible for an ast_frame to have a datalen greater than + zero while the actual data is NULL during Packet Loss + Concealment. Most codecs don't support PLC so this doesn't affect + them. This patch catches the malformed frame and prevents the + crash from occuring. Additional efforts to determine why it is + possible for a frame to look like this are still being + investigated. (issue #16979) ........ + +2009-11-06 16:42 +0000 [r228410] Joshua Colp + + * /, main/abstract_jb.c: Merged revisions 228409 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7 + lines Fix a bug caused by a partially invalid frame (from the + jitterbuffer) passing through the Asterisk core. (closes issue + #15560) Reported by: jvandal (closes issue #15709) Reported by: + covici ........ + +2009-11-06 15:42 +0000 [r228268-228339] David Vossel + + * /, main/astfd.c: Merged revisions 228338 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009) + | 5 lines fixes crash in astfd.c (closes issue #15981) Reported + by: slavon ........ + + * funcs/func_audiohookinherit.c: fixes memory leak in + func_audiohookinherit.c (closes issue #15394) Reported by: boroda + Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks + (license 790) Tested by: dbrooks, boroda + +2009-11-05 22:59 +0000 [r228233] Mark Michelson + + * funcs/func_cdr.c: Fix XML in func_cdr.c + +2009-11-05 22:12 +0000 [r228191-228196] Tilghman Lesher + + * apps/app_meetme.c: Yet another error message in the dialplan + (thanks, rmudgett/russellb) + + * apps/app_meetme.c: MEETME_INFO should not return a literal error + message to the dialplan. (closes issue #15450) Reported by: + JimVanM Patches: meetmeinfopatch.diff.txt uploaded by dbrooks + (license 790) Tested by: JimVanM + +2009-11-05 21:23 +0000 [r228189] Jeff Peeler + + * apps/app_chanspy.c: Fix the fix for chanspy option o In 224178, I + assumed the uploaded patch was correct as it had received + positive feedback. The flags were being checked in the incorrect + location. Upon testing the fix this time it was also found that + the flags from the dialplan weren't being copied to the + chanspy_translation_helper. (closes issue #16167) Reported by: + marhbere + +2009-11-05 19:34 +0000 [r228145] David Brooks + + * channels/chan_misdn.c, /: Merged revisions 228078 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 + Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash + related to chan_misdn connection. Patch submitted by + gknispel_proformatique, tested by francesco_r. "I have many crash + since i have upgraded to Asterisk 1.4.27-rc2. Attached a full + bt." This patch zeros out an ast_frame. (closes issue #16041) + Reported by: francesco_r ........ + +2009-11-05 19:16 +0000 [r228080] Jason Parker + + * channels/chan_vpb.cc, /: Merged revisions 228079 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov + 2009) | 8 lines Fix crash on VPB exception when no hardware is + present. (closes issue #14970) Reported by: tzafrir Patches: + vpb_exception.diff uploaded by tzafrir (license 46) Tested by: + markwaters ........ + +2009-11-05 17:26 +0000 [r228015-228049] Tilghman Lesher + + * main/frame.c: Rework codecs command to comply with the 64-bit + scheme + + * apps/app_externalivr.c: Don't crash if no arguments are passed. + (closes issue #16119) Reported by: thedavidfactor + +2009-11-04 23:50 +0000 [r227914-227945] Jeff Peeler + + * /, res/res_monitor.c: Merged revisions 227944 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) + | 14 lines Fix incorrect filename comparsion after monitor file + change The logic to detect if a requested file is indeed a + different file from the current file was incorrect. The main + issue being confusion of the use of filename_base which was + previously set without pathing information and then compared to + another full path. Robust file comparison logic has been added to + properly check if two files are the same even if symlinks are + used. (closes issue #15313) Reported by: caspy Patches: + 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license + 325) but mostly tilghman's work ........ + + * addons/chan_ooh323.c: Update chan_ooh323 to support the expanded + codec bitfield from 227580. + +2009-11-04 22:10 +0000 [r227898] Alexandr Anikin + + * addons/ooh323c/src/oochannels.h, + addons/ooh323c/src/ooCmdChannel.h, addons/chan_ooh323.c, + addons/ooh323c/src/printHandler.h, addons/ooh323c/src/ooq931.h, + addons/ooh323c/src/ootrace.h, addons/chan_ooh323.h, + addons/ooh323c/src/ooasn1.h, addons/ooh323c/src/ootypes.h, + addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c, + addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooTimer.c, + addons/ooh323c/src/ooLogChan.h, + addons/ooh323c/src/ooCapability.c, + addons/ooh323c/src/ooStackCmds.h, addons/ooh323c/src/dlist.c, + addons/ooh323c/src/eventHandler.c, + addons/ooh323c/src/ooCapability.h, + addons/ooh323c/src/eventHandler.h, addons/Makefile, + addons/ooh323cDriver.c, addons/ooh323c/src/ooDateTime.c, + addons/ooh323c/src/rtctype.c, addons/ooh323cDriver.h, + addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/encode.c, + addons/ooh323c/src/ooUtils.c, addons/ooh323c/src/ooGkClient.c, + addons/ooh323c/src/ooDateTime.h, addons/ooh323c/src/ooCalls.h, + addons/ooh323c/src/ooh323ep.c, addons/ooh323c/src/ooGkClient.h, + addons/ooh323c/src/ooports.c, addons/ooh323c/src/ooh323ep.h, + addons/ooh323c/src/memheap.c, addons/ooh323c/src/ooh323.c, + addons/ooh323c/src/h323/H323-MESSAGESDec.c, + addons/ooh323c/src/ooh245.c, addons/ooh323c/src/memheap.h, + addons/ooh323c/src/ooh323.h, addons/ooh323c/src/decode.c, + addons/ooh323c/src/context.c, addons/ooh323c/src/perutil.c, + addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLDec.c, + addons/ooh323c/src/ooh245.h, addons/ooh323c/src/ooSocket.c, + addons/ooh323c/src/h323/H235-SECURITY-MESSAGESDec.c, + addons/ooh323c/src/oochannels.c, + addons/ooh323c/src/ooCmdChannel.c, + addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooSocket.h, + addons/ooh323c/src/ooCommon.h, addons/ooh323c/src/ooq931.c, + addons/ooh323c/src/ootrace.c: Reworked chan_ooh323 channel + module. Many architectural and functional changes. Main changes + are threading model chanes (many thread in ooh323 stack instead + of one), modifications and improvements in signalling part, + additional codecs support (726, speex), t38 mode support. This + module tested and used in production environment. (closes issue + #15285) Reported by: may213 Tested by: sles, c0w, OrNix Review: + https://reviewboard.asterisk.org/r/324/ + +2009-11-04 21:39 +0000 [r227829-227897] Matthew Nicholson + + * apps/app_dial.c, CHANGES: Added the 'a' option to app dial and + modified app_dial to set the answertime when the called channel + answers. This change causes answertime to be correct even if the + called channel hangs up during an announcement triggered by the + A() option. (closes issue #15936) Reported by: falves11 Patches: + dial-macro-billsec-fix1.diff uploaded by mnicholson (license 96) + dial-caller-answer1.diff uploaded by mnicholson (license 96) + Tested by: falves11, mnicholson + + * apps/app_dial.c, /: Merged revisions 227827 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov + 2009) | 10 lines This patch modifies the Dial application to + monitor the calling channel for hangups while playing back + announcements. (closes issue #16005) Reported by: falves11 + Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson + (license 96) Tested by: mnicholson, falves11 Review: + https://reviewboard.asterisk.org/r/407/ ........ + +2009-11-04 20:35 +0000 [r227824] Tilghman Lesher + + * include/asterisk/unaligned.h: Fixes for gcc 4.4 + +2009-11-04 20:13 +0000 [r227759] Matthew Nicholson + + * channels/chan_sip.c: Modify the SDP parsing code to parse session + and media level items separately. With the new code, media level + proprieties should no longer be confused with session level + proprieties. This change also reorganizes some of the SDP parsing + code which should make it easier to manage in the future. (closes + issue #14994) Reported by: frawd Tested by: frawd, mnicholson, + file Review: https://reviewboard.asterisk.org/r/414/ + +2009-11-04 19:26 +0000 [r227712-227739] Joshua Colp + + * /, static-http/prototype.js: Merged revisions 227735 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov + 2009) | 5 lines Fix a security issue where it may be possible for + someone to execute a cross-site AJAX request exploit. + (AST-2009-009) ........ + + * /, channels/chan_sip.c: Merged revisions 227700 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5 + lines Fix a security issue where sending a REGISTER with a + differing username in the From URI and Authorization header would + reveal whether it was valid or not. (AST-2009-008) ........ + +2009-11-04 16:41 +0000 [r227646] Mark Michelson + + * main/frame.c: Add a couple more casts so that code compiles + correctly. + +2009-11-04 16:35 +0000 [r227645] Tilghman Lesher + + * include/asterisk/pbx.h: mmichelson reported a compilation error + related to codec bit expansion that should be resolved with a + simple include of frame_defs.h + +2009-11-04 16:25 +0000 [r227643] Jeff Peeler + + * channels/chan_dahdi.c: fix trunk building + +2009-11-04 16:17 +0000 [r227579-227615] Tilghman Lesher + + * channels/chan_sip.c, channels/chan_iax2.c: Two other trunk build + fixes (reported by seanbright on #asterisk-dev) + + * addons/format_mp3.c: Fix trunk building + + * main/udptl.c, main/autoservice.c, apps/app_dahdibarge.c, + main/frame.c, channels/chan_local.c, main/rtp_engine.c, + include/asterisk/autoconfig.h.in, apps/app_record.c, + apps/app_test.c, bridges/bridge_softmix.c, + apps/app_alarmreceiver.c, codecs/ex_alaw.h, codecs/ex_adpcm.h, + formats/format_wav_gsm.c, formats/format_sln16.c, + codecs/ex_gsm.h, channels/chan_iax2.c, main/indications.c, + res/res_rtp_multicast.c, channels/chan_dahdi.c, + include/asterisk/bridging_technology.h, pbx/pbx_spool.c, + channels/sig_analog.c, include/asterisk/audiohook.h, + channels/chan_skinny.c, configure, main/strcompat.c, + include/asterisk/compat.h, formats/format_pcm.c, main/features.c, + channels/chan_alsa.c, apps/app_amd.c, formats/format_h263.c, + apps/app_url.c, apps/app_externalivr.c, formats/format_jpeg.c, + main/bridging.c, codecs/ex_ulaw.h, apps/app_milliwatt.c, + formats/format_gsm.c, apps/app_dial.c, main/pbx.c, + formats/format_wav.c, channels/chan_bridge.c, apps/app_echo.c, + apps/app_fax.c, include/asterisk/slin.h, channels/chan_agent.c, + configure.ac, formats/format_ogg_vorbis.c, apps/app_disa.c, + include/asterisk/unaligned.h, codecs/ex_speex.h, + include/asterisk/channel.h, apps/app_talkdetect.c, + channels/iax2-parser.c, apps/app_speech_utils.c, + channels/iax2-parser.h, channels/chan_misdn.c, + apps/app_waitforring.c, channels/iax2.h, codecs/codec_dahdi.c, + main/audiohook.c, apps/app_chanspy.c, formats/format_g726.c, + include/asterisk/frame_defs.h (added), + include/asterisk/translate.h, include/asterisk/slinfactory.h, + channels/chan_unistim.c, channels/chan_vpb.cc, + channels/chan_multicast_rtp.c, formats/format_sln.c, + apps/app_meetme.c, apps/app_dictate.c, codecs/ex_g722.h, + codecs/ex_g726.h, channels/chan_gtalk.c, res/res_musiconhold.c, + apps/app_followme.c, formats/format_siren7.c, + include/asterisk/abstract_jb.h, main/asterisk.exports, + main/channel.c, formats/format_ilbc.c, channels/chan_phone.c, + main/dial.c, main/manager.c, funcs/func_volume.c, res/res_agi.c, + apps/app_mp3.c, main/app.c, doc/codec-64bit.txt (added), + formats/format_h264.c, include/asterisk/rtp_engine.h, + include/asterisk/frame.h, formats/format_siren14.c, + codecs/ex_ilbc.h, channels/chan_mgcp.c, apps/app_jack.c, + res/res_rtp_asterisk.c, apps/app_nbscat.c, channels/chan_sip.c, + codecs/ex_lpc10.h, apps/app_festival.c, main/slinfactory.c, + main/translate.c, res/res_adsi.c, channels/chan_console.c, + channels/h323/chan_h323.h, channels/sig_pri.c, apps/app_queue.c, + channels/chan_oss.c, channels/chan_jingle.c, + formats/format_vox.c, include/asterisk/bridging.h, + main/abstract_jb.c, main/file.c, channels/chan_h323.c, + formats/format_g723.c, codecs/codec_ulaw.c, apps/app_sms.c, + include/asterisk/pbx.h, main/dsp.c, formats/format_g729.c: Expand + codec bitfield from 32 bits to 64 bits. Reviewboard: + https://reviewboard.asterisk.org/r/416/ + + * configure, include/asterisk/autoconfig.h.in, configure.ac: + chan_misdn will fail to compile if the redirect_dn member is + missing + +2009-11-04 08:22 +0000 [r227545] Olle Johansson + + * main/manager.c: Add destruction of iterators to avoid problems + with refcounters (per Russell's review of another patch) + +2009-11-04 03:15 +0000 [r227509] Tilghman Lesher + + * apps/app_queue.c: Don't crash when state_interface is NULL. + +2009-11-03 22:13 +0000 [r227462-227464] Russell Bryant + + * res/res_pktccops.c: Resolve another warning. + + * main/manager.c, pbx/pbx_config.c: Resolve a warning from gcc + 4.4.1. + + * channels/chan_mgcp.c: Resolve some dev-mode warnings. + +2009-11-03 21:26 +0000 [r227448] David Brooks + + * main/manager.c, include/asterisk/manager.h, tests/test_amihooks.c + (added): AMI hook interface This patch, originally submitted by + jozza, enables custom modules to send actions to AMI and receive + messages from AMI via a hook interface. Included is a simple test + module to illustrate the interface. (closes issue #14635) + Reported by: jozza Review: + https://reviewboard.asterisk.org/r/412/ + +2009-11-03 21:21 +0000 [r227435] Matthew Nicholson + + * main/cdr.c, apps/app_forkcdr.c, configs/cdr_custom.conf.sample, + funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h, + CHANGES: This patch adds a sequence field to CDRs that can be + combined with the linkedid or uniqueid field to uniquely identify + a CDR. (closes issue #15180) Reported by: Nick_Lewis Patches: + cdr-sequence10.diff uploaded by mnicholson (license 96) Tested + by: mnicholson + +2009-11-03 21:16 +0000 [r227424] Joshua Colp + + * configs/queues.conf.sample, apps/app_queue.c: Add support for + using a hint when configuring a state interface using the format + hint:@. (closes issue #15168) Reported by: + p_lindheimer Patches: queue_extenstate5_1.4.svn.patch uploaded by + GameGamer43 (license 894) + +2009-11-03 19:59 +0000 [r227372] Jason Parker + + * Makefile, main/Makefile: Fix some build issues on Solaris. + (closes issue #14517) (SWP-109) Reported by: asgaroth Patches: + bug_14517.diff uploaded by snuffy (license 35) Tested by: + asgaroth, snuffy, dougm, qwell + +2009-11-03 19:48 +0000 [r227361-227368] Leif Madsen + + * apps/app_controlplayback.c: Change warning message to debug + message. app_controlplayback outputs a warning, when in fact it + is normal. (closes issue #16071) Reported by: atis Patches: + controlplayback_warning.patch uploaded by atis (license 242) + + * configs/extensions.conf.sample: Additional fixes to the + extensions.conf.sample file. Update the extensions.conf.sample + [stdexten] context so that we use the variable instead of + requiring it to be passed explicitly. Also updated uses of the + [stdexten] context throughout. (closes issue #15858) Reported by: + pprindeville Patches: stdexten-context-update.txt uploaded by + lmadsen (license 10) Tested by: pprindeville + +2009-11-03 18:22 +0000 [r227298] Matthew Nicholson + + * channels/chan_sip.c: Fixed a spelling error in the q850 reason + header option in the output of sip show settings. + +2009-11-03 17:58 +0000 [r227277] Richard Mudgett + + * /: Recorded merge of revisions 227275 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) + | 4 lines Make sure the outgoing flag is cleared if a new channel + fails to get created for outgoing calls. This is the relevant + portion of asterisk/trunk -r226648 ........ + +2009-11-03 17:56 +0000 [r227276] Tilghman Lesher + + * channels/chan_mgcp.c: Code guidelines fixes only + +2009-11-03 17:12 +0000 [r227238] David Vossel + + * channels/chan_sip.c: user.conf entries in SIP were not having + their peer type set. (closes issue #16120) Reported by: jsmith + +2009-11-03 16:56 +0000 [r227237] Olle Johansson + + * funcs/func_speex.c: Adding some clarifications to func_speex + doxygen docs. The functions needed doesn't exist in Speex 1.05 + which is what a lot of distros use. 1.2 seems to have been in + beta status for years, and does include the sexy functions needed + for func_speex to work. + +2009-11-03 15:37 +0000 [r227167] Joshua Colp + + * /: Merged revisions 227166 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 + lines Fix a bug where an RPID header could be generated with a + blank username in the URI. (closes issue #15909) Reported by: + kobaz ........ + +2009-11-03 15:19 +0000 [r227162] Leif Madsen + + * configs/extensions.conf.sample: Update extensions.conf.sample + file to fix incorrect extensions. (closes issue #15857) Reported + by: pprindeville Patches: stdexten.patch#2 uploaded by + pprindeville (license 347) Tested by: pprindeville + +2009-11-03 11:11 +0000 [r227091] Olle Johansson + + * Makefile, /, channels/chan_sip.c: Merged revisions 227088 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 + lines Use proper response code when violating Contact ACL's. + https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a + quick review. (EDVX-003) ........ + +2009-11-02 22:29 +0000 [r227049] Tilghman Lesher + + * configs/mgcp.conf.sample, include/asterisk/pktccops.h (added), + CHANGES, res/res_pktccops.c (added), channels/chan_mgcp.c, + configs/res_pktccops.conf.sample (added): Add PacketCable NCS 1.0 + support for Docsis/Eurodocsis networks (closes issue #12950) + Reported by: alea-soluciones Patches: + ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones + (license 514) Tested by: alea-soluciones, adomjan, urtho, + nahuelgreco + +2009-11-02 20:59 +0000 [r226973-226974] David Brooks + + * channels/chan_sip.c: SIP channel name uniqueness SIP channel + names were supposed to be unique by way of a name suffix derived + from the pointer to the channel's private data. Uniqueness was + preserved on 32-bit systems, but not on 64-bit systems. This + patch, as suggested by kpfleming, replaces this suffix with a + simple incremented unsigned int. (closes issue #15152) Reported + by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ + + * /: SIP channel name uniqueness SIP channel names were supposed to + be unique by way of a name suffix derived from the pointer to the + channel's private data. Uniqueness was preserved on 32-bit + systems, but not on 64-bit systems. This patch, as suggested by + kpfleming, replaces this suffix with a simple incremented + unsigned int. (closes issue #15152) Reported by: palbrecht + Review: https://reviewboard.asterisk.org/r/420/ + +2009-11-02 20:43 +0000 [r226970] Olle Johansson + + * main/http.c: Adding external reference for doxygen + +2009-11-02 18:08 +0000 [r226890] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 226889 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | + 11 lines Fix a bug where the recorded privacy introduction file + would not get removed if the caller hung up while the called + party had not yet answered. This was fixed by introducing an + argument to the 'n' option which, when enabled, removes the + introduction file under all scenarios. This was done to preserve + the behavior that has existed for quite some time. (closes issue + #14674) Reported by: ulogic Patches: bug14674.patch uploaded by + jpeeler (license 325) ........ + +2009-11-02 17:34 +0000 [r226882] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, UPGRADE.txt, + channels/sig_pri.c: DAHDI ISDN channel names will not allow + device state to work. (Interim solution.) Since ISDN works like + SIP and not analog ports in regard to devices, the device state + based on the ISDN channel number could not work. This has not + been an issue until the advent of PTMP NT mode. Previously, ISDN + lines were used as trunks and did not have to keep track of + specific devices. As an interim solution until device states are + properly implemented, the channel name is being changed to the + following format to use the generic device state support: + DAHDI/i/[:]- Dialplan + hints would thus be: exten => xxx,hint,DAHDI/i2/5551212 This will + work with the following restrictions: * The number of + devices/phones cannot exceed the number of B channels. (i.e., BRI + has 2) * Each device/phone can only have one number. No shared + MSN's. * The phones/devices probably should not use + subaddressing. + +2009-11-02 17:15 +0000 [r226812] Tilghman Lesher + + * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226811 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) + | 8 lines Don't allow two separate instances of safe_asterisk + when restarting from the init script. (closes issue #14562) + Reported by: davidw Patches: Initially + 20091022__issue14562.diff.txt uploaded by tilghman (license 14) + Modified to 20091030__Issue14562_diff.txt uploaded by davidw + (license 780) Tested by: davidw ........ + +2009-11-02 14:57 +0000 [r226687] Matthew Nicholson + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: This patch + adds support for a draft proposal for adding Q.850 reason headers + to sip messages. (closes issue #13385) Reported by: adomjan + Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded + by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch + uploaded by adomjan (license 487) + chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by + adomjan (license 487) sip-q850-hangupcause1.diff uploaded by + mnicholson (license 96) Tested by: adomjan + +2009-10-30 23:26 +0000 [r226648] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_pri.c: Cleanup some flags on + DAHDI PRI channel hangup. * Cleanup some flags on DAHDI PRI + channel hangup. (sig_pri split) * Make sure the outgoing flag is + cleared if a new channel fails to get created for outgoing calls. + * Remove some unused flags since sig_pri was split. + +2009-10-30 04:08 +0000 [r226606] Russell Bryant + + * include/asterisk/doxygen/architecture.h (added), + res/res_rtp_asterisk.c, res/res_rtp_multicast.c, + include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen, + main/asterisk.c: Add an "Asterisk Architecture Overview" section + to the doxygen documentation. This is a side project I've been + poking at this week. The intent is to discuss Asterisk + architecture in a top down fashion to help new developers + understand how Asterisk is put together. There is a ton of stuff + to write about, so this will just continue to evolve over time. + +2009-10-29 18:13 +0000 [r226532] Joshua Colp + + * channels/chan_local.c, /, doc/tex/localchannel.tex: Merged + revisions 226531 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 + lines Add an option to enabling passing music on hold start and + stop requests through instead of acting on them in chan_local. + (closes issue #14709) Reported by: dimas ........ + +2009-10-29 12:20 +0000 [r226490] Olle Johansson + + * channels/chan_local.c: Doxygen documentation update + +2009-10-28 20:50 +0000 [r226453] Tzafrir Cohen + + * build_tools/get_documentation: remove empty awk pattern (//) + Solaris 10 nawk doesn't lthe empty pattern ike '//' for 'always'. + Just remove that. No pattern at all always matches. + +2009-10-28 20:11 +0000 [r226378-226384] Leif Madsen + + * /, configs/sip.conf.sample: Merged revisions 226382 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 + Oct 2009) | 9 lines Update documentation in sip.conf.sample. + Update the documentation in sip.conf.sample in order to make it + more clear that directmedia/canreinvite do not cause Asterisk to + ignore reINVITEs. It is only used to stop Asterisk from + generating a reINVITE, but does not stop it from accepting them + if necessary. (closes issue #15644) Reported by: lmadsen ........ + + * doc/tex/channelvariables.tex: Merged revisions 226377 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) + | 7 lines Update CALLINGSUBADDR channel variable documentation. + (closes issue #15734) Reported by: alecdavis Patches: + channelvariables.tex.diff.txt uploaded by alecdavis (license 585) + Tested by: alecdavis ........ + +2009-10-28 18:04 +0000 [r226305] Tilghman Lesher + + * /, include/asterisk/linkedlists.h: Merged revisions 226304 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) + | 2 lines Fix documentation (pointed out by TheDavidFactor on + #-dev) ........ + +2009-10-28 08:47 +0000 [r226227-226270] Tzafrir Cohen + + * contrib/upstart/asterisk.user.conf: Remove extra cleanup in case + we have more than one Asterisk. /var/run would be cleaned on + startup on most systems anyway. + + * contrib/upstart/asterisk.user.conf (added): another variation of + the upstart script + +2009-10-27 21:03 +0000 [r226184] Olle Johansson + + * Makefile: Adding compile time flags for Snow Leopard, Leopard and + some other animals + +2009-10-27 20:22 +0000 [r226159] Tilghman Lesher + + * main/manager.c, /: Merged revisions 226138 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) + | 7 lines Manager output is not always NULL-terminated, so force + a NULL at the end of the filestream. (closes issue #15495) + Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded + by tilghman (license 14) Tested by: pdf ........ + +2009-10-27 16:48 +0000 [r226099] Terry Wilson + + * res/res_http_post.c: Don't prepend the URI prefix to the post + directory + +2009-10-27 13:30 +0000 [r226060] Joshua Colp + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add + support for receiving unsolicited MWI NOTIFY messages. This + change adds a configuration option to SIP peers, + unsolicited_mailbox, which configures a virtual mailbox to use + for received new/old MWI information. This virtual mailbox can + then be used by any device supporting MWI. (closes issue #13028) + Reported by: AsteriskRocks Patches: + bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj + (license 830) + +2009-10-26 22:46 +0000 [r226018] Tzafrir Cohen + + * /, configure, configure.ac: detect ARM Linux EABI OSARCH as + linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even + if host_os is linux-gnueabi * When checking if we are Linux, + check OSARCH rather than host_os The newer ARM ABI ("EABI") shows + the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch + sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is + tested for the value of 'linux-gnu' in one or two places in the + tree. This patch also fixes the check libcap to check for $OSARCH + rather than $host_os . See also: + http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via + svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4 + +2009-10-26 22:04 +0000 [r225955-225956] Kevin P. Fleming + + * main/editline/makelist.in, channels/chan_sip.c, UPGRADE.txt, + UPGRADE-1.6.txt, doc/lang/language-criteria.txt: Fix building in + REF_DEBUG mode. + + * main/astobj2.c: Correct broken logic from revision 225405. The + code committed in revision 225405 was broken; instead of removing + the unreference code, the logic used to decide when to do it + should have been reversed. This patch corrects the situation, and + makes reference counting work properly again. + +2009-10-26 19:40 +0000 [r225912] Jeff Peeler + + * channels/chan_sip.c: ACL check not present for verifying SIP + INVITEs The ACL check in check_peer_ok was missing and has now + been restored. The missing check allowed for calls to be made on + prohibited networks where an ACL was defined in sip.conf and the + allowguest option was set to off. See the AST security advisory + below for more information. Merge code associated with + AST-2009-007. (closes issue #16091) Reported by: thom4fun + +2009-10-26 16:07 +0000 [r225872] Richard Mudgett + + * channels/chan_dahdi.c: Make conditionals create previous code + when libpri/ss7 are present. + +2009-10-26 13:29 +0000 [r225767-225836] Tzafrir Cohen + + * channels/chan_dahdi.c: span numbers in pri debug / error messages + Prefix PRI trace messages with the span number. This makes the + trace readable even when you have a multi-port device. (closes + issue #15054) Reported by: tzafrir Patches: + dahdi_pri_debug_spannum.diff uploaded by tzafrir (license 46) + + * channels/chan_dahdi.c: Re-arange code a bit to build in dev-mode + without ss7 No change of functionality here. Just localized a + variable and indented code into blocks. + + * channels/chan_dahdi.c: Make chan_dahdi build even without PRI / + SS7 (Note: still some strange build warnings without SS7 in + dev-mode) + +2009-10-24 14:40 +0000 [r225727] Kevin P. Fleming + + * channels/chan_sip.c: Improve performance of pedantic mode dialog + searching in chan_sip. This patch changes chan_sip to use the new + astobj2 OBJ_MULTIPLE iterator support to make pedantic mode + dialog searching in find_call() not require a linear search of + all dialogs in the list of dialogs. This patch does *not* change + the dialog matching logic (more on that later), just improves the + searching performance. + +2009-10-23 16:57 +0000 [r225692] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, configure, + include/asterisk/autoconfig.h.in, configure.ac, CHANGES, + channels/sig_pri.c: Add to chan_dahdi ISDN HOLD, Call deflection, + and keypad facility support. * Added handling of received + HOLD/RETRIEVE messages and the optional ability to transfer a + held call on disconnect similar to an analog phone. * Added + CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI + PTMP. Will reroute/deflect an outgoing call when receive the + message. Can use the DAHDISendCallreroutingFacility to send the + message for the supported switches. * Added ability to + send/receive keypad digits in the SETUP message. Send keypad + digits in SETUP message: + Dial(DAHDI/g1[/K][/extension]) Access any received + keypad digits in SETUP message by: ${CHANNEL(keypad_digits)} * + Added support for BRI PTMP NT mode. + +2009-10-23 16:40 +0000 [r225690] Sean Bright + + * Makefile, agi/Makefile, agi/agi.xml (added): Optionally build and + install the sample AGIs in the agi/ directory. + +2009-10-23 14:41 +0000 [r225650] David Vossel + + * channels/chan_sip.c: Fixes an iterator memory leak and + uninitialized memory + +2009-10-23 14:02 +0000 [r225582] Kevin P. Fleming + + * Makefile, /: Merged revisions 225581 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct + 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on + every build. For some reason the menuselect.makeopts file was + listed as PHONY in the Makefile, resulting in 'make' needing to + rebuild it for every build. This then resulted in the embedded + module rules being rebuilt on every build, which can be slow and + is unnecessary. This patch fixes the problem by properly allowing + 'make' to know when the menuselect.makeopts file needs to be + rebuilt (defining the proper dependencies). ........ + +2009-10-22 22:24 +0000 [r225483-225515] Leif Madsen + + * README: Update README documentation. Update the README + documentation to correctly describe which CLI command you should + use when attempting to get help from the CLI. (closes issue + #16064) Reported by: thedavidfactor Patches: readme.patch + uploaded by thedavidfactor (license 903) + + * /, doc/valgrind.txt, contrib/valgrind.supp (added): Merged + revisions 225484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) + | 11 lines Clean valgrind output by suppressing false errors. + Update valgrind.txt documentation and add valgrind.supp file in + order to allow those who are creating valgrind output to have + less false errors in the logfile. (closes issue #16007) Reported + by: atis Patches: valgrind.txt.diff uploaded by atis (license + 242) asterisk2.supp uploaded by atis (license 242) Tested by: + atis, amorsen ........ + + * include/asterisk/doxyref.h, + include/asterisk/doxygen/asterisk-git-howto.h (added): Add + Asterisk Git HowTo documentation. Added documentation on how to + create a local git repository from SVN. This documentation was + added via doxygen. (closes issue #15814) Reported by: tzafrir + Patches: git-asterisk-howto uploaded by tzafrir (license 46) + +2009-10-22 20:07 +0000 [r225446] Richard Mudgett + + * channels/sig_pri.c: Search for the subaddress only within the + extension section of the dial string. + Dial(DAHDI/(g|G|r|R)[c|r|d][/extension]) + +2009-10-22 19:55 +0000 [r225445] David Vossel + + * main/tcptls.c, channels/chan_sip.c, apps/app_externalivr.c, + include/asterisk/tcptls.h: SIP TCP/TLS: move client connection + setup/write into tcp helper thread, various related + locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS + connection setup into the TCP helper thread: Connection setup + takes awhile and before this it was being done while holding the + monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: + Through the use of a packet queue and an alert pipe, the TCP + helper thread can now be woken up to write data as well as read + data. 3.Locking error: sip_xmit returned an XMIT_ERROR without + giving up the tcptls_session lock. This lock has been completely + removed from sip_xmit and placed in the new sip_tcptls_write() + function. 4.Memory leak: When creating a tcptls_client the + tls_cfg was alloced but never freed unless the tcptls_session + failed to start. Now the session_args for a sip client are an ao2 + object which frees the tls_cfg on destruction. 5.Pointer to stack + variable: During sip_prepare_socket the creation of a client's + ast_tcptls_session_args was done on the stack and stored as a + pointer in the newly created tcptls_session. Depending on the + events that followed, there was a slight possibility that pointer + could have been accessed after the stack returned. Given the new + changes, it is always accessed after the stack returns which is + why I found it. Notable code changes 1.I broke tcptls.c's + ast_tcptls_client_start() function into two functions. One for + creating and allocating the new tcptls_session, and a separate + one for starting and handling the new connection. This allowed me + to create the tcptls_session, launch the helper thread, and then + establish the connection within the helper thread. 2.Writes to a + tcptls_session are now done within the helper thread. This is + done by using an alert pipe to wake up the thread if new data + needs to be sent. The thread's sip_threadinfo object contains the + alert pipe as well as the packet queue. 3.Since the threadinfo + object contains the alert pipe, it must now be accessed outside + of the helper thread for every write (queuing of a packet). For + easy lookup, I moved the threadinfo objects from a linked list to + an ao2_container. (closes issue #13136) Reported by: pabelanger + Tested by: dvossel, whys (closes issue #15894) Reported by: + dvossel Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/380/ + +2009-10-22 19:33 +0000 [r225440] Sean Bright + + * Makefile, utils/Makefile, utils/utils.xml (added), + doc/janitor-projects.txt: Add the programs in utils/ to + menuselect. Nothing in utils/ is now built by default except for + astcanary. Review: https://reviewboard.asterisk.org/r/353/ + +2009-10-22 19:10 +0000 [r225406] Tilghman Lesher + + * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: + Permit storage of voicemail secrets in a separate file, located + within the spool directory. (closes issue #14276) Reported by: + klaus3000 Patches: app_voicemail.c-svn-trunk-r214898.txt uploaded + by klaus3000 (license 65) Tested by: jamesgolovich + +2009-10-22 18:41 +0000 [r225405] Kevin P. Fleming + + * main/astobj2.c: Fix a refcount error introduced by yesterday's + OBJ_MULTIPLE commit. When an object is being unlinked from its + container *and* being returned to the caller, we do not want to + decrement the reference count after unlinking it from the + container, as the reference that the container held is what we + are returning to the caller... and if it was the only remaining + reference to the object, that could result in the object being + destroyed. + +2009-10-22 17:11 +0000 [r225360] Tilghman Lesher + + * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h: + Merged revisions 225105 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) + | 4 lines Fix documentation for ast_softhangup() and correct the + misuse thereof. (closes issue #16103) Reported by: majorbloodnok + ........ + +2009-10-22 16:33 +0000 [r225357] Richard Mudgett + + * main/channel.c, configure, include/asterisk/autoconfig.h.in, + configure.ac, funcs/func_connectedline.c, + include/asterisk/channel.h, CHANGES, channels/sig_pri.c, + funcs/func_callerid.c: Add support for calling and called + subaddress. Partial support for COLP subaddress. The Telecom + Specs in NZ suggests that SUB ADDRESS is always on, so doing + "desk to desk" between offices each with an asterisk box over the + ISDN should then be possible, without a whole load of DDI numbers + required. (closes issue #15604) Reported by: alecdavis Patches: + asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license + 585) Some minor modificatons were made. Tested by: alecdavis, + rmudgett Review: https://reviewboard.asterisk.org/r/405/ + +2009-10-21 21:58 +0000 [r225307] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 225243 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 + Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames + with no destination call number It is possible for the PBX thread + to queue up signaling frames before a destination call number is + received. This can result in signaling frames being sent out with + no destination call number. Since recent versions of Asterisk + require accurate destination callnumbers for all Full Frames, + this can cause a VNAK loop to occur. To resolve this no signaling + frames are sent until a destination callnumber is received, and + destination call numbers are now only required for iax_pvt + matching when the frame is an ACK. Review: + https://reviewboard.asterisk.org/r/413/ ........ + +2009-10-21 21:15 +0000 [r225244-225245] Kevin P. Fleming + + * doc/tex/manager.tex, channels/chan_sip.c: Add 'mohsuggest' + configuration option to 'sip show peer' CLI command and + SIPShowPeer AMI action. (closes issue #15990) Reported by: + _brent_ Patches: sip_peer_info_mohsuggest-r3.patch uploaded by + brent (license 388) Review: + https://reviewboard.asterisk.org/r/381/ + + * main/channel.c, main/manager.c, apps/app_directed_pickup.c, + apps/app_softhangup.c, funcs/func_channel.c, + include/asterisk/astobj2.h, res/snmp/agent.c, + include/asterisk/channel.h, include/asterisk/lock.h, + apps/app_chanspy.c, main/astobj2.c, main/cli.c: Finish + implementaton of astobj2 OBJ_MULTIPLE, and convert + ast_channel_iterator to use it. This patch finishes the + implementation of OBJ_MULTIPLE in astobj2 (the case where + multiple results need to be returned; OBJ_NODATA mode already was + supported). In addition, it converts ast_channel_iterators (only + the targeted versions, not the ones that iterate over all + channels) to use this method. During this work, I removed the + 'ao2_flags' arguments to the ast_channel_iterator constructor + functions; there were no uses of that argument yet, there is only + one possible flag to pass, and it made the iterators less + 'opaque'. If at some point in the future someone really needs an + ast_channel_iterator that does not lock the container, we can + provide constructor(s) for that purpose. Review: + https://reviewboard.asterisk.org/r/379/ + +2009-10-21 16:46 +0000 [r225170-225172] Russell Bryant + + * /, main/translate.c: Merged revisions 225171 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225171 | russell | 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009) + | 2 lines Revert 225169, as this doesn't account for the + possibility of a list of frames. ........ + + * /, main/translate.c: Merged revisions 225169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225169 | russell | 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009) + | 2 lines Isolate the frame returned from ast_translate(). + ........ + +2009-10-21 15:42 +0000 [r225102] Tilghman Lesher + + * apps/app_meetme.c: Apparently, I don't need to specify the ".so" + suffix to get a match + +2009-10-21 15:35 +0000 [r225089] Joshua Colp + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add + support for specifying the IP address to use for media streams in + sip.conf This is the second commit for this and documents the + text stream using the configured IP address and fixes a bug in + the original patch where the UDPTL stream would also use the + different IP address. (closes issue #14729) Reported by: _brent_ + Patches: media_address.patch uploaded by brent (license 388) + +2009-10-21 15:21 +0000 [r225048] Tilghman Lesher + + * apps/app_meetme.c, CHANGES: Turn on DENOISE filter for all + conference participants. (Fixes SWP-238) + +2009-10-21 15:04 +0000 [r225034] Joshua Colp + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Revert + media_address commit, I'm going to roll a fix to the SDP + generation in the next version. + +2009-10-21 14:39 +0000 [r225033] David Vossel + + * configs/iax.conf.sample, /, channels/chan_sip.c, + configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions + 225032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) + | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller + id removes '(', ' ', ')', non-trailing '.', and '-' from the + string. This means values such as 555.5555 and test-test result + in 555555 and testtest. There are instances, such as Skype + integration, where a specific value is passed via caller id that + must be preserved unmodified. This patch makes the shrinking of + caller id optional in chan_sip and chan_iax in order to support + such cases. By default this option is on to preserve previous + expected behavior. (closes issue #15940) Reported by: dimas + Patches: v2-15940.patch uploaded by dimas (license 88) + 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) + Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/408/ ........ + +2009-10-21 13:34 +0000 [r225003] Joshua Colp + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add + support for specifying the IP address to use for media streams in + sip.conf (closes issue #14729) Reported by: _brent_ Patches: + media_address.patch uploaded by brent (license 388) + +2009-10-21 03:09 +0000 [r224932] Russell Bryant + + * main/frame.c, /, main/translate.c, include/asterisk/dsp.h, + codecs/codec_dahdi.c, include/asterisk/frame.h, + include/asterisk/translate.h, main/dsp.c: Merged revisions 224931 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) + | 5 lines Isolate frames returned from a DSP instance or codec + translator. The reasoning for these changes are the same as what + I wrote in the commit message for rev 222878. ........ + +2009-10-21 02:43 +0000 [r224930] Richard Mudgett + + * channels/sig_pri.c: Make PRI_SUBCMD_xxx handling subaddress + friendly. + +2009-10-20 22:09 +0000 [r224856] Tilghman Lesher + + * funcs/func_speex.c, /, main/audiohook.c: Merged revisions 224855 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) + | 5 lines Pay attention to the return value of the manipulate + function. While this looks like an optimization, it prevents a + crash from occurring when used with certain audiohook callbacks + (diagnosed with SVN trunk, backported to 1.4 to keep the source + consistent across versions). ........ + +2009-10-20 17:47 +0000 [r224774] Joshua Colp + + * /, main/features.c: Merged revisions 224773 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 + lines Add support for relaying early media in the features + attended transfer option. (closes issue #14828) Reported by: + licedey ........ + +2009-10-20 12:44 +0000 [r224738] Matthew Nicholson + + * CHANGES: Added information to CHANGES about the dynamic range + compression feature added to dahdi. + +2009-10-19 23:47 +0000 [r224671] Kevin P. Fleming + + * res/res_rtp_asterisk.c, /: Merged revisions 224670 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 + Oct 2009) | 7 lines Correct timestamp calculations when RTP + sample rates over 8kHz are used. While testing some endpoints + that support 16kHz and 32kHz sample rates, some log messages were + generated due to calc_rxstamp() computing timestamps in a way + that produced odd results, so this patch sanitizes the result of + the computations. ........ + +2009-10-19 22:02 +0000 [r224637] Matthew Nicholson + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add + dynamic range compression support for analog channels. (closes + issue AST-29) + +2009-10-19 19:49 +0000 [r224567] Joshua Colp + + * apps/app_dial.c, /: Merged revisions 224565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 + lines Do not attempt early media bridging (ie: direct RTP setup) + if options are enabled that should prevent it. (closes issue + #14763) Reported by: cupotka ........ + +2009-10-19 19:40 +0000 [r224562] Kevin P. Fleming + + * formats/format_siren14.c: Remove useless debugging message. + +2009-10-19 15:50 +0000 [r224527] Tilghman Lesher + + * doc/janitor-projects.txt: Remove a completed project and add + another + +2009-10-19 14:32 +0000 [r224491] Joshua Colp + + * channels/sig_pri.h, channels/sig_pri.c: Add a callback to sig_pri + which is called when sig_pri is going to queue a control frame on + a channel. + +2009-10-19 00:05 +0000 [r224446-224448] Tilghman Lesher + + * apps/app_voicemail.c: Allow ODBC storage to be queried with + multiple mailboxes, and remove multiple goto's. This corrects an + issue reported on the -users list. + + * configs/res_odbc.conf.sample: Clarify that "forcecommit" is NOT + an alias for "autocommit", but instead controls the default + disposition of uncommitted transactions. + +2009-10-17 16:39 +0000 [r224403] Tilghman Lesher + + * include/asterisk/app.h, main/app.c: Remove unnecessary typedef + +2009-10-17 02:01 +0000 [r224331-224335] Jeff Peeler + + * channels/chan_dahdi.c: fix typo, sorry + + * channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions + 224330 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) + | 13 lines Fix stale caller id data from being reported in AMI + NewChannel event The problem here is that chan_dahdi is designed + in such a way to set certain values in the dahdi_pvt only once. + One of those such values is the configured caller id data in + chan_dahdi.conf. For PRI, the configured caller id data could be + overwritten during a call. Instead of saving the data and + restoring, it was decided that for all non-analog channels it was + simply best to not set the configured caller id in the first + place and also clear it at the end of the call. (closes issue + #15883) Reported by: jsmith ........ + +2009-10-16 20:40 +0000 [r224261] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 224260 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) + | 18 lines Never released PRI channels when using Busy() or + Congestion() dialplan apps. When the Busy() or Congestion() + application is used towards ISDN (an ISDN progress is sent), the + responding ISDN Disconnect or Release may contain the ISDN cause + user busy or one of the congestion causes. In chan_dahdi.c these + causes will only set the needbusy or needcongestion flags and not + activate the softhangup procedure. Unfortunately only the latter + can interrupt the endless wait loop of Busy()/Congestion(). + Result: PRI channels staying in state busy for the rest of + asterisk life or until the other end times out and forces the + call to clear. (issue #14292) Reported by: tomaso Patches: + disc_rel_userbusy.patch uploaded by tomaso (license 564) (This + patch is unrelated to the issue.) ........ + +2009-10-15 22:33 +0000 [r224225] Tilghman Lesher + + * include/asterisk/app.h, main/pbx.c, main/app.c: Create an API for + adding an optional time unit onto the ends of time periods. Two + examples of its use are included, and the usage could be expanded + in some cases into certain configuration options where time + periods are specified. + +2009-10-15 15:57 +0000 [r224178] Jeff Peeler + + * apps/app_chanspy.c: Readd removed ability to allow listening to + one side of the call in app_chanspy (Option o) (closes issue + #15675) Reported by: john8675309 Patches: + issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested + by: jgutierrez on users list: + http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html + +2009-10-15 14:37 +0000 [r224144] Doug Bailey + + * configs/chan_dahdi.conf.sample: chan_dahdi.conf.sample changes + for DTMF CID detect Explains new options for detecting DTMF CID + on fxo lines (issue #9096) Reported by: fleed Patches: + chan_dahid_sample_config.patch uploaded by sum (license 766) + +2009-10-15 06:48 +0000 [r224074-224109] Terry Wilson + + * res/res_calendar_caldav.c: Properly handle PUT requests for + CALENDAR_WRITE() + + * res/res_calendar.c: Add missing 'getnum' field + +2009-10-14 17:48 +0000 [r224035] Jeff Peeler + + * configs/sip_notify.conf.sample, channels/chan_sip.c, CHANGES: + Allow for adding message body to the SIP NOTIFY message Ability + has been added to both manager command SIPnotify as well as + console command sip notify. Message body is stored in the + "Content" variable. An example is present in sip_notify.conf. + (closes issue #13926) Reported by: jthurman Patches: + sip-notify-svn189463.diff uploaded by gareth (license 208) Tested + by: gareth + +2009-10-13 22:14 +0000 [r223992] Terry Wilson + + * res/res_calendar.c: use Calendar: instead of Calendar/ for + devstate + +2009-10-13 17:11 +0000 [r223911-223912] Richard Mudgett + + * include/asterisk/pbx.h: Fix some doxygen format problems and trim + trailing whitespace. + + * res/res_calendar.c: Fix compiler warning. + +2009-10-13 01:58 +0000 [r223874-223875] Terry Wilson + + * apps/app_originate.c: Revert inadvertant code commit to + app_originate + + * apps/app_originate.c, include/asterisk/calendar.h, + res/res_calendar.c: Fix handling of notification calls w/ the + dialing api + +2009-10-12 23:48 +0000 [r223832] Jeff Peeler + + * apps/app_dial.c, /: Merged revisions 223804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) + | 8 lines Ensure ringing continues for branched calls after + progress is received While waiting for an answer, don't send + progress for branched calls for which ringing was sent. (closes + issue #15028) Reported by: fnordian ........ + +2009-10-12 20:58 +0000 [r223756] David Vossel + + * configs/iax.conf.sample: Clarifies trunkmaxsize, trunkfreq, and + trunkmtu iax2 options SWP-151 + +2009-10-12 15:32 +0000 [r223652-223693] Kevin P. Fleming + + * /: Recorded merge of revisions 223692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223692 | kpfleming | 2009-10-12 10:30:40 -0500 (Mon, 12 Oct + 2009) | 13 lines Remove automatic switching from T.38 to voice + mode in chan_sip. chan_sip has some code to automatically switch + from T.38 mode to voice mode when a voice frame is written to the + channel while it is in T.38 mode; this was intended to handle the + situation when a FAX transmission has ended and the channel is + not yet hung up, but is causing problems at the beginning of FAX + sessions as well when there are still voice frames 'in flight' at + the time the T.38 negotiation completes. This patch removes the + automatic switchover. (issue #16025) Reported by: jamicque + ........ + + * channels/chan_sip.c, apps/app_fax.c: Remove automatic switching + from T.38 to voice mode in chan_sip. chan_sip has some code to + automatically switch from T.38 mode to voice mode when a voice + frame is written to the channel while it is in T.38 mode; this + was intended to handle the situation when a FAX transmission has + ended and the channel is not yet hung up, but is causing problems + at the beginning of FAX sessions as well when there are still + voice frames 'in flight' at the time the T.38 negotiation + completes. This patch removes the automatic switchover, and + changes app_fax to explicitly switch off T.38 mode when the FAX + transmission process ends. (closes issue #16025) Reported by: + jamicque + +2009-10-11 22:19 +0000 [r223617] Mark Michelson + + * channels/chan_sip.c: Check the proper page for the SENDRPID flag. + If a pending reinvite were sent, we might not properly send + connected party info since we were checking the wrong flag. This + was a rare occurrence, but could still happen nevertheless. + +2009-10-11 18:35 +0000 [r223487-223553] Russell Bryant + + * /: Merged revisions 223550 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223550 | russell | 2009-10-11 13:34:37 -0500 (Sun, 11 Oct 2009) + | 2 lines Remove a duplicate ao2_iterator_destroy(). ........ + + * main/autoservice.c, /: Merged revisions 223485-223486 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) + | 6 lines Don't use data outside of its scope. The purpose of + this code was to have a hangup frame put on the list of deferred + frames. However, the code that read the hangup frame was outside + of the scope of where the hangup frame was declared. ........ + r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) + | 2 lines Remove some unnecessary code. ........ + +2009-10-10 20:02 +0000 [r223449] Terry Wilson + + * res/res_calendar_icalendar.c, res/res_calendar_caldav.c: Fix + handling of floating times and dates + +2009-10-10 08:30 +0000 [r223413-223415] Olle Johansson + + * configs/cdr_pgsql.conf.sample: Adding note about TLS usage + + * configs/res_ldap.conf.sample: Add an additional note on TLS + support + + * configs/res_ldap.conf.sample: Adding some information on TLS + support + +2009-10-09 22:04 +0000 [r223370] Terry Wilson + + * res/res_calendar_icalendar.c, res/res_calendar_caldav.c: Properly + return "free" on confirmed events that are free CONFIRMED status + doesn't imply busy or free, that is handled with the TRANSP + field. Luckily, libical already sets the is_busy status on the + span for us. + +2009-10-09 20:58 +0000 [r223330] Kevin P. Fleming + + * apps/app_fax.c: Initiate T.38 switchover when acting as called + party, regardless of FAX direction. SendFAX() and ReceiveFAX() + can be given options to indicate whether they should act as the + calling or called party; this mode should be used to decide + whether to initiate a switchover to T.38, not the direction that + the FAX transfer will take place. (closes issue #16039) Reported + by: jamicque + +2009-10-09 18:34 +0000 [r223273] Matthew Nicholson + + * main/channel.c, /: Merged revisions 223225 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct + 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING + when originating calls. (closes issue #15104) Reported by: + nblasgen Patches: manager-timeout1.diff uploaded by mnicholson + (license 96) Tested by: nblasgen, mnicholson ........ + +2009-10-09 18:17 +0000 [r223211-223215] Mark Michelson + + * /: Recorded merge of revisions 223213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct + 2009) | 3 lines Fix potential memory leak in app_dial.c ........ + + * apps/app_dial.c: Fix potential memory leaks. ABE-1998 + +2009-10-09 17:53 +0000 [r223206] David Vossel + + * /, channels/chan_sip.c: Merged revisions 223205 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) + | 10 lines fixes sip registration using authuser in user.conf + (closes issue #14954) Reported by: tornblad Tested by: + mmichelson, tornblad, dvossel ........ + +2009-10-09 17:14 +0000 [r223136] Matthew Nicholson + + * cdr/cdr_sqlite3_custom.c: Don't close the sqlite database when + reloading. Only close the database when unloading. (closes issue + #15953) Reported by: frawd Patches: sqlite3_rev220097.diff + uploaded by frawd (license 610) Tested by: frawd + +2009-10-09 16:54 +0000 [r223088-223132] David Vossel + + * channels/chan_sip.c: 'auth=' did not parse md5 secret correctly + (closes issue #15949) Reported by: ebroad Patches: + authparsefix.patch uploaded by ebroad (license 878) + 15949_trunk.diff uploaded by dvossel (license 671) Tested by: + ebroad + + * channels/chan_sip.c: p->peerauth is always empty in + transmit_register() When using callbackextension or specifing the + peer name in a registration string, the peer's specific auth + settings set by the "auth=" strings within the peer definition + are not used by the registration. Thanks to ebroad for reporting + the issue and providing the patch. (closes issue #15955) Reported + by: ebroad Patches: regauthfix.patch uploaded by ebroad (license + 878) + +2009-10-09 15:00 +0000 [r223016-223053] Terry Wilson + + * res/res_calendar.c: Don't add Attendees during copy, replace them + + * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, + res/res_calendar_caldav.c, include/asterisk/calendar.h, + res/res_calendar.c: Remove global variable that makes dlopen + unhappy This isn't the best way to do this, but it is the + easiest. There are some limitations that are going to need to be + addressed at some point with reloads and when I (or someone else) + work on that, then the API can be updated to handle passing the + private config data that the calendar tech modules need in a + better way as well. + +2009-10-08 22:57 +0000 [r222947-223015] David Vossel + + * channels/chan_sip.c: fixed comment line for do_magic_pickup + + * channels/chan_sip.c: Deadlock between ast_cel_report_event and + ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner + channel while only the pvt lock is held. Since pbx_exec calls + ast_cel_report_event which attempts to lock the channel, invalid + locking order occurs. Channels should be locked before pvt's. + (closes issue #15512) Reported by: lmsteffan Patches: + ast_cel_deadlock_15512.diff uploaded by dvossel (license 671) + + * channels/chan_sip.c: makes externtcpport and externtlsport static + variables externtcpport and externtlsport need to be declared as + static variables. Thanks to russell for finding and pointing this + out. + +2009-10-08 19:52 +0000 [r222880] Russell Bryant + + * include/asterisk/file.h, main/frame.c, /, main/file.c, + include/asterisk/frame.h: Merged revisions 222878 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 + Oct 2009) | 44 lines Make filestream frame handling safer by + isolating frames before returning them. This patch is related to + a number of issues on the bug tracker that show crashes related + to freeing frames that came from a filestream. A number of fixes + have been made over time while trying to figure out these + problems, but there re still people seeing the crash. (Note that + some of these bug reports include information about other + problems. I am specifically addressing the filestream frame crash + here.) I'm still not clear on what the exact problem is. However, + what is _very_ clear is that we have seen quite a few problems + over time related to unexpected behavior when we try to use + embedded frames as an optimization. In some cases, this + optimization doesn't really provide much due to improvements made + in other areas. In this case, the patch modifies filestream + handling such that the embedded frame will not be returned. + ast_frisolate() is used to ensure that we end up with a + completely mallocd frame. In reality, though, we will not + actually have to malloc every time. For filestreams, the frame + will almost always be allocated and freed in the same thread. + That means that the thread local frame cache will be used. So, + going this route doesn't hurt. With this patch in place, some + people have reported success in not seeing the crash anymore. + (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon + Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell + (license 2) Tested by: aragon, russell (closes issue #15817) + Reported by: zerohalo Tested by: zerohalo (closes issue #15845) + Reported by: marhbere Review: + https://reviewboard.asterisk.org/r/386/ ........ + +2009-10-08 19:35 +0000 [r222873] David Vossel + + * include/asterisk/netsock.h, main/netsock.c: fixes an + ast_netsock_list memory leak. ABE-1998 Review: + https://reviewboard.asterisk.org/r/395/ + +2009-10-08 16:44 +0000 [r222799] Richard Mudgett + + * /, channels/misdn_config.c: Merged revisions 222797 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 + Oct 2009) | 12 lines Fix memory leak if chan_misdn config + parameter is repeated. Memory leak when the same config option is + set more than once in an misdn.conf section. Why must this be + considered? Templates! Defining a template with default port + options and later adding to or overriding some of them. Patches: + memleak-misdn.patch JIRA ABE-1998 ........ + +2009-10-07 22:58 +0000 [r222761] David Vossel + + * main/channel.c, main/pbx.c, channels/chan_misdn.c, + channels/chan_sip.c, main/features.c, include/asterisk/channel.h: + Deadlock in channel masquerade handling Channels are stored in an + ao2_container. When accessing an item within an ao2_container the + proper locking order is to first lock the container, and then the + items within it. In ast_do_masquerade both the clone and original + channel must be locked for the entire duration of the function. + The problem with this is that it attemptes to unlink and link + these channels back into the ao2_container when one of the + channel's name changes. This is invalid locking order as the + process of unlinking and linking will lock the ao2_container + while the channels are locked!!! Now, both the channels in + do_masquerade are unlinked from the ao2_container and then locked + for the entire function. At the end of the function both channels + are unlocked and linked back into the container with their new + names as hash values. This new method of requiring all channels + and tech pvts to be unlocked before ast_do_masquerade() or + ast_change_name() required several changes throughout the code + base. (closes issue #15911) Reported by: russell Patches: + masq_deadlock_trunk.diff uploaded by dvossel (license 671) Tested + by: dvossel, atis (closes issue #15618) Reported by: lmsteffan + Patches: deadlock_local_attended_transfers_trunk.diff uploaded by + dvossel (license 671) Tested by: lmsteffan, dvossel Review: + https://reviewboard.asterisk.org/r/387/ + +2009-10-07 21:56 +0000 [r222692] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 222691 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 + Oct 2009) | 14 lines chan_misdn.c:process_ast_dsp() memory leak + misdn.conf: astdtmf must be set to "yes". With "no", buffer loss + does not occur. The translated frame "f2" when passing through + ast_dsp_process() is not freed whenever it is not used further in + process_ast_dsp(). Then in the end it is never ever freed. + Patches: translate.patch JIRA ABE-1993 ........ + +2009-10-07 20:08 +0000 [r222652] Jeff Peeler + + * channels/chan_dahdi.c: Change ringt (ring timeout) styles to be + consistent across chan_dahdi. (closes issue #15684) Reported by: + alecdavis Patches: chan_dahdi.bug15684.diff2.txt uploaded by + alecdavis (license 585) Tested by: alecdavis + +2009-10-07 18:57 +0000 [r222614-222615] Olle Johansson + + * res/res_config_ldap.c: Formatting, moving error messages to + ERROR, removing references to unexisting debug output. No + functionality changes. + + * cel/cel_pgsql.c, res/res_config_pgsql.c, cdr/cdr_pgsql.c: Use + extref for doxygen references to external libraries (in this case + PostgreSQL) + +2009-10-07 18:04 +0000 [r222548] Jason Parker + + * configs/queues.conf.sample: Remove 'keepstats' queue option from + sample config, as it's no longer used. + https://reviewboard.asterisk.org/r/115/ (closes issue #15820) + Reported by: kshumard + +2009-10-07 17:44 +0000 [r222543] David Vossel + + * /, channels/chan_sip.c: Merged revisions 222542 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) + | 8 lines crash on transfer handle_invite_replaces() attempts to + uplock a pvt's owner channel without first verifing that it + exists. (issue #16027) ........ + +2009-10-06 23:56 +0000 [r222463] Jeff Peeler + + * channels/chan_dahdi.c, /: Merged revisions 222462 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 + Oct 2009) | 8 lines Add missing unlock(s) in dahdi_read (two + cases in trunk) (closes issue #15683) Reported by: alecdavis + ........ + +2009-10-06 22:49 +0000 [r222398-222399] David Vossel + + * CHANGES: Updates CHANGES to reflect the new externtcpport and + externtlsport sip options + + * channels/chan_sip.c, configs/sip.conf.sample: contact header port + ignored transport when using externip This patch adds support for + TCP/TLS in the Contact header when using NAT, specifically + externip or externhost. The original issue was that Asterisk sent + 5060 as the port in the contact header whether TLS was used or + not. Additionally, this patch adds 2 config options to sip.conf, + specifically externtcpport and externtlsport. This allows a user + to specify different external ports for TCP and TLS other than + those used internally, this is especially useful in in a PAT/port + redirection setup. Thanks to ebroad for reporting the issue and + providing the patch! (closes issue #15880) Reported by: ebroad + Patches: portmap.patch uploaded by ebroad (license 878) + externtXXport_v2.patch uploaded by ebroad (license 878) Tested + by: ebroad Review: https://reviewboard.asterisk.org/r/392/ + +2009-10-06 20:35 +0000 [r222351] Jeff Peeler + + * channels/chan_dahdi.c: Fix 222298 (crash during destruction of + second channel when variable set with setvar). I mistakenly + reasoned that setvar would be used on all channels. Since it can + be set per channel, give each dahdi channel a copy of the + variable. (related to #15899) + +2009-10-06 19:31 +0000 [r222309] Tilghman Lesher + + * res/res_config_pgsql.c, cdr/cdr_pgsql.c: Change schema query to + involve the use of an optional schema parameter. This change is + done in such a way as to allow the driver to continue to function + with older databases which don't have these features. (closes + issue #16000) Reported by: jamicque Patches: + 20091002__issue16000.diff.txt uploaded by tilghman (license 14) + 20091002__issue16000__1.6.1.diff.txt uploaded by tilghman + (license 14) Tested by: jamicque + +2009-10-06 19:24 +0000 [r222298] Jeff Peeler + + * channels/chan_dahdi.c: Fix crash during destruction of second + channel when variable set with setvar. The setvar line in + chan_dahdi.conf is shared among all the channels, so make sure to + only free the resources only when the last channel is destroyed. + (closes issue #15899) Reported by: tzafrir + +2009-10-06 19:17 +0000 [r222273] Tilghman Lesher + + * res/ael/pval.c: When we call a gosub routine, the variables + should be scoped to avoid contaminating the caller. This affected + the ~~EXTEN~~ hack, where a subroutine might have changed the + value before it was used in the caller. Patch by myself, tested + by ebroad on #asterisk + +2009-10-06 16:17 +0000 [r222237] Tzafrir Cohen + + * channels/chan_dahdi.c: Make sure digit events are not reported as + "ERROR" dahdievent_to_analogevent used a simple switch statement + to convert DAHDI event numbers to "ANALOG_*" event numbers. + However "digit" events (DAHDI_EVENT_PULSEDIGIT, + DAHDI_EVENT_DTMFDOWN, DAHDI_EVENT_DTMFUP) are accompannied by the + digit in the low word of the event number. This fix makes + dahdievent_to_analogevent() return the event number as-is for + such an event. This is also required to fix #15924 (in addition + to r222108). + +2009-10-06 01:24 +0000 [r222110-222176] Kevin P. Fleming + + * /, channels/chan_sip.c, funcs/func_dialgroup.c, + include/asterisk/astobj2.h, res/res_phoneprov.c, + channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c, + channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c, + res/res_calendar.c, res/res_clialiases.c: Recorded merge of + revisions 222152 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct + 2009) | 20 lines Fix ao2_iterator API to hold references to + containers being iterated. See Mantis issue for details of what + prompted this change. Additional notes: This patch changes the + ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum + instead of a macro, with a name that fits our naming policy; + also, it is now necessary to call ao2_iterator_destroy() on any + iterator that has been created. Currently this only releases the + reference to the container being iterated, but in the future this + could also release other resources used by the iterator, if the + iterator implementation changes to use additional resources. + (closes issue #15987) Reported by: kpfleming Review: + https://reviewboard.asterisk.org/r/383/ ........ + + * main/udptl.c, channels/chan_sip.c, configs/udptl.conf.sample, + UPGRADE.txt, configs/sip.conf.sample: Allow non-compliant T.38 + endpoints to be supportable via configuration option. Many T.38 + endpoints incorrectly send the maximum IFP frame size they can + accept as the T38FaxMaxDatagram value in their SDP, when in fact + this value is supposed to be the maximum UDPTL payload size + (datagram size) they can accept. If the value they supply is + small enough (a commonly supplied value is '72'), T.38 UDPTL + transmissions will likely fail completely because the UDPTL + packets will not have enough room for a primary IFP frame and the + redundancy used for error correction. If this occurs, the + Asterisk UDPTL stack will emit log messages warning that data + loss may occur, and that the value may need to be overridden. + This patch extends the 't38pt_udptl' configuration option in + sip.conf to allow the administrator to override the value + supplied by the remote endpoint and supply a value that allows + T.38 FAX transmissions to be successful with that endpoint. In + addition, in any SIP call where the override takes effect, a + debug message will be printed to that effect. This patch also + removes the T38FaxMaxDatagram configuration option from + udptl.conf.sample, since it has not actually had any effect for a + number of releases. In addition, this patch cleans up the T.38 + documentation in sip.conf.sample (which incorrectly documented + that T.38 support was passthrough only). (issue #15586) Reported + by: globalnetinc + +2009-10-05 19:20 +0000 [r222108] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h: Add a few missing events to + analog_handle_event. The reported bug was actually only for + pulsedigit, dtmfup, and dtmfdown handling. Also added recognition + for fax events (just some verbose output) and fixed handling for + the ec_disabled_event. In order to make comparing the analog + version of events to the DAHDI events easier, the ordering has + been changed to follow that of the DAHDI events. (closes issue + #15924) Reported by: tzafrir + +2009-10-02 17:34 +0000 [r222030] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 222026 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 + Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a + memcpy. ........ + +2009-10-02 16:59 +0000 [r221920-221971] Tilghman Lesher + + * /, main/astobj2.c: Merged revisions 221970 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) + | 2 lines Ensure the result of the hash function is positive. + Negative array offsets suck. ........ + + * main/logger.c: Initialize a variable that we check immediately + upon startup. (closes issue #15973) Reported by: atis + +2009-10-02 01:49 +0000 [r221844-221881] Richard Mudgett + + * channels/misdn/isdn_lib.c: Whitespace change. + + * channels/misdn/isdn_lib.c: Whitespace change. + + * channels/misdn/isdn_lib_intern.h, /, channels/misdn/isdn_lib.c: + Merged revisions 221769 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) + | 26 lines Occasionally losing use of B channels in chan_misdn. I + have not been able to reproduce the problem of losing channels. + However, I have seen in the code a reentrancy problem that might + give these symptoms. The reentrancy patch does several things: 1) + Guards B channel and B channel structure allocation. 2) Makes the + B channel structure find routines more precise in locating + records. 3) Never leave a B channel allocated if we received + cause 44. The last item may cause temporary outgoing call + problems, but they should clear when the line becomes idle. + (closes issue #15490) Reported by: slutec18 Patches: + issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett + (license 664) Tested by: rmudgett, slutec18 (closes issue #15458) + Reported by: FabienToune Patches: + issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett + (license 664) Tested by: FabienToune, rmudgett, slutec18 ........ + +2009-10-02 00:08 +0000 [r221777-221781] Tilghman Lesher + + * main/say.c: One more off-by-one in trunk + + * main/rtp_engine.c, /, main/say.c, main/asterisk.c: Merged + revisions 221776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) + | 2 lines Fix a bunch of off-by-one errors ........ + +2009-10-01 20:18 +0000 [r221709] Richard Mudgett + + * UPGRADE.txt, CHANGES: Move DAHDI/ISDN channel naming note from + CHANGES to UPGRADE.txt. + +2009-10-01 20:09 +0000 [r221705] Tilghman Lesher + + * channels/chan_sip.c: Revision 220906 (a merge from 1.4) was not + merged correctly, causing a problem with non-dynamic peers. + +2009-10-01 19:48 +0000 [r221701] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, CHANGES: Prevent + deadlock if chan_dahdi attempts to change PRI channel names. The + PRI channels can no longer change the channel name if a different + B channel is selected during call negotiation. To prevent using + the channel name to infer what B channel a call is using and to + avoid name collisions, the channel name format is changed. The + new channel naming for PRI channels is: + DAHDI/ISDN-- + +2009-10-01 19:33 +0000 [r221697] David Vossel + + * channels/chan_sip.c: outbound tls connections were not defaulting + to port 5061 (closes issue #15854) Reported by: dvossel Patches: + sip_port_config_trunk.diff uploaded by dvossel (license 671) + Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/357/ + +2009-10-01 16:27 +0000 [r221592-221627] Kevin P. Fleming + + * UPGRADE.txt: Sync up UPGRADE.txt with the 1.6.2 version. + + * main/udptl.c, configs/udptl.conf.sample: Remove ability to + control T.38 FAX error correction from udptl.conf. chan_sip has + had the ability to control T.38 FAX error correction mode on a + per-peer (or global) basis for a couple of releases now, which is + where it should have been all along. This patch removes the + ability to configure it in udptl.conf, but issues a warning if + the user tries to do, telling them to look at sip.conf.sample for + how to configure it now. For any SIP peers that are T.38 enabled + in sip.conf, there is already a default for FEC error correction + even if the user does not specify any mode, so this change will + not turn off error correction by default, it will have the same + default value that has been in the udptl.conf sample file. + +2009-10-01 15:26 +0000 [r221589] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 221588 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct + 2009) | 2 lines Use unsigned ints for portinuri flags. ........ + +2009-10-01 07:00 +0000 [r221554] Olle Johansson + + * channels/chan_sip.c: Simplify code for porturi, use TRUE/FALSE + constructs when it's just TRUE or FALSE. + +2009-09-30 23:04 +0000 [r221484] Matthew Nicholson + + * channels/chan_sip.c: Cleaned up merge from r221432 + +2009-09-30 21:15 +0000 [r221436] Matthias Nick + + * apps/app_queue.c: Prevents from division by zero + +2009-09-30 20:40 +0000 [r221432] Matthew Nicholson + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 221360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep + 2009) | 10 lines Fix SRV lookup and Request-URI generation in + chan_sip. This patch adds a new field "portinuri" to the sip + dialog struct and the sip peer struct. That field is used during + RURI generation to determine if the port should be included in + the RURI. It is also used in some places to determine if an SRV + lookup should occur. (closes issue #14418) Reported by: klaus3000 + Tested by: klaus3000, mnicholson Review: + https://reviewboard.asterisk.org/r/369/ ........ + +2009-09-30 19:42 +0000 [r221368] Matthias Nick + + * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged + revisions 221153,221157,221303 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | + 2 lines check bounds - prevents for buffer overflow ........ + r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | + 8 lines added a new dialplan function 'CSV_QUOTE' and changed the + cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr + Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: + mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, + 30 Sep 2009) | 2 lines changed the prototype definition of + csv_quote ........ + +2009-09-30 18:47 +0000 [r221266-221300] Terry Wilson + + * res/res_rtp_asterisk.c: Remove spurious debug + + * res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c, + include/asterisk/rtp_engine.h: Use rtp properties instead of + adding a callback Thanks, Josh. + + * res/res_rtp_asterisk.c, main/rtp_engine.c, /, + channels/chan_sip.c, configs/sip.conf.sample, + include/asterisk/rtp_engine.h: Merged revisions 221086 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) + | 25 lines Change the SSRC by default when our media stream + changes Be default, change SSRC when doing an audio stream + changes Asterisk doesn't honor marker bit when reinvited to + already-bridged RTP streams,resulting in far-end stack discarding + packets with "old" timestamps that areactually part of a new + stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is + a reinvite, unless the 'constantssrc' is set to true in sip.conf. + The original issue reported to Digium support detailed the + following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based + Application Server Call comes in fromITSP, Asterisk dials the app + server which sends a re-invite back toAsterisk--not to negotiate + to send media directly to the ITSP, but to indicatethat it's + changing the stream it's sending to Asterisk. The app + servergenerates a new SSRC, sequence numbers, timestamps, and + sets the marker bit on the new stream. Asterisk passes through + the teimstamp of the new stream, butdoes not reset the SSRC, + sequence numbers, or set the marker bit. When the timestamp on + the new stream is older than the timestamp on the originalstream, + the ITSP (which doesn't know there has been any change) discards + the newframes because it thinks they are too old. This patch + addresses this by changing the SSRC on a stream update unless + constantssrc=true is set in sip.conf. Review: + https://reviewboard.asterisk.org/r/374/ ........ + +2009-09-30 16:56 +0000 [r221201] Tilghman Lesher + + * main/channel.c, /: Merged revisions 221200 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) + | 7 lines Avoid a potential NULL dereference. (closes issue + #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt + uploaded by tilghman (license 14) Tested by: kobaz ........ + +2009-09-30 15:11 +0000 [r221085-221090] Sean Bright + + * apps/app_voicemail.c: Modify VoiceMailMain()'s a() argument to + allow mailboxes to be specified by name. (closes issue #14740) + Reported by: pj Patches: issue14740_09022009.diff uploaded by + seanbright (license 71) Tested by: seanbright, lmadsen + + * apps/app_voicemail.c: Clarify documentation for VoiceMailMain()'s + a() option. We require box numbers, not names as the + documentation implies. (issue #14740) Reported by: pj Patches: + __20090729-app_voicemail-documentation.patch uploaded by lmadsen + (license 10) Tested by: seanbright, lmadsen + +2009-09-30 04:32 +0000 [r221044] Tilghman Lesher + + * funcs/func_lock.c: Allow locks to be inherited through a + masquerade without causing starvation. (closes issue #14859) + Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded + by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt + uploaded by tilghman (license 14) Tested by: atis, tilghman + +2009-09-29 21:28 +0000 [r220920-220995] Mark Michelson + + * main/cel.c: Fix channel reference leak. ast_cel_report_event + would geet a reference to the bridged channel. However, certain + return paths, such as if CEL was not enabled, would result in a + reference leak. All return paths now properly unref the channel. + (closes issue #15991) Reported by: mmichelson + + * main/rtp_engine.c: Get rid of annoying and cryptic debug + messages. + +2009-09-29 19:57 +0000 [r220906] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 220873 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) + | 9 lines Reduce CPU usage related to building a peer merely for + devicestates. This fixes a 100% CPU problem in the SIP driver, + found by profiling the driver while the problem was occurring. + (closes issue #14309) Reported by: pkempgen Patches: + 20090924__issue14309.diff.txt uploaded by tilghman (license 14) + Tested by: pkempgen, vrban ........ + +2009-09-29 19:49 +0000 [r220904] Matthew Nicholson + + * apps/app_confbridge.c: Fix options 'm' and 's'. They were swapped + in the code. Also document the fact that app_confbridge does not + automatically answer the channel. (closes issue #15964) Reported + by: shrift + +2009-09-29 16:58 +0000 [r220833] Jeff Peeler + + * apps/app_voicemail.c: Make deletion of temporary greetings work + properly with IMAP_STORAGE When imapgreetings was set to yes, the + message was being deleted but wasn't actually being expunged. + When imapgreetings was set to no, the file based message was not + being deleted at all. All good now! (closes issue #14949) + Reported by: noahisaac Patches: vm_tempgreeting_removal.patch + uploaded by noahisaac (license 748), modified by me + +2009-09-28 21:02 +0000 [r220792] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_pri.c: Miscellaneous minor + changes. + +2009-09-28 19:11 +0000 [r220721] Sean Bright + + * /, Makefile.rules: Merged revisions 220717 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep + 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect, + explicitly pass -O0 to the compiler so we override any default + optimization levels for a particular install. ........ + +2009-09-28 19:10 +0000 [r220718] Jeff Peeler + + * channels/chan_sip.c: Fix building of registration entry in + build_peer when using callbackextension Check for remotesecret + option was unintentionally always true, which therefore caused + the secret option to never be used. Thanks to dvossel for + pointing out the exact fix. (closes issue #15943) Reported by: + tpsast + +2009-09-28 15:27 +0000 [r220672] Richard Mudgett + + * channels/sig_pri.h, channels/sig_pri.c: Locking issues dealing + with service_lock. * Removed unneeded and uninitialized + service_lock. * Fixed potential locking imbalance in + pri_dchannel():PRI_EVENT_RESTART. * Fixed verbose message typo in + pri_dchannel():PRI_EVENT_RESTART. + +2009-09-27 20:40 +0000 [r220629] Michiel van Baak + + * funcs/func_callerid.c: add name argument for the CALLERID + dialplan function to the xml documentation. Pointed out to me on + IRC by snuff-home. Thanks + +2009-09-26 15:10 +0000 [r220586] Tilghman Lesher + + * include/asterisk/aes.h: Allow AES to compile, when OpenSSL is not + present. + +2009-09-25 19:56 +0000 [r220543] Richard Mudgett + + * channels/sig_pri.c: Reduce indentation in sig_pri_available(). + +2009-09-25 14:50 +0000 [r220494-220496] Kevin P. Fleming + + * main/manager.c: Eliminate unnecessary include of version.h in + manager.c. Including version.h here causes this file to get + recompiled after every commit or update, which is not needed. + + * main/channel.c: Correct sense of logic test committed in revision + 220494. + + * main/channel.c: Don't use hash-based lookups for + ast_channel_get_by_name_prefix(). ast_channel_get_full() tries to + use OBJ_POINTER to optimize name-based channel lookups, but this + will not work properly when the channel's full name was not + supplied; for name-prefix searches, there is no value in doing a + hash-based lookup, and in fact doing so could result in many + channels being skipped. + +2009-09-25 10:54 +0000 [r220457] Philippe Sultan + + * channels/chan_jingle.c, configs/jabber.conf.sample, + include/asterisk/jabber.h, channels/chan_gtalk.c, CHANGES, + doc/jabber.txt, res/res_jabber.c: Add JABBER_RECEIVE as a + dialplan function, implement SendText in Jingle channels + JABBER_RECEIVE (along with JabberSend) makes Asterisk interact + with users over XMPP to process calls. SendText can be used + instead of JabberSend in the context of XMPP based voice channels + (chan_gtalk and chan_jingle). (closes issue #12569) Reported by: + eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo + Review: https://reviewboard.asterisk.org/r/88/ + +2009-09-24 22:53 +0000 [r220417] Tilghman Lesher + + * UPGRADE.txt, main/asterisk.c: Change the default behavior of Set, + AGI, and pbx_realtime to 1.6 behavior by default (starting in + 1.6.3). + +2009-09-24 20:37 +0000 [r220365] David Vossel + + * main/tcptls.c: fixes tcptls_session memory leak caused by ref + count error (closes issue #15939) Reported by: dvossel Review: + https://reviewboard.asterisk.org/r/375/ + +2009-09-24 20:29 +0000 [r220344] Jeff Peeler + + * apps/app_dial.c, main/features.c, include/asterisk/features.h: + Add bridge related dial flags to the bridge app Most of the + functionality here is gained simply by setting the feature flag + on the bridge config. However, the dial limit functionality has + been moved from app_dial to the features code and has been made + public so both app_dial and the bridge app can use it. (closes + issue #13165) Reported by: tim_ringenbach Patches: + app_bridge_options_r138998.diff uploaded by tim ringenbach + (license 540), modified by me + +2009-09-24 19:57 +0000 [r220295] Olle Johansson + + * configs/sip.conf.sample: Documentation in the commit messages is + soon forgotten, please add it to the docs in the product. + +2009-09-24 19:41 +0000 [r220289] Tilghman Lesher + + * main/pbx.c, /, apps/app_disa.c, apps/app_playback.c: Merged + revisions 220288 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) + | 6 lines Implicitly sending a progress signal breaks some + applications. Call Progress() in your dialplan if you explicitly + want progress to be sent. (Reverts change 216430, closes issue + #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing + list + http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html + ........ + +2009-09-24 18:19 +0000 [r220217] Sean Bright + + * Makefile, /: Merged revisions 220213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep + 2009) | 1 line Resolve parallel build warnings. Reported by Klaus + Darilion on the asterisk-dev mailing list. ........ + +2009-09-24 16:33 +0000 [r220174] Matthew Nicholson + + * channels/chan_sip.c: Ensure the numeric portion of the + P-Asserted-Identity header is properly escaped. + +2009-09-24 14:44 +0000 [r220100] Sean Bright + + * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220099 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep + 2009) | 2 lines Remove the remaining bashisms in the + Makefile/mkpkgconfig ........ + +2009-09-24 08:36 +0000 [r220028] Michiel van Baak + + * build_tools/mkpkgconfig, /: Merged revisions 220027 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 + Sep 2009) | 7 lines mkpkgconfig does not need bash so make it use + /bin/sh This fixes building on all systems that don't have bash + at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on + #asterisk-dev ........ + +2009-09-24 07:39 +0000 [r219951-219987] Tilghman Lesher + + * apps/app_directory.c: Fix two possible crashes, one only in 1.6.1 + and one in 1.6.1 forward. (closes issue #15739) Reported by: + DLNoah, jeffg Patches: 20090914__issue15739.diff.txt uploaded by + tilghman (license 14) 20090922__issue15739.diff.txt uploaded by + tilghman (license 14) Tested by: DLNoah, jeffg + + * configs/mgcp.conf.sample, CHANGES, channels/chan_mgcp.c: Add + support for 'setvar=' for MGCP device lines, like other channel + drivers provide. (closes issue #14818) Reported by: + alea-soluciones Patches: + chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea + (license 514) + + * doc/lang/language-criteria.txt: Update fax number to the legal + fax, not the generic fax. (closes issue #15946) Reported by: + jtodd Patches: leif-is-a-wuss.txt uploaded by jtodd (license 870) + Tested by: jparker, tilghman, jtodd, russellb, mmichelson, + seanbright, kpfleming, and the rest of the usual suspects + +2009-09-23 17:46 +0000 [r219895] Leif Madsen + + * include/asterisk/doxyref.h, + include/asterisk/doxygen/mantisworkflow.h (added): Add Mantis + work flow documention. This commit adds the doxygen changes that + I've made to describe the Mantis work flow documentation for the + open source issue tracker. This should make it easier to + determine the flow of issues through the issue tracker, and what + those statuses mean. (closes issue #15902) Reported by: lmadsen + Patches: mantisworkflow.h uploaded by lmadsen (license 10) + Review: https://reviewboard.asterisk.org/r/367/ + +2009-09-22 21:43 +0000 [r219818] Tilghman Lesher + + * /, apps/app_voicemail.c: Merged revisions 219816 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 + Sep 2009) | 10 lines When IMAP variables were changed during a + reload, Voicemail did not use the new values. This change + introduces a configuration version variable, which ensures that + connections with the old values are not reused but are allowed to + expire normally. (closes issue #15934) Reported by: + viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by + tilghman (license 14) Tested by: viniciusfontes ........ + +2009-09-21 16:59 +0000 [r219721] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 219720 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 + Sep 2009) | 3 lines Reverting merge 219520. This change was not + necessary. ........ + +2009-09-20 17:55 +0000 [r219654] Tilghman Lesher + + * /, main/file.c: Merged revisions 219653 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) + | 8 lines Really stop the stream, when ast_closestream() is + called. (closes issue #15129) Reported by: bmh Patches: + 20090918__issue15129.diff.txt uploaded by tilghman (license 14) + Review: https://reviewboard.asterisk.org/r/372/ ........ + +2009-09-19 02:59 +0000 [r219587] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 219586 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 + Sep 2009) | 6 lines Make sure the iax_pvt exists before + dereferencing it. This fixes the latest crash posted on issue + 15609. (issue #15609) ........ + +2009-09-18 23:20 +0000 [r219451-219520] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 219519 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 + Sep 2009) | 9 lines iax2 frame double free The iax frame's + retrans sched id was written over right before iax2_frame_free + was called. In iax2_frame_free that retrans id is used to delete + the sched item. By writing over the retrans field before the + sched item could be deleted, it was possible for a retransmit to + occur on a freed frame. ........ + + * /, channels/chan_sip.c: Merged revisions 219450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) + | 14 lines via-header branches not updated correctly on INVITE + INVITE requests must always contain a new unique branch id. When + a new branch id is created for an INVITE, the dialog's + invite_branch variable must be updated so CANCEL requests use the + correct branch id. (closes issue #15262) Reported by: maniax + Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety + (license 608) invite_new_branch_trunk.diff uploaded by dvossel + (license 671) Tested by: maniax, dvossel ........ + +2009-09-18 13:54 +0000 [r219412] Tilghman Lesher + + * apps/app_voicemail.c: Missing value setting line for + maxsecs/maxmessage (closes issue #15696) Reported by: + fhackenberger Patches: maxsecs.patch uploaded by fhackenberger + (license 592) + +2009-09-17 22:37 +0000 [r219371] David Vossel + + * channels/chan_sip.c: fixes deadlock when performing directed + pickup w Invite/replaces (closes issue #15340) Reported by: + lmsteffan Patches: deadlock.patch uploaded by lmsteffan (license + 779) Tested by: lmsteffan + +2009-09-17 22:22 +0000 [r219324] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 219320 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep + 2009) | 6 lines Send a 100 Trying response when we detect a + spiral. This was problematic during spiral tests at SIPit... + along with some other things as well. ........ + +2009-09-17 21:59 +0000 [r219304] David Vossel + + * /, channels/chan_sip.c: Merged revisions 219303 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) + | 21 lines INVITE w/Replaces deadlock fix This patch cleans up + the locking logic in chan_sip.c's handle_invite_replaces() + function as well as making use of ast_do_masquerade() rather than + forcing the masquerade on an ast_read(). The code had several + redundant unlocks that would result in 'freed more times than + we've locked!' errors. I cleaned these up as well as moving all + the unlock logic to the end of the function. This patch should + also resolve the issue people were having with the replacecall + channel never being unlocked with one legged calls. (closes issue + #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff + uploaded by dvossel (license 671) Tested by: irroot, dvossel + Review: https://reviewboard.asterisk.org/r/371/ ........ + +2009-09-17 19:57 +0000 [r219264] Joshua Colp + + * channels/chan_sip.c: Ensure no spaces exist before "refresher=" + when doing the comparison. + +2009-09-17 16:25 +0000 [r219230] Sean Bright + + * apps/app_chanspy.c: Get this compiling under dev-mode. + +2009-09-17 15:18 +0000 [r219139] Matthew Nicholson + + * main/channel.c, /, include/asterisk/cdr.h: Merged revisions + 219136 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep + 2009) | 10 lines Prevent a potential race condition and crash + when hanging up a channel by removing the channel from the + channel list before begining channel tear down. This fix may + potentially cause problems with CDR backends that access the + channel a CDR is associated with via the channel list. This fix + makes the channel unavabile at the time when the CDR backend is + invoked. This has been documented in include/asterisk/cdr.h. + (closes issue #15316) Reported by: vmarrone Tested by: mnicholson + Review: https://reviewboard.asterisk.org/r/362/ ........ + +2009-09-17 00:58 +0000 [r219007-219105] Tilghman Lesher + + * CHANGES, apps/app_chanspy.c: Add the 'E' option to exit ChanSpy, + once the single channel it spied upon hangs up. In addition, + there's a bit of cleanup to the arguments and documentation, in + which I discovered that the last feature added to this + application duplicated an option (oops!) and changed that option + so that it now works. (closes issue #14909) Reported by: junky + Patches: __20090901-spy_hangup_trunk.diff uploaded by lmadsen + (license 10) Tested by: amilcar, junky, flujan, lmadsen + + * /, main/config.c, configs/extensions.conf.sample: Merged + revisions 219023 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) + | 8 lines Properly deal with quotes in the arguments of '#exec' + includes. (closes issue #15583) Reported by: pkempgen Patches: + 20090726__issue15583.diff.txt uploaded by tilghman (license 14) + 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license + 169) Tested by: pkempgen ........ + + * configure, include/asterisk/autoconfig.h.in, configure.ac: Detect + whether we actually have the long double type, before looking for + those functions. (closes issue #15017) Reported by: tzafrir + Patches: 20090916__issue15017.diff.txt uploaded by tilghman + (license 14) Tested by: tzafrir + +2009-09-16 20:32 +0000 [r218973] Sean Bright + + * res/res_jabber.c: Remove some unused defines from res_jabber. + (closes issue #15359) Reported by: snuffy Patches: + bug_res_jabber_unused_defines.diff uploaded by snuffy (license + 35) + +2009-09-16 19:25 +0000 [r218933] Mark Michelson + + * channels/chan_sip.c: Reverse order of args to fread. This way, we + don't always write a null byte into byte 1 of the buffer (closes + issue #15905) Reported by: ebroad Patches: freadfix.patch + uploaded by ebroad (license 878) Tested by: ebroad + +2009-09-16 18:31 +0000 [r218918] Joshua Colp + + * channels/chan_sip.c: On TCP and TLS connections do not attempt to + stop retransmission of the packet internally. This was preventing + responses from being properly processed because the packet was + not being found causing handle_response to return prematurely. + +2009-09-16 18:06 +0000 [r218868] David Brooks + + * main/pbx.c, /: Merged revisions 218867 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) + | 13 lines Fixes CID pattern matching behavior to mirror that of + extension pattern matching. Pattern matching for extensions uses + a type of scoring system, giving values for specificity to each + character in the pattern. Unfortunately, this is done character + by character, in order. This does lead to some less specific + patterns being first in line for matching, but it will usually + get the job done. This patch merely brings CID matching to the + same level as extension matching. This patch does not attempt to + tackle the problem shared by extension matching. (closes issue + #14708) Reported by: klaus3000 ........ + +2009-09-16 13:34 +0000 [r218799] Russell Bryant + + * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged + revisions 218798 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) + | 9 lines Remove the IAXy firmware from Asterisk. The firmware + can now be found on downloads.digium.com, where the rest of our + binary downloads live. This was the last part of our Asterisk + tarballs that was considered non-free by Debian. :-) (closes + issue #15838) Reported by: paravoid ........ + +2009-09-15 22:33 +0000 [r218731] Tilghman Lesher + + * /, apps/app_voicemail.c: Merged revisions 218730 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 + Sep 2009) | 6 lines If the user enters the same password as + before, don't signal an error when the change does nothing. + (closes issue #15492) Reported by: cbbs70a Patches: + 20090713__issue15492.diff.txt uploaded by tilghman (license 14) + ........ + +2009-09-15 19:22 +0000 [r218687] David Vossel + + * channels/chan_sip.c: upward bound checking for port string to int + conversion + +2009-09-15 16:15 +0000 [r218586] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 218578 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep + 2009) | 8 lines Send request contact header field with response + to registrer queries instead of the address of record. (closes + issue #14438) Reported by: ravindrad Patches: regquerypatch + uploaded by ravindrad (license 684) Tested by: ravindrad ........ + +2009-09-15 16:12 +0000 [r218583] Jeff Peeler + + * channels/chan_dahdi.c: Add some changes related to 218430. * + Remove thread_spawned in handle_init_event since it was never + used * Always check handle_init_event in case a channel is + destroyed + +2009-09-15 16:04 +0000 [r218579] Tilghman Lesher + + * /, apps/app_followme.c: Merged revisions 218577 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) + | 9 lines Ensure FollowMe sets language in channels it creates. + Also, not in the original bug report, but related fields are + accountcode and musicclass, and the inheritance of datastores. + (closes issue #15372) Reported by: Romik Patches: + 20090828__issue15372.diff.txt uploaded by tilghman (license 14) + Tested by: cervajs ........ + +2009-09-15 15:40 +0000 [r218504-218566] Mark Michelson + + * channels/chan_sip.c: Use a better method of ensuring + null-termination of the buffer while reading the SDP when using + TCP. + + * channels/chan_sip.c: Ensure that SDP read from TCP socket is + null-terminated. + +2009-09-15 15:02 +0000 [r218500] Kevin P. Fleming + + * /: Merged revisions 218497 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep + 2009) | 1 line Use proper hostname for downloading sound files. + ........ + +2009-09-15 14:59 +0000 [r218499] Mark Michelson + + * channels/chan_sip.c: Fix off-by-one error when reading SDP sent + over TCP. + +2009-09-15 10:24 +0000 [r218465] Tzafrir Cohen + + * channels/chan_dahdi.c: Fix false error message on + DAHDI_EVENT_REMOVED (RESULT_SUCCESS == 0) + +2009-09-14 22:38 +0000 [r218430] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Merged revisions 218401 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) + | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent + crash in do_monitor. After talking to rmudgett about some of his + recent iflist locking changes, it was determined that the only + place that would destroy a channel without being explicitly to do + so was in handle_init_event. The loop to walk the interface list + has been modified to wait to destroy the channel until the + dahdi_pvt of the channel to be destroyed is no longer needed. + (closes issue #15378) Reported by: samy ........ + +2009-09-14 20:08 +0000 [r218365] Richard Mudgett + + * channels/chan_dahdi.c: Add support for multiple interface lists. + Also unlink the sig_pri_pri.pvts[] pointer in + destroy_dahdi_pvt(). + +2009-09-14 19:29 +0000 [r218361] Tilghman Lesher + + * /, configs/voicemail.conf.sample, sounds/Makefile, + apps/app_voicemail.c: Recorded merge of revisions 218331 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) + | 4 lines Don't say "Please try again" if we don't give the user + another chance to try again. (issue #15055, SWP-129) Reported by: + jthurman ........ + +2009-09-14 18:16 +0000 [r218295] Joshua Colp + + * main/features.c: Do not attempt to add a parking extension if an + error occurred while reading the configuration. + +2009-09-14 14:57 +0000 [r218224] Matthew Nicholson + + * /, apps/app_directed_pickup.c: Merged revisions 218223 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep + 2009) | 8 lines Ensure we don't pickup ourselves when doing + pickup by exten. (closes issue #15100) Reported by: lmsteffan + Patches: (modified) pickup.patch uploaded by lmsteffan (license + 779) ........ + +2009-09-13 17:34 +0000 [r218184] Tzafrir Cohen + + * channels/chan_phone.c: gcc 4.4: Remove a nop memset size 0 that + annoys gcc This memset doesn't write beyond the end of the + buffer. (tmpbuf has size of 4). + +2009-09-13 05:51 +0000 [r218150] Moises Silva + + * channels/chan_dahdi.c: get rid of mfcr2 monitor thread condition, + is problematic + +2009-09-12 13:08 +0000 [r218107] Michiel van Baak + + * res/res_rtp_asterisk.c: use the actual given ip address for 'rtp + set debug ip ' instead of the word 'ip' (closes issue + #15711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt + uploaded by mvanbaak (license 7) Tested by: davidw + +2009-09-11 05:58 +0000 [r217990-218050] Tilghman Lesher + + * main/pbx.c: Check the origination priority for more matches, not + the current priority. Found by Pavel Troller on the -dev list. + + * /, apps/app_queue.c: Merged revisions 217989 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) + | 3 lines Don't ring another channel, if there's not enough time + for a queue member to answer. (Fixes AST-228) ........ + +2009-09-10 23:49 +0000 [r217954-217987] Jeff Peeler + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Cleanup approach in 217804 and don't reach inside the sig_pvt. + + * channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h: Allow do not disturb to be set on analog + channels via the CLI and AMI. + +2009-09-10 23:12 +0000 [r217916] Tilghman Lesher + + * contrib/scripts/iax-friends.sql, channels/chan_sip.c, + channels/chan_iax2.c: Make calltoken support work with realtime + users and peers. In the course of this, I also found that the + results of ast_gethostbyname were being used incorrectly in both + chan_iax2 and chan_sip, so both have been fixed. + +2009-09-10 22:31 +0000 [r217873-217912] Richard Mudgett + + * channels/chan_dahdi.c: Cleaned up chan_dahdi iflist handling and + locking. * Fixed walking the iflist so it is always done with the + iflock locked. * Simplified iflist walking routines. * Created + chan_dahdi iflist insertion and extraction routines. * Fixed + duplicate_pseudo() malloc fail handling. * Fixed infinite loop in + action_dahdishowchannels() when showing a single channel. + + * channels/chan_dahdi.c: Miscellaneous minor changes. + +2009-09-10 21:07 +0000 [r217807] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 217806 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 + Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call + Token security patch inadvertently broke the use of encryption + due to the reorganization of code in the socket_process() + function. When encryption is used, an incoming full frame must + first be decrypted before the information elements can be parsed. + The security release mistakenly moved IE parsing before + decryption in order to process the new Call Token IE. To resolve + this, decryption of full frames is once again done before looking + into the frame. This involves searching for an existing callno, + checking the pvt to see if encryption is turned on, and + decrypting the packet before the internal fields of the full + frame are accessed. (closes issue #15834) Reported by: karesmakro + Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel + (license 671) Tested by: dvossel, karesmakro Review: + https://reviewboard.asterisk.org/r/355/ ........ + +2009-09-10 20:52 +0000 [r217744-217804] Jeff Peeler + + * channels/chan_dahdi.c: Fix crash during attended transfer over + PRI. The owner pointers in the sig_pri_chan structure were not + getting updated in dahdi_fixup. + + * channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h: Stop caller id transmission when offhook + event detected. This fixes the problem that would occur if an + analog phone was picked up while the caller id was being sent. + The caller id before sent the whole spill even after pickup and + is now corrected. + +2009-09-10 19:39 +0000 [r217730] Matthias Nick + + * res/res_musiconhold.c: Sets the correct musicclass after an + announcement (closes issue #15279) Reported by: mbeckwell + Patches: patch.txt uploaded by mnick (license ) Tested by: mnick + (closes issue #15832) Reported by: mbeckwell Patches: patch.txt + uploaded by mnick (license 874) Tested by: mnick + +2009-09-10 18:29 +0000 [r217663] Olle Johansson + + * channels/chan_sip.c: Don't assign UINT_MAX to an INT. + +2009-09-10 18:17 +0000 [r217638] Tilghman Lesher + + * res/res_config_odbc.c, configure, + include/asterisk/autoconfig.h.in, configure.ac: Verify support + for wide ODBC character types before using them. (closes issue + #15870) Reported by: nic_bellamy + +2009-09-10 12:06 +0000 [r217593] Olle Johansson + + * channels/chan_sip.c: Include ActionID in all events that are + responsed to AMI Action SIPShowRegistry (closes issue #15868) + Reported by: nic_bellamy Patches: + manager_SIPshowregistry_actionid.patch uploaded by nic bellamy + (license 299) + +2009-09-10 00:35 +0000 [r217560] Richard Mudgett + + * channels/chan_dahdi.c: Fix available() for SS7, MFC/R2, and + pseudo channels. + +2009-09-09 21:48 +0000 [r217524] Moises Silva + + * channels/chan_dahdi.c: ast_log replaced for ast_verbose in MFCR2 + event notifications + +2009-09-09 20:09 +0000 [r217482] Olle Johansson + + * channels/chan_sip.c: Don't report transfer success until we + actually know. 1xx messages are not final. Related to #12713 + Patch by oej A big thank you to file for finally fixing the + transfer() dialplan application. I've been waiting for years for + this. Great work! + +2009-09-09 18:52 +0000 [r217445] Tzafrir Cohen + + * res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc 4.4 + has more strict rules for aliasing. It doesn't like a struct + sockaddr_in pointer pointing to a struct sockaddr. So we make it + a union. + +2009-09-09 12:11 +0000 [r217408] Sean Bright + + * main/manager.c: Properly terminate the response to the manager + Ping action. In passing, correct the formatting of the Timestamp + attribute so that there is a space after the colon and before the + value. (closes issue #15861) Reported by: Ivan + +2009-09-09 10:39 +0000 [r217367-217368] Olle Johansson + + * channels/chan_sip.c: Not having any TLS session to write to is a + serious XMIT_ERROR. + + * channels/chan_sip.c: Formatting and doxygen updates + +2009-09-08 23:37 +0000 [r217331-217332] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h, channels/sig_pri.c: Fix memory leak of + sig_xxx private structures. + + * channels/chan_dahdi.c: Miscellaneous minor code cleanup in + mkintf(). + +2009-09-08 22:17 +0000 [r217286] Sean Bright + + * apps/app_meetme.c: Fix compilation of app_meetme. Reported by + ebroad in #asterisk-bugs + +2009-09-08 21:17 +0000 [r217236] Richard Mudgett + + * channels/sig_pri.c: Remove duplicate entry in the sig_pri_pri + private pointer array. + +2009-09-08 20:28 +0000 [r217199] Tilghman Lesher + + * /, apps/app_meetme.c: Merged revisions 217156 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) + | 7 lines When MOH is playing on the channel, announcements sent + through the conference are not heard. (closes issue #14588) + Reported by: voipas Patches: 20090716__issue14588__2.diff.txt + uploaded by tilghman (license 14) Tested by: lmadsen, twisted, + tilghman ........ + +2009-09-08 20:06 +0000 [r217158] Mark Michelson + + * include/asterisk/event.h: Add doxygen to ast_event_subscribe for + the description. Most importantly, note that a NULL description + will cause a crash, as I just experienced that firsthand. + +2009-09-08 18:06 +0000 [r217113] Russell Bryant + + * addons/format_mp3.c: Fix audio problems with format_mp3. This + problem was introduced when the AST_FRIENDLY_OFFSET patch was + merged. I'm surprised that nobody noticed any trouble when + testing that patch, but this fixes the code that fills in the + buffer to start filling in after the offset portion of the + buffer. (closes issue #15850) Reported by: 99gixxer Patches: + issue15850.diff1.txt uploaded by russell (license 2) Tested by: + 99gixxer + +2009-09-08 16:37 +0000 [r217074] Kevin P. Fleming + + * configure, include/asterisk/autoconfig.h.in, configure.ac: Ensure + that the default autoconf CFLAGS are not used. A recent change to + the configure script that allows the user to specify CFLAGS + and/or LDFLAGS to the script had the unfortunate side effect of + letting autoconf's default CFLAGS (-g -O2) feed in to the rest of + the build system, thereby overriding the DONT_OPTIMIZE setting in + menuselect. That problem is now corrected. + +2009-09-08 15:30 +0000 [r217033] Tilghman Lesher + + * res/res_limit.c: Remove what appears to be an unnecessary define. + (closes issue #15851) Reported by: tzafrir + +2009-09-08 15:23 +0000 [r217015] Tzafrir Cohen + + * contrib/scripts/live_ast: live_ast: Fix asterisk.conf instead of + regenerating it * Don't write asterisk.conf from scratch. Fix the + existing one. * Pass extra 'make' command-line arguments to + 'install' and 'samples'. * Fix some extra typos. closes issue + #15019 . + +2009-09-08 14:26 +0000 [r216993] David Vossel + + * channels/chan_sip.c: caller id number empty parse_uri was not + being given the correct scheme's, as a result, uri parsing did + not parse the username correctly. One of the side effects of this + is an empty caller id. (closes issue #15839) Reported by: ebroad + Patches: blank_cidv2.patch uploaded by ebroad (license 878) + parse_uri_fix.diff uploaded by dvossel (license 671) Tested by: + ebroad, dvossel + +2009-09-07 20:23 +0000 [r216883-216956] Olle Johansson + + * doc/manager_1_1.txt: Fixing formatting + + * doc/manager_1_1.txt: Add new actions under "new actions" and not + in the top of the document + + * channels/chan_sip.c: Moving another function declared in the + middle of forward declarations. Please follow the structure of + the source code, thanks. Chan_sip is messy enough as it is :-) + + * channels/chan_sip.c: Move "deprecated_username" to a flag like + the others - unsigned int blah:1 + + * channels/chan_sip.c: - Doxygen additions - Remove unused string + in sip_registry -- "random" - Someone added a function in the + middle of all forward declarations... Weird. Moved it out of that + section. + + * channels/chan_sip.c: Clean up the "offered_media" code - Add + variable for number of known media streams instead of hardcoding + in definition of sip_pvt - Rename "text" to "codecs" - beacuse + it's what it is - Add documentation for future developers so that + we make sure that we define new sdp media types for SRTP-variants + +2009-09-07 17:15 +0000 [r216846] Tilghman Lesher + + * configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: Allow + multiple rows to be fetched within the normal mode of operation. + +2009-09-07 16:35 +0000 [r216652-216842] Olle Johansson + + * channels/chan_sip.c: Make sure we reset global_exclude_static at + channel reload + + * channels/chan_sip.c: Move capability into sip_cfg. While at it, + make sure we reset it at channel reload. + + * channels/chan_sip.c: Move global_regcontext into the sip_cfg + structure + + * channels/chan_sip.c: Move contact_ha to sip_cfg structure + + * channels/chan_sip.c: Doxygen updates + + * channels/chan_sip.c: Since it's possible to have more than 999 + calls, I'm changing the call counter roof to something higher. + + * channels/chan_sip.c: add doxygen and remove duplicate declaration + of variable + + * channels/chan_sip.c: After many years, remove VOCAL_DATA_HACK + definition + + * channels/chan_sip.c: Remove unneeded header files (tested on + Linux and OS/X) + + * channels/chan_sip.c: Don't send MESSAGE with sendtext() if + recepient doesn't allow MESSAGE requests + + * channels/chan_sip.c: Add some doxygen + + * channels/chan_sip.c: Fix typo + + * channels/chan_sip.c: If there is no session timer in the INVITE, + set it to default value (not unset minimum = -1) Patch by oej + closes issue #15621 Reported by: fnordian Tested by: atis + + * configs/sip.conf.sample: Update sip.conf.sample documentation, + reorganize a bit + + * channels/chan_sip.c: Simplify the code in this function + +2009-09-04 19:32 +0000 [r216594] David Vossel + + * channels/chan_sip.c: sip peer matching by address only with + TCP/TLS This patch removes the contact header matching logic and + adds logic to match all tcp/tls connections by ip only. Thanks to + oej for finding the issue and suggesting solutions. Review: + https://reviewboard.asterisk.org/r/354/ + +2009-09-04 19:29 +0000 [r216593] Sean Bright + + * apps/app_voicemail.c: Use ast_free() instead of free(). + +2009-09-04 17:50 +0000 [r216547-216551] Tilghman Lesher + + * include/asterisk/lock.h: Fix trunk breakage. + + * main/pbx.c, UPGRADE-1.6.txt: Enable turning off the application + delimiter warning with the 'dontwarn' option. Suggested on the + -dev list, and implemented in an alternate way by me. + +2009-09-04 15:05 +0000 [r216506] Michiel van Baak + + * /, main/utils.c: Merged revisions 216435 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) + | 2 lines make asterisk compile under devmode with DEBUG_THREADS + enabled on OpenBSD ........ + +2009-09-04 14:02 +0000 [r216438] Olle Johansson + + * main/pbx.c, /, channels/chan_sip.c, apps/app_disa.c, + configs/sip.conf.sample, apps/app_playback.c: Merged revisions + 216430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 + lines Make apps send PROGRESS control frame for early media and + fix too early media issue in SIP The issue at hand is that some + legacy (dying) PBX systems send empty media frames on PRI links + *before* any call progress. The SIP channel receives these frames + and by default signals 183 Session progress and starts sending + media. This will cause phones to play silence and ignore the + later 180 ringing message. A bad user experience. The fix is + twofold: - We discovered that asterisk apps that support early + media ("noanswer") did not send any PROGRESS frame to indicate + early media. Fixed. - We introduce a setting in chan_sip so that + users can disable any relay of media frames before the outbound + channel actually indicates any sort of call progress. In 1.4, + 1.6.0 and 1.6.1, this will be disabled for backward + compatibility. In later versions of Asterisk, this will be + enabled. We don't assume that it will change your Asterisk phone + experience - only for the better. We encourage third-party + application developers to make sure that if they have + applications that wants to send early media, add a PROGRESS + control frame transmission to make sure that all channel drivers + actually will start sending early media. This has not been the + default in Asterisk previous to this patch, so if you got + inspiration from our code, you need to update accordingly. Sorry + for the trouble and thanks for your support. This code has been + running for a few months in a large scale installation (over 250 + servers with PRI and/or BRI links to old PBX systems). That's no + proof that this is an excellent patch, but, well, it's tested :-) + ........ + +2009-09-04 14:00 +0000 [r216431-216437] Michiel van Baak + + * include/asterisk/lock.h: make sure canlog is set so we can + compile with DEBUG_THREADS enabled on OpenBSD + + * /: Recorded merge of revisions 216432 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r216432 | mvanbaak | 2009-09-04 15:53:09 +0200 (Fri, 04 Sep 2009) + | 2 lines make chan_sip compile under devmode again ........ + + * /: Recorded merge of revisions 216369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r216369 | mvanbaak | 2009-09-04 15:16:29 +0200 (Fri, 04 Sep 2009) + | 4 lines Make sure 'start' is always initialized. This is the + same as rev 216222 in trunk but 1.4 is affected as well ........ + +2009-09-04 13:14 +0000 [r216368] Russell Bryant + + * channels/chan_sip.c: Do not treat every SIP peer as if they were + configured with insecure=port. There was a problem in the + function responsible for doing peer matching by IP address and + port number such that during the second pass for checking for a + peer configured with insecure=port, it would end up treating + every peer as if it had been configured that way. These changes + fix the logic in the peer IP and port comparison callback to + handle insecure=port checking properly. This problem was + introduced when SIP peers were converted to astobj2. Many thanks + to dvossel for noticing this while working on another peer + matching issue. + +2009-09-04 12:05 +0000 [r216335] Olle Johansson + + * doc/janitor-projects.txt: Adding to the janitor list. For new + readers: The janitor list is a list of tasks we need help with in + the Asterisk project. Taking up one of these is often a good way + to get into Asterisk development and getting a lot of developers + in the project to be grateful. It's stuff we could spend time on + when the bug tracker is empty, when our employers hasn't filled + our task lists and our servers is running bugfree and happily + without any issues. If you want to start working on one of these + small projects, feel free to ask for help in the #asterisk-dev + channel on IRC or asterisk-dev mailing list. We'll be more than + happy to help you to start and reach goal. Thank you for your + help. + +2009-09-04 10:48 +0000 [r216264] Russell Bryant + + * /, doc/IAX2-security.txt (added): Merged revisions 216263 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r216263 | russell | 2009-09-04 05:48:00 -0500 + (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 + Sep 2009) | 2 lines Add a plain text version of the IAX2 security + document. ........ ................ + +2009-09-04 06:08 +0000 [r216222] Michiel van Baak + + * main/astobj2.c: make sure 'start' is always initialized. Makes + asterisk compile with --enable-dev-mode + +2009-09-03 21:09 +0000 [r216186] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_pri.c: Lets try not to use + C++ keywords for variable names. + +2009-09-03 19:40 +0000 [r216094] Doug Bailey + + * include/asterisk/callerid.h, channels/chan_dahdi.c, + channels/sig_analog.c, channels/sig_analog.h: Added detection + DTMF CID without polarity change alert. Added detection of DTMF + tone energy levels on FXO channels in chan_dahdi monitoring loop + so DTMF CID can be detected without the need of a polarity change + precursor. (closes issue #9096) Reported by: fleed Patches: + 9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819) + Tested by: cyberplant, sum, maturs + +2009-09-03 19:38 +0000 [r216009-216092] Russell Bryant + + * /, UPGRADE.txt: Merged revisions 216085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r216085 | russell | 2009-09-03 14:36:46 -0500 + (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 + Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt. + ........ ................ + + * /, doc/IAX2-security.pdf (added): Merged revisions 216008 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r216008 | russell | 2009-09-03 13:44:58 -0500 + (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 + Sep 2009) | 2 lines Add IAX2 security document related to + AST-2009-006. ........ ................ + +2009-09-03 18:42 +0000 [r216006] Kevin P. Fleming + + * main/file.c, doc/lang/language-criteria.txt (added): Document + language prompt submission process. This patch adds a document + describing the language prompt submission process, licensing + terms and other issues related to that process. In addition, it + modifies the sound file searching process to support language + codes with any number of suffices (not limited to just "xx" or + "xx_YY"), so that prompts can be named with gender, + customer/company, etc. suffices as well. (closes issue #15771) + Reported by: jtodd Patches: language-criteria.txt uploaded by + jtodd + +2009-09-03 16:31 +0000 [r215955] David Vossel + + * configs/iax.conf.sample, include/asterisk/acl.h, + channels/iax2-parser.h, include/asterisk/astobj2.h, + channels/iax2.h, main/acl.c, channels/chan_iax2.c, + channels/iax2-parser.c, main/astobj2.c: Merge code associated + with AST-2009-006 (closes issue #12912) Reported by: rathaus + Tested by: tilghman, russell, dvossel, dbrooks + +2009-09-03 13:02 +0000 [r215891] Olle Johansson + + * channels/chan_sip.c: Add known internal IP address when + autodomain=yes (closes issue #14573) Reported by: pj Patches: + sip-internip-autodomain1.diff uploaded by mnicholson (license 96) + modified by oej Tested by: pj + +2009-09-03 05:57 +0000 [r215838] Michiel van Baak + + * doc/manager_1_1.txt: Document that SIPshowpeer and SKINNYshowline + now include the configured parkinglot in their response. Prodded + by snuff-work on #asterisk-dev IRC channel + +2009-09-03 03:43 +0000 [r215800-215801] Tilghman Lesher + + * channels/chan_sip.c: Default the callback extension to "s". This + is a regression. (closes issue #15764) Reported by: elguero + Change-type: bugfix + + * include/asterisk.h: Revert attempt to standardize with + _POSIX_C_SOURCE. This did not function in the way that was + intended, causing more compatibility issues than it solved. It is + best, therefore, that it be simply removed. (Discussed with + kpfleming; agreement to remove was reached.) + +2009-09-02 23:31 +0000 [r215758] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 215682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) + | 18 lines Re-send non-100 provisional responses to prevent + cancellation From section 13.3.1.1 of RFC 3261: If the UAS + desires an extended period of time to answer the INVITE, it will + need to ask for an "extension" in order to prevent proxies from + canceling the transaction. A proxy has the option of canceling a + transaction when there is a gap of 3 minutes between responses in + a transaction. To prevent cancellation, the UAS MUST send a + non-100 provisional response at every minute, to handle the + possibility of lost provisional responses. (closes issue #11157) + Reported by: rjain Tested by: twilson Review: + https://reviewboard.asterisk.org/r/315/ ........ + +2009-09-02 23:25 +0000 [r215757] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, CHANGES, channels/sig_pri.c: Made + chan_dahdi able to ignore incoming calls that are not in a MSN + list for ISDN PTMP CPE spans. + +2009-09-02 21:39 +0000 [r215681] David Vossel + + * channels/chan_sip.c: port string to int conversion using sscanf + There are several instances where a port is parsed from a uri or + some other source and converted to an int value using atoi(), if + for some reason the port string is empty, then a standard port is + used. This logic is used over and over, so I created a function + to handle it in a safer way using sscanf(). + +2009-09-02 21:23 +0000 [r215622-215665] Michiel van Baak + + * channels/chan_sip.c, channels/chan_skinny.c: add Parkinglot info + to sip show peer and skinny show line If we had this + from the start, debugging the 'parking not using configured + parkinglot' bug would have been easier. + + * main/features.c: - lock channel before looking for a channel + variable - Init the parkings list member of struct parkinglot. + Thanks Sean for the explanation why this should be here. + +2009-09-02 19:49 +0000 [r215608] Doug Bailey + + * channels/chan_dahdi.c, channels/sig_analog.c: Fix issue where + DTMF CID detect was placing channels into signed linear mode made + analog_set_linear_mode return back the mode that was being + overwritten so it could be restored later. + +2009-09-02 18:37 +0000 [r215567] Tilghman Lesher + + * main/Makefile, main/app.c: Close up to the soft open file limit + (same on Linux, but varies drastically on OS X). Also, a Makefile + fix for Darwin (OS X). (closes issue #14542) Reported by: jtodd + Patches: 20090901__issue14542.diff.txt uploaded by tilghman + (license 14) Tested by: jtodd, tilghman Change-type: bugfix + +2009-09-02 17:26 +0000 [r215522] David Vossel + + * channels/chan_sip.c: SIP uri parsing cleanup Now, the scheme + passed to parse_uri can either be a single scheme, or a list of + schemes ',' delimited. This gets rid of the whole problem of + having to create two buffers and calling parse_uri twice to check + for separate schemes. Review: + https://reviewboard.asterisk.org/r/343/ + +2009-09-02 16:20 +0000 [r215479] Michiel van Baak + + * channels/chan_skinny.c: like in chan_sip's sip_new skinny should + copy the configured parkinglot from a line to the newly created + channel. This makes callparking honor the configured parkinglot + for skinny lines as well. + +2009-09-02 16:08 +0000 [r215466] David Vossel + + * channels/chan_sip.c: SIP support for keep-alive event keep-alive + events are used by Sipura/Linksys for NAT keepalive. There + currently don't appear to be any problems with NAT, but everytime + a keep-alive event is received, Asterisk responds with a "489 Bad + event". This error may indicate to a user that NAT problems exist + just because this even is not supported. Now, rather than respond + with an error, the packet is consumed and a "200 ok" is sent just + to indicate we received the packet. (issue #15084) Patches: + chan_sip.keepalive.v1.diff uploaded by IgorG (license 20) + +2009-09-02 15:56 +0000 [r215419-215462] Michiel van Baak + + * channels/chan_sip.c: Honor configured parkinglot when parking and + retrieving parked calls Thank oej for pointing out the fact that + sip_new did not copy parkinglot from the peer into the newly + created channel. (closes issue #15538) Reported by: gracedman + Patches: 2009090100_sipnewparkinglot-161.diff.txt uploaded by + mvanbaak (license 7) With mod by me to also fix callparking as + well (this uploaded patch only fixed retrieving a parked call) + Tested by: gracedman, mvanbaak + + * include/asterisk.h: Let's compile again on OpenBSD + +2009-09-02 06:23 +0000 [r215382] Olle Johansson + + * CHANGES, res/res_mutestream.c (added): Adding MUTEAUDIO() + dialplan function and MuteAudio AMI action (pinepeach) Review: + https://reviewboard.asterisk.org/r/345/ + +2009-09-02 01:16 +0000 [r215338] Dwayne M. Hubbard + + * /, apps/app_softhangup.c: Merged revisions 215270 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 + Sep 2009) | 12 lines Use strrchr() so SoftHangup will correctly + truncate multi-hyphen channel names In general channel names are + in the form Foo/Bar-Z, but the channel name could have multiple + hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the + channel name at the last hyphen. (closes issue #15810) Reported + by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by + dhubbard (license 733) ........ + +2009-09-01 23:41 +0000 [r215222-215301] Tilghman Lesher + + * channels/chan_sip.c, funcs/func_channel.c, CHANGES: Add + MASTER_CHANNEL() dialplan function, as well as a useful usage. + (closes issue #13140) Reported by: cpina Patches: + 20090807__issue13140.diff.txt uploaded by tilghman (license 14) + Tested by: lmadsen Change-type: feature + + * channels/chan_sip.c: Fix register such that lines with a + transport string, but without an authuser, parse correctly. + (AST-228) + +2009-09-01 20:44 +0000 [r215212] Russell Bryant + + * addons/format_mp3.c: Fix memory corruption caused by format_mp3. + format_mp3 claimed that it provided AST_FRIENDLY_OFFSET in frames + returned by read(). However, it lied. This means that other parts + of the code that attempted to make use of the offset buffer would + end up corrupting the fields in the ast_filestream structure. + This resulted in quite a few crashes due to unexpected values for + fields in ast_filestream. This patch closes out quite a few bugs. + However, some of these bugs have been open for a while and have + been an area where more than one bug has been discussed. So with + that said, anyone that is following one of the issues closed + here, if you still have a problem, please open a new bug report + for the specific problem you are still having. If you do, please + ensure that the bug report is based on the newest version of + Asterisk, and that this patch is applied if format_mp3 is in use. + Thanks! (closes issue #15109) Reported by: jvandal Tested by: + aragon, russell, zerohalo, marhbere, rgj (closes issue #14958) + Reported by: aragon (closes issue #15123) Reported by: + axisinternet (closes issue #15041) Reported by: maxnuv (closes + issue #15396) Reported by: aragon (closes issue #15195) Reported + by: amorsen Tested by: amorsen (closes issue #15781) Reported by: + jensvb (closes issue #15735) Reported by: thom4fun (closes issue + #15460) Reported by: marhbere + +2009-09-01 19:50 +0000 [r215161] Kevin P. Fleming + + * main/frame.c: Ensure that frame dumps of + AST_CONTROL_T38_PARAMETERS frames are properly decoded. + +2009-09-01 14:40 +0000 [r215110] Olle Johansson + + * channels/chan_sip.c: Removing whitespace that causes red dots in + reviewboard + +2009-08-31 22:02 +0000 [r215069-215070] Tilghman Lesher + + * main/http.c: Fix a trunk compilation warning. + + * main/manager.c: Properly initialize the session to prevent a + crash. (closes issue #15774) Reported by: lasko Patches: + 20090831__issue15774.diff.txt uploaded by tilghman (license 14) + Tested by: lasko + +2009-08-31 18:17 +0000 [r215023] Olle Johansson + + * funcs/func_volume.c: By copying this code I got bad comments in + reviewboard... Better fix the original. + +2009-08-31 16:18 +0000 [r214819-214945] Tilghman Lesher + + * channels/chan_local.c, /: Merged revisions 214940 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 + Aug 2009) | 7 lines Also unlock the "other" channel, when + returning, due to glare. (closes issue #15787) Reported by: + tim_ringenbach Patches: chan_local.diff uploaded by tim + ringenbach (license 540) Tested by: tim_ringenbach ........ + + * Makefile: Force Darwin on ppc platforms to compile with a target + level that supports aliasing. + + * include/asterisk.h, main/poll.c: Various patches, to enable + Asterisk to once again compile on Mac OS X. One note on defining + _POSIX_C_SOURCE: while this feature test macro works to require + certain behaviors on Linux, it works differently on *BSD + platforms to REMOVE certain API calls that are not in the POSIX + specification, such as vasprintf(3). Thus, defining it while + depending upon vasprintf (and other extensions to the POSIX + standard) to be defined is a recipe to ensure that Asterisk is + only buildable on Linux. Hence, this define which was meant to + INCREASE portability, effectively ensures the opposite. + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + pbx/pbx_lua.c: If lua is detected with the lua5.1 prefix (or + not), adjust the include path accordingly. Based upon feedback to + a release announcement on the -users list. See + http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html + +2009-08-28 22:44 +0000 [r214777] Russell Bryant + + * configure: Update configure script so that CONFIG_CFLAGS and + CONFIG_LDFLAGS doesn't break the build. + +2009-08-28 20:14 +0000 [r214702] Tilghman Lesher + + * main/channel.c, /: Merged revisions 214701 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) + | 8 lines Modify comment to be a bit more accurate. We have kept + this comment around long enough, that it's pretty clear that + we're keeping the code, because changing the code would require a + pretty fundamental architectural shift. We've also taken + criticism in some quarters, because it was believed that it was + referring to the code being nasty. No, the code isn't nasty, just + the operation itself is rather odd. Fixed for eternity (probably + not). ........ + +2009-08-28 20:01 +0000 [r214696] Kevin P. Fleming + + * Makefile, include/asterisk/autoconfig.h.in, configure.ac, + makeopts.in: Ensure that CFLAGS and/or LDFLAGS provided to + configure script are preserved. Cross-compilation environments + want to provide 'defaults' for compiler and linker options, and + frequently do this by specifying CFLAGS and LDFLAGS in the + environment or as command-line arguments to the configure script. + This patch modifies the configure script and Makefile to preserve + these settings and ensure they are used in the build process. + +2009-08-28 19:13 +0000 [r214654] Richard Mudgett + + * channels/sig_pri.c: Move discardremoteholdretrieval test so it + applies only to the specific notification indicator values. + +2009-08-28 18:41 +0000 [r214650] Mark Michelson + + * include/asterisk/sched.h: Fix some incorrect documentation of + sched_thread functions. + +2009-08-28 16:50 +0000 [r214360-214611] Tilghman Lesher + + * res/res_musiconhold.c: Remove unnecessary define for Solaris + (closes issue #15358) Reported by: snuffy Patches: + bug_res_moh_remove_unneeded_include.diff uploaded by snuffy + (license 35) + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + autoconf/libcurl.m4 (added): Merged revisions 214517 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 + Aug 2009) | 7 lines Use autoconf to detect libcurl, as this + enables cross-compilation checks, something we didn't allow + before. (closes issue #15714) Reported by: pprindeville Patches: + 20090813__issue15714.diff.txt uploaded by tilghman (license 14) + Tested by: pprindeville ........ + + * main/manager.c: Ensure that we check for the special value + CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by: + a_villacis Patches: + asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch + uploaded by a villacis (license 660) (Plus a few of my own, to + catch the remaining places within manager.c where it could have + been a problem) + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + autoconf/ast_ext_lib.m4: Merged revisions 214436 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 + Aug 2009) | 2 lines One more build system change, to make the + descriptions look better, if we have better information. ........ + + * /, configure, include/asterisk/autoconfig.h.in, + autoconf/ast_ext_lib.m4: Merged revisions 214357 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 + Aug 2009) | 3 lines Make autoheader descriptions render correctly + in our autoconfig.h file. (Figured out while working with issue + #14906) ........ + +2009-08-27 15:57 +0000 [r214309-214355] Jeff Peeler + + * doc/tex/channelvariables.tex: Add forgotten documentation for new + channel variables added in 214309. + + * main/features.c, CHANGES: Add two new dialplan variables when + using features Added DYNAMIC_FEATURENAME which holds the last + triggered dynamic feature. Added DYNAMIC_PEERNAME which holds the + unique channel name on the other side and is set when a dynamic + feature is triggered. (closes issue #14663) Reported by: tamiel + Patches: 20090313_features.diff uploaded by tamiel (license 712) + Tested by: tamiel + +2009-08-26 21:56 +0000 [r214272] Richard Mudgett + + * configs/chan_dahdi.conf.sample: Minor punctuation change. + +2009-08-26 16:53 +0000 [r214199] Tilghman Lesher + + * channels/chan_sip.c: Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) + (closes issue #15362) Reported by: klaus3000 Patches: + chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license + 65) + +2009-08-26 16:38 +0000 [r214195] David Vossel + + * main/channel.c, /: Merged revisions 214194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009) + | 19 lines ast_write() ignores ast_audiohook_write() results In + ast_write(), if a channel has a list of audiohooks, those lists + are written to and the resulting frame is what ast_write() should + continue with. The problem was the returned audiohook frame was + not being handled at all, and the original frame passed into it + did not contain the mixed audio, so essentially audio was being + lost. One result of this was chan_spy's whisper mode no longer + worked. To complicate the issue, frames passed into ast_write may + either be a single frame, or a list of frames. So, as the list of + frames is processed in the audiohook_write, the returned frames + had to be added to a new list. (closes issue #15660) Reported by: + corruptor Tested by: dvossel ........ + +2009-08-25 22:39 +0000 [r213900-214152] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac: Not + all versions of gnu-linux use glibc, which contains iconv. Some + (especially embedded systems) don't have iconv at all. (closes + issue #15169) Reported by: pprindeville + + * /, main/say.c: Merged revisions 214068-214069 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009) + | 6 lines Fix pronunciation of German dates. (closes issue + #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded + by Benjamin Kluck (license 803) ........ r214069 | tilghman | + 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should + always compile before committing... ........ + + * pbx/pbx_dundi.c: DUNDILOOKUP function in 1.6 should use comma + delimiters. (closes issue #15322) Reported by: chappell Patches: + dundilookup-0015322.patch uploaded by chappell (license 8) + + * main/pbx.c, /: Merged revisions 213970 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009) + | 7 lines Improve error message by informing user exactly which + function is missing a parethesis. (closes issue #15242) Reported + by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by + dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by + loloski (license 68) ........ + + * Makefile: The DTD should be installed in the same path as the + rest of the XML documentation. (closes issue #15344) Reported by: + tzafrir Patches: makefile_appdocs_dtd.diff uploaded by tzafrir + (license 46) + + * Makefile, /: Merged revisions 213899 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009) + | 4 lines Use the default runlevels for Debian derivatives, + instead of making up our own. (closes issue #14730) Reported by: + pkempgen ........ + +2009-08-24 16:43 +0000 [r213833] Jeff Peeler + + * apps/app_voicemail.c: Fix storage of greetings when using + IMAP_STORAGE The store macro was not getting called preventing + storage of IMAP greetings at all. This has been corrected along + with fixing checking if the imapgreetings option is turned on to + store the greeting in IMAP. Lastly, the attachment filename was + incorrectly using the full path instead of just the basename, + which was causing problems with retrieval of the greeting. + (closes issue #14950) Reported by: noahisaac (closes issue + #15729) Reported by: lmadsen + +2009-08-24 04:46 +0000 [r213790] Moises Silva + + * channels/chan_dahdi.c: improve handling of + openr2_chan_disconnect_call API failure, unlikely, but happened + on openr2 library bug + +2009-08-21 23:18 +0000 [r213748] Richard Mudgett + + * configure, configure.ac, channels/sig_pri.c: Update configure + script for libpri COLP feature dependency requirements. + +2009-08-21 22:36 +0000 [r213738] Tilghman Lesher + + * channels/chan_sip.c: Clarifying comments in sip_register, and + removing a dead section + +2009-08-21 22:22 +0000 [r213716] David Vossel + + * channels/chan_sip.c: Register request line contains wrong address + when user domain and register host differ (closes issue #15539) + Reported by: Nick_Lewis Patches: chan_sip.c-registraraddr.patch + uploaded by Nick (license 657) register_domain_fix_1.6.2 uploaded + by dvossel (license 671) Tested by: Nick_Lewis, dvossel + +2009-08-21 21:39 +0000 [r213697] Kevin P. Fleming + + * apps/app_voicemail.c: Ensure that realtime mailboxes properly + report status on subscription. This patch modifies + app_voicemail's response to mailbox status subscriptions (via the + internal event system) to ensure that a subscription triggers an + explicit poll of the mailbox, so the subscriber can get an + immediate cached event with that status. Previously, the cache + was only populated with the status of non-realtime mailboxes. + (closes issue #15717) Reported by: natmlt + +2009-08-21 21:02 +0000 [r213635] David Vossel + + * channels/chan_sip.c: fixes sip register parsing when user@domain + is used (issue #15008) (issue #15672) + +2009-08-21 16:53 +0000 [r213560] Tilghman Lesher + + * include/asterisk.h, /: Merged revisions 213559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009) + | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files. + (closes issue #15698) Reported by: slavon Patches: + 20090817__issue15698.diff.txt uploaded by tilghman (license 14) + Tested by: slavon, tilghman ........ + +2009-08-21 16:04 +0000 [r213494] Jason Parker + + * /, configs/queues.conf.sample: Merged revisions 213493 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) | + 5 lines Clarify queues.conf comments to specify that variables + should be set in the dialplan. (closes issue #15755) Reported by: + trendboy ........ + +2009-08-21 04:09 +0000 [r213454] Moises Silva + + * channels/chan_dahdi.c: increment the mfcr2 monitor count when + clearing the call request + +2009-08-21 03:48 +0000 [r213450] Terry Wilson + + * main/loader.c: Make LOAD_ORDER actually work + +2009-08-20 22:13 +0000 [r213414] Tilghman Lesher + + * apps/app_queue.c: Add original position, when logging a caller + entering a queue. (closes issue #15146) Reported by: arabe + Patches: asterisk-trunk.patch uploaded by arabe (license 786) + +2009-08-20 21:33 +0000 [r213404] Jeff Peeler + + * apps/app_voicemail.c: Fix greeting retrieval from IMAP Properly + check for the current voicemail state and if it doesn't exist, + create it. (closes issue #14597) Reported by: wtca Patches: + 14597_v2.patch uploaded by mmichelson (license 60) Tested by: + jpeeler + +2009-08-20 20:29 +0000 [r213327] Matthew Nicholson + + * main/features.c: Fix a crash by checking the proper pointer for + validity before deferencing it. (closes issue #15751) Reported + by: atis Patches: ast_bridge_call_peer_cdr.patch uploaded by atis + (license 242) + +2009-08-20 19:56 +0000 [r213284] Jeff Peeler + + * apps/app_voicemail.exports (added), /: Merged revisions 213283 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r213283 | jpeeler | 2009-08-20 14:53:34 -0500 (Thu, 20 Aug 2009) + | 2 lines Make all the symbols for the C-client callbacks global + ........ + +2009-08-20 15:29 +0000 [r213248] Tilghman Lesher + + * addons/res_config_mysql.c: Select uncommented lines, not + commented ones. (closes issue #15746) Reported by: makoto + +2009-08-20 03:26 +0000 [r213216] Moises Silva + + * channels/chan_dahdi.c: fixed bug caused by calling ast_request + without calling ast_call on an R2 channel, ie, CHANISAVAIL + +2009-08-19 22:38 +0000 [r213179] Jason Parker + + * main/ulaw.c, main/alaw.c: Fix compile when certain G711 + menuselect options are enabled. (closes issue #15697) Reported + by: slavon + +2009-08-19 21:21 +0000 [r213113] David Vossel + + * /, apps/app_mixmonitor.c: Merged revisions 213103 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 + Aug 2009) | 8 lines Fixes memory leak caused by incorrectly + freeing mixmonitor (closes issue #15699) Reported by: edantie + Patches: mixmonitor.patch uploaded by edantie (license 862) + ........ + +2009-08-19 21:05 +0000 [r213093-213098] Tilghman Lesher + + * channels/chan_sip.c, configs/sip.conf.sample: Better parsing for + the "register" line Allows characters that are otherwise used as + delimiters to be used within certain fields (like the secret). + (closes issue #15008, closes issue #15672) Reported by: tilghman + Patches: 20090818__issue15008.diff.txt uploaded by tilghman + (license 14) Tested by: lmadsen, tilghman + + * channels/chan_sip.c: If we have realtime caching enabled, 'sip + reload' must purge users/peers, even if the config files haven't + changed. (closes issue #12869) Reported by: bcnit Patches: + 20090819__issue12869__2.diff.txt uploaded by tilghman (license + 14) Tested by: lasko + +2009-08-19 15:32 +0000 [r213046] Russell Bryant + + * main/features.c: Don't blow up on a NULL cdr. Reported in + #asterisk-dev. + +2009-08-18 23:53 +0000 [r213007] Richard Mudgett + + * channels/sig_pri.h, CHANGES, channels/sig_pri.c: Add COLP support + to chan_dahdi/sig_pri. Add Connected Line Presentation (COLP) + support to chan_dahdi/libpri as an addition to issue 8824. This + is the chan_dahdi/sig_pri portion. COLP support is now available + for any switch for which libpri supports COLP (currently ETSI + PTP, ETSI PTMP, and Q.SIG) with this patch. (closes issue #14068) + Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/340/ + +2009-08-18 20:33 +0000 [r212922-212939] Kevin P. Fleming + + * /: Remove some accidentally-committed properties. + + * CREDITS, /, UPGRADE-1.4.txt, sounds/sounds.xml, + build_tools/prep_tarball, sounds/Makefile, doc/tex/asterisk.tex: + Convert this branch to Opsound music-on-hold. For more details: + http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ + +2009-08-18 19:49 +0000 [r212857-212883] Tilghman Lesher + + * addons/res_config_mysql.c: Clarify some of the error messages, to + help upgraders. + + * configs/extconfig.conf.sample: Make the default extconfig.conf + match entries with the sample res_mysql.conf. This eliminates a + future source of possible confusion with the configuration of + 1.6.1 and higher. + +2009-08-18 18:57 +0000 [r212844] Olle Johansson + + * apps/app_meetme.c: Small doxygen changes + +2009-08-18 16:38 +0000 [r212764] Sean Bright + + * main/manager.c, /: Merged revisions 212763 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug + 2009) | 11 lines Delay the creation of temporary files until we + have a valid manager command to handle. Without this patch, + asterisk creates a temporary file before determining if the + specified command is valid. If invalid, we weren't properly + cleaning up the file. (closes issue #15730) Reported by: zmehmood + Patches: M15730.diff uploaded by junky (license 177) Tested by: + zmehmood ........ + +2009-08-18 16:29 +0000 [r212758] Richard Mudgett + + * /, channels/misdn/isdn_lib.c: Merged revisions 212727 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 Aug 2009) + | 1 line Removed some deadwood and added some doxygen comments. + ........ + +2009-08-17 20:40 +0000 [r212672] Kevin P. Fleming + + * include/asterisk.h: Relax check for XOPEN_VERSION. It's not clear + that we actually require XOPEN_VERSION to be 600 or greater at + this time, so skip the check for now. + +2009-08-17 19:57 +0000 [r212627] Tilghman Lesher + + * apps/app_voicemail.c: Check the return value of opendir(3), or we + may crash. (closes issue #15720) Reported by: tobias_e + +2009-08-17 18:50 +0000 [r212574-212581] Sean Bright + + * channels/chan_agent.c: Correct spelling of AGENTACCEPTDTMF in + chan_agent. (closes issue #15668) Reported by: davidw + + * main/logger.c: Correct the return value check for + ast_safe_system. The logic here was reversed as ast_safe_system + returns -1 on error and not on success. Fix suggested by + reporter. (closes issue #15667) Reported by: loic + +2009-08-17 16:50 +0000 [r212506] Jeff Peeler + + * /, channels/misdn_config.c: Merged revisions 212498 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 + Aug 2009) | 12 lines Fix segfault when reloading chan_misdn. If + more ports were specified than configured in misdn.conf a reload + would crash asterisk. The problem was the unconfigured port was + using data from the previously configured port. When the data for + an unconfigured port was freed a crash would result from the + double free. (closes issue #12113) Reported by: agupta Patches: + bug12113.patch uploaded by jpeeler (license 325) ........ + +2009-08-17 16:25 +0000 [r212463] Kevin P. Fleming + + * include/asterisk.h, main/xml.c: Define our desires for POSIX and + X/OPEN API features properly. Based on a post on the gcc-help + mailing list and some subsequent reading, we can increase our + portability to various platforms by directly defining the POSIX + and X/OPEN API feature sets we wish to have available. This patch + does that, and also includes a double-check to ensure that the + system we are compiling on can actually provide the requested + feature sets. + +2009-08-17 15:42 +0000 [r212431] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 212430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix + uninitialized variable causing random MWI indications. (closes + issue #15727) Reported by: doda Patches: dahdi_changes.patch + uploaded by doda (license 853) ........ r212430 | rmudgett | + 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix + uninitialized variable. ........ + +2009-08-16 19:27 +0000 [r212390] Joshua Colp + + * main/rtp_engine.c, include/asterisk/rtp_engine.h: Add two more + API calls for getting the current glue and channel in bridging + code. + +2009-08-15 11:36 +0000 [r212339-212343] Michiel van Baak + + * res/res_calendar.c: cast time_t type variables to long where + needed. This makes res_calendar.c compile on OpenBSD and the same + cast is used in a lot of other places where time_t type vars are + used. (closes issue #15656) Reported by: mvanbaak Patches: + 2009081100-rescalendarcompilefix.diff.txt uploaded by mvanbaak + (license 7) + + * main/xmldoc.c: Add an empty line after each option when printing + the documentation of a function/application. This will make + reading the docs on the CLI way more easy. (closes issue #15694) + Reported by: mvanbaak Patches: + 2009081100-extralinesoptionlist.diff.txt uploaded by mvanbaak + (license 7) + +2009-08-14 23:07 +0000 [r212287-212291] Jeff Peeler + + * channels/sig_analog.c: Add braces where missing and a few + whitespace fixes in sig_analog (closes issue #15678) Reported by: + alecdavis Patches: sig_analog_mainly_braces.diff.txt uploaded by + alecdavis (license 585) Tested by: alecdavis + + * channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h: More code that somehow got left out of + sig_analog * confirmanswer option now respected * check and set + waiting for dialtone timer * unneeded needcallerid flag removed + from analog_subchannel * ss_astchan does not need to be a void + pointer * swap_channels callback updated to trunk * analog_hangup + now resets channel to default law + +2009-08-14 17:36 +0000 [r212249] Tilghman Lesher + + * funcs/func_curl.c: Add SSL_VERIFYPEER, as requested on the -users + list + +2009-08-13 17:33 +0000 [r212199] Richard Mudgett + + * channels/chan_misdn.c: Send a generic return result when we + receive a CallDeflection facility message in chan_misdn. ETSI + 300-196 implies that a facility return result without arguments + does not have the operation-value. This fact implies for ETSI + that you can only use the invoke-id to match requests with + responses. + +2009-08-13 16:44 +0000 [r212161] Joshua Colp + + * main/rtp_engine.c, include/asterisk/rtp_engine.h: Add an API call + for retrieving the engine in use by an RTP instance. + +2009-08-13 15:46 +0000 [r212113] Kevin P. Fleming + + * channels/chan_sip.c: Ensure that T38FaxVersion is put into + outgoing SDP in the proper case. + +2009-08-13 13:51 +0000 [r212067] Joshua Colp + + * channels/chan_sip.c: Check an actual populated variable when + seeing if we need to do video or not. + +2009-08-13 11:37 +0000 [r212027] Gavin Henry + + * contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif: Fixed typo (closes issue #15710) + Reported by: suretec + +2009-08-12 23:14 +0000 [r211947-211957] Matthew Nicholson + + * /, apps/app_queue.c: Merged revisions 211953 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug + 2009) | 10 lines This patch adds additional checking when + generating queue log TRANSFER events. The additional checks + prevent generation of false TRANSFER events in certain + situations. (closes issue #14536) Reported by: aragon Patches: + queue-log-xfer-fix1.diff uploaded by mnicholson (license 96) + Tested by: aragon, mnicholson ........ + + * channels/chan_sip.c, configs/sip.conf.sample: This patch adds + support for choosing a realm based on the domain in the From or + To header in the incoming request. Eligible domains are taken + from the domains list in the config file. This functionality is + enabled when domainsasrealm is enabled in the config file. + (closes issue #11361) Reported by: arkadia Patches: + sip_realm_mnich_to_added_2.patch uploaded by arkadia (license + 233) Tested by: arkadia + +2009-08-12 20:47 +0000 [r211908] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h: Fix chan_dahdi option ringtimeout + dahdi_read relies on the dahdi_pvt copy of ringt which was not + getting set in sig_analog. This patch adds a callback to do so. + (closes issue #15288) Reported by: alecdavis Patches: + chan_dahdi.ringtimeout.diff.txt uploaded by alecdavis (license + 585) Tested by: alecdavis + +2009-08-12 19:53 +0000 [r211876] Matthew Nicholson + + * channels/chan_sip.c: Make asterisk handle 423 Interval Too Short + messages better. This change uses separate values for the + acceptable minimum expiry provided by the 423 error and the + expiry value stored in the configuration file. Previously, the + value pulled from the configuration file would be overwritten. + (closes issue #14366) Reported by: Nick_Lewis Patches: + sip-expiry-fix1.diff uploaded by mnicholson (license 96) + chan_sip.c-reqexpiry.patch uploaded by Nick (license 657) Tested + by: mnicholson + +2009-08-12 16:00 +0000 [r211767] Gavin Henry + + * res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif: Added three new attributes and + applied a patch to res_config_ldap.c attributetype ( + AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC + 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR + caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) + attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC + 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR + caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) + attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent' + DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch + SUBSTR caseIgnoreSubstringsMatch SYNTAX + 1.3.6.1.4.1.1466.115.121.1.15) and patch + fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725) + Reported by: macogeek Patches: + fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license + 863) Tested by: suretec + +2009-08-12 10:11 +0000 [r211732] Russell Bryant + + * channels/chan_jingle.c, channels/chan_unistim.c, + channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_gtalk.c, channels/chan_mgcp.c: Always specify which + RTP engine is desired for a new RTP instance. This fixes a crash + reported in #asterisk-dev where chan_mgcp unexpectedly allocated + an RTP instance from res_rtp_multicast, since by not specifying + an engine, you get the first one in the list of engines. + +2009-08-10 23:21 +0000 [r211675] Richard Mudgett + + * channels/chan_dahdi.c: Encapsulate testing for which signaling + styles are used by sig_pri. Created the + dahdi_sig_pri_lib_handles() function and SIG_PRI_LIB_HANDLE_CASES + macro to simplify testing for which signaling styles are handled + by sig_pri. + +2009-08-10 19:49 +0000 [r211539-211584] Tilghman Lesher + + * doc/CODING-GUIDELINES, /: Merged revisions 211583 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10 + Aug 2009) | 1 line Conversion specifiers, not format specifiers + ........ + + * cel/cel_pgsql.c, funcs/func_speex.c, funcs/func_rand.c, + apps/app_dahdibarge.c, main/frame.c, addons/chan_ooh323.c, + apps/app_readfile.c, /, apps/app_record.c, + apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c, + res/res_http_post.c, channels/chan_iax2.c, main/indications.c, + main/config.c, main/cli.c, pbx/pbx_loopback.c, + channels/chan_dahdi.c, pbx/pbx_spool.c, res/res_smdi.c, + channels/chan_skinny.c, main/features.c, main/http.c, main/pbx.c, + funcs/func_sprintf.c, funcs/func_timeout.c, apps/app_privacy.c, + codecs/codec_speex.c, channels/chan_agent.c, funcs/func_math.c, + apps/app_disa.c, apps/app_morsecode.c, channels/iax2-provision.c, + funcs/func_cut.c, apps/app_talkdetect.c, main/netsock.c, + res/res_config_curl.c, channels/chan_misdn.c, + apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c, + addons/cdr_mysql.c, pbx/pbx_config.c, apps/app_mixmonitor.c, + apps/app_chanspy.c, main/asterisk.c, res/res_odbc.c, + cel/cel_adaptive_odbc.c, main/timing.c, apps/app_voicemail.c, + doc/CODING-GUIDELINES, addons/app_mysql.c, utils/muted.c, + apps/app_meetme.c, main/utils.c, res/res_musiconhold.c, + cdr/cdr_pgsql.c, apps/app_followme.c, res/res_config_sqlite.c, + main/enum.c, utils/frame.c, channels/misdn_config.c, + main/channel.c, res/ael/pval.c, main/cdr.c, funcs/func_enum.c, + channels/chan_phone.c, main/manager.c, apps/app_setcallerid.c, + apps/app_osplookup.c, funcs/func_odbc.c, res/res_agi.c, + apps/app_minivm.c, channels/xpmr/xpmr.c, res/res_config_ldap.c, + apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, + res/res_config_pgsql.c, funcs/func_dialplan.c, main/dnsmgr.c, + channels/chan_sip.c, res/res_limit.c, apps/app_waitforsilence.c, + agi/eagi-test.c, main/acl.c, apps/app_waituntil.c, + apps/app_originate.c, channels/sig_pri.c, apps/app_queue.c, + channels/chan_oss.c, agi/eagi-sphinx-test.c, + channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c, + apps/app_sms.c, utils/extconf.c, apps/app_stack.c, + apps/app_verbose.c, addons/app_saycountpl.c, main/dsp.c, + addons/res_config_mysql.c: AST-2009-005 + +2009-08-10 18:01 +0000 [r211475] Michiel van Baak + + * channels/chan_skinny.c: add manager events when a skinny device + registers/unregisters like we have in chan_sip (closes issue + #15499) Reported by: arifzaman Patches: + 2009072600-skinnymanagerevents.diff.txt uploaded by mvanbaak + (license 7) + +2009-08-10 17:17 +0000 [r211435] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_pri.c: Fix PRI/BRI channels + when in alarm condition to only be marked for hangup if T309 is + not enabled. + +2009-08-10 15:53 +0000 [r211392] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Restoring some code to sig_pri. Not sure if it is really needed. + Putting some DSP code back into sig_pri that was removed by the + chan_dahdi/sig_pri reorganization. + +2009-08-10 15:46 +0000 [r211390] Russell Bryant + + * main/channel.c: Fix up some issues with getting a channel by + "name". Even though the get_channel_by_name() API advertised that + you could search by name or uniqueid (just as the old API did), + searching by uniqueid was not actually implemented. This patch + fixes that problem. The ast_channel_get_full() function now makes + a second search attempt by uniqueid if the parameter was a name. + The channel comparison function also now knows how to compare by + unqieueid. Finally, a bug was fixed in passing where OBJ_POINTER + was being passed in some scenarios where it should not have been. + +2009-08-10 14:07 +0000 [r211347] Joshua Colp + + * channels/chan_sip.c: Fix retrieval of the port used for the video + stream when adding SDP to a SIP message. (closes issue #15121) + Reported by: jsmith + +2009-08-09 15:42 +0000 [r211232-211275] Tilghman Lesher + + * /, main/astfd.c: Merged revisions 211274 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009) + | 2 lines Small oops. Clear the flags which have been checked. + ........ + + * apps/app_stack.c: Check for NULL frame, before dereferencing + pointer. (closes issue #15617) Reported by: rain + +2009-08-07 23:30 +0000 [r211191-211197] Richard Mudgett + + * channels/chan_dahdi.c: Fixed some unsafe down cast pointer + operations for sig_pri. You cannot cast the struct + dahdi_pvt.sig_pvt pointer to a specific signaling private pointer + without first checking that it is in fact pointing to the correct + signaling private structure. + + * channels/sig_pri.c: Fix static on line when PRI does overlap + dialing. The wrong encoding law was used because = was used when + it should have been ==. + +2009-08-07 20:12 +0000 [r211113] Russell Bryant + + * /: Recorded merge of revisions 211112 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009) + | 4 lines Resolve a deadlock involving app_chanspy and + masquerades. (ABE-1936) ........ + +2009-08-07 18:17 +0000 [r211040] Tilghman Lesher + + * /, apps/app_queue.c: Merged revisions 211038 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) + | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name, + not the membername. This is a partial revert of revision 82590, + which was an attempted cleanup, but in reality, it broke + QUEUE_MEMBER_LIST, which has always been intended as a method by + which component interfaces could be queried from the queue. + Membername isn't useful here, because that field cannot be used + to obtain further information about the member. See the + documentation on QUEUE_MEMBER_LIST, RemoveQueueMember, + QUEUE_MEMBER_PENALTY, and the various AMI commands which take a + member argument for further justification. (closes issue #15664) + Reported by: rain Patches: app_queue-queue_member_list.diff + uploaded by rain (license 327) ........ + +2009-08-07 13:08 +0000 [r210992] Kevin P. Fleming + + * main/udptl.c: Workaround broken T.38 endpoints that offer tiny + MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as + the maximum IFP size that should be sent to them, rather than the + maximum packet payload size. If such an endpoint also requests + UDPRedundancy as the error correction mode, we'll end up + calculating a tiny maximum IFP size, so small as to be unusable. + This patch sets a lower bound on what we'll consider the remote's + maximum IFP size to be, assuming that endpoints that do this + really can accept larger packets than they've offered to accept. + (closes issue #15649) Reported by: dazza76 + +2009-08-06 21:46 +0000 [r210908-210914] Tilghman Lesher + + * main/channel.c, /: Merged revisions 210913 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009) + | 7 lines Because channel information can be accessed outside of + the channel thread, we must lock the channel prior to modifying + it. (closes issue #15397) Reported by: caspy Patches: + 20090714__issue15397.diff.txt uploaded by tilghman (license 14) + Tested by: caspy ........ + + * include/asterisk/app.h, main/app.c, apps/app_stack.c: Allow Gosub + to recognize quote delimiters without consuming them. (closes + issue #15557) Reported by: rain Patches: + 20090723__issue15557.diff.txt uploaded by tilghman (license 14) + Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ + +2009-08-06 20:15 +0000 [r210866-210869] Richard Mudgett + + * channels/sig_analog.c: Miscellaneous minor fixes to sig_analog. * + Sanity adjustments to __analog_ss_thread for sig_analog + environment. * Deleted some duplicated code. * Fixed + analog_ss_thread_start passing the wrong pointer. + + * channels/sig_pri.c: Sanity adjustments to pri_ss_thread for + sig_pri environment. + +2009-08-06 17:47 +0000 [r210817] Joshua Colp + + * channels/chan_sip.c: Accept additional T.38 reinvites after an + initial one has been handled. Discussion of this subject has + yielded that it is not actually acceptable to change T.38 + parameters after the initial reinvite but declining is harsh and + can cause the fax to fail when it may be possible to allow it to + continue. This patch changes things so that additional T.38 + reinvites are accepted but parameter changes ignored. This gives + the fax a fighting chance. (closes issue #15610) Reported by: + huangtx2009 + +2009-08-06 16:07 +0000 [r210777] Kevin P. Fleming + + * configure, include/asterisk/autoconfig.h.in, apps/app_fax.c, + configure.ac: Minor improvements to app_fax. This patch makes + some small changes to handle watchdog timeouts in a better way, + and also uses a 'cleaner' method of including the spandsp header + files. (closes issue #14769) Reported by: andrew Patches: + app_fax-20090406.diff uploaded by andrew (license 240) + v1-14769.patch uploaded by dimas (license 88) Tested by: freh, + deti, caspy, dimas, sgimeno, Dovid + +2009-08-05 23:44 +0000 [r210640-210732] Richard Mudgett + + * channels/sig_pri.c: Fix potential deadlock issue with + USERUSERINFO channel variable. + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + More changes from chan_dahdi that did not make it into sig_pri. * + Q.SIG channel mapping option. * discardremoteholdretrieval + option. * libPRI debug defines. * pri_set_overlapdial() now set + correctly. * pthread creation of pri_ss_thread now matches. + + * /, channels/sig_pri.c: Merged revisions 210575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) + | 14 lines Dialplan starts execution before the channel setup is + complete. * Issue 15655: For the case where dialing is complete + for an incoming call, dahdi_new() was asked to start the PBX and + then the code set more channel variables. If the dialplan hungup + before these channel variables got set, asterisk would likely + crash. * Fixed potential for overlap incoming call to erroneously + set channel variables as global dialplan variables if the + ast_channel structure failed to get allocated. * Added missing + set of CALLINGSUBADDR in the dialing is complete case. (closes + issue #15655) Reported by: alecdavis ........ + +2009-08-05 18:49 +0000 [r210564] Leif Madsen + + * doc/tex/imapstorage.tex, /: Merged revisions 210563 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 + Aug 2009) | 11 lines Update imapstorage.txt documentation. + Updated the imapstorage.txt documentation to reflect that issues + with c-client versions older than 2007 seem to cause crashing + issues that are not seen with more recent versions. Documentation + has been updated to reflect this. (closes issue #14496) Reported + by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt + uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, + dbrooks ........ + +2009-08-05 14:09 +0000 [r210522] Russell Bryant + + * main/file.c: Revert some silly code that snuck into trunk from my + working copy. Sorry! + +2009-08-05 08:03 +0000 [r210478] Michiel van Baak + + * addons/mp3: ignore the .i files when compiling in 'DONT_OPTIMIZE' + in the addons/mp3 directory + +2009-08-04 17:46 +0000 [r210353-210387] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Fix CALLERID() values for sig_pri on incoming calls. + + * main/channel.c, include/asterisk/channel.h: Initial minimum + ast_party_caller support. + + * channels/chan_dahdi.c: Removed some dead code. + +2009-08-04 15:35 +0000 [r210302] Jeff Peeler + + * main/features.c: Fix broken call pickup The find_channel_by_group + callback was only looking at the channel that was attempting to + make the pickup instead of the other channels in the container. + +2009-08-04 14:53 +0000 [r210190-210238] Kevin P. Fleming + + * Makefile, /: Merged revisions 210237 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug + 2009) | 10 lines Eliminate spurious compiler warnings from system + headers on *BSD platforms. Ensure that system headers located in + /usr/local/include are actually treated as system headers by the + compiler, and not as local headers which are subject to warnings + from the -Wundef compiler option and others. (closes issue + #15606) Reported by: mvanbaak ........ + + * contrib/scripts/realtime_pgsql.sql, channels/chan_sip.c, + channels/chan_skinny.c, configs/mgcp.conf.sample, + doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt, + configs/res_ldap.conf.sample, configs/sip.conf.sample, + configs/skinny.conf.sample, channels/chan_mgcp.c, + doc/chan_sip-perf-testing.txt: Rename 'canreinvite' option to + 'directmedia', with backwards compatibility. It is clear from + multiple mailing list, forum, wiki and other sorts of posts that + users don't really understand the effects that the 'canreinvite' + config option actually has, and that in some cases they think + that setting it to 'no' will actually cause various other + features (T.38, MOH, etc.) to not work properly, when in fact + this is not the case. This patch changes the proper name of the + option to what it should have been from the beginning + ('directmedia'), but preserves backwards compatibility for + existing configurations. + +2009-08-03 18:05 +0000 [r210094-210154] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_pri.c: Changes from + chan_dahdi that did not make it into sig_pri. * Moved + SUPPORT_USERUSER to sig_pri.c * Fix PRI_DEADLOCK_AVOIDANCE + parameter. * Whitespace changes. * Added missing unlock in + pri_dchannel():PRI_EVENT_RING case. * Balanced curly braces. * + ast_debug/ast_log changes from chan_dahdi. * sig_pri_indicate() + should default to return -1 if the indication is not handled. + + * channels/sig_pri.h, channels/sig_analog.c, channels/sig_pri.c: + Trim trailing whitespace. + +2009-08-03 14:29 +0000 [r210027] Mark Michelson + + * main/channel.c: Fix order and redundancy of channel rename + manager events in ast_do_masquerade. Patch contributed by Mark + Spencer. + +2009-08-03 14:01 +0000 [r209993] Matthew Nicholson + + * addons/chan_mobile.c, configs/chan_mobile.conf.sample: Add an + 'sms' option to mobile.conf to manually enable or disable SMS + support. (closes issue #15071) Reported by: ughnz Patches: + optional-sms1.diff uploaded by mnicholson (license 96) Tested by: + ughnz, mnicholson + +2009-08-01 23:33 +0000 [r209958-209959] Bradley Latus + + * doc/tex/realtime.tex: Update documentation in relation to + UnixODBC (closes issue #15516) Reported by: snuffy Patches: + bug_odbc_tex_update_v2.diff uploaded by snuffy (license 35) + + * doc/CODING-GUIDELINES: (closes issue #15515) + +2009-08-01 11:29 +0000 [r209835-209887] Russell Bryant + + * /, main/db1-ast/mpool/mpool.c: Merged revisions 209879 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009) + | 5 lines Resolve a valgrind warning about a read from + uninitialized memory. (issue #15396) Reported by: aragon ........ + + * /, apps/app_milliwatt.c: Merged revisions 209838 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 + Aug 2009) | 13 lines Modify how Playtones() is used in + Milliwatt() to resolve gain issue. When Milliwatt() was changed + internally to use Playtones() so that the proper tone was used, + it introduced a drop in gain in the output signal. So, use the + playtones API directly and specify a volume argument such that + the output matches the gain of the original Milliwatt() code. + (closes issue #15386) Reported by: rue_mohr Patches: + issue_15386.rev2.diff uploaded by russell (license 2) Tested by: + rue_mohr ........ + + * main/event.c: Fix ast_event_queue_and_cache() to actually do the + cache() part. (closes issue #15624) Reported by: ffossard Tested + by: russell + +2009-08-01 01:04 +0000 [r209760-209761] Kevin P. Fleming + + * Makefile: Revert accidental Makefile change. + + * Makefile, channels/chan_dahdi.c, channels/chan_misdn.c, /, + main/Makefile, channels/misdn/ie.c, pbx/pbx_config.c, + utils/frame.c: Merged revisions 209759 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul + 2009) | 7 lines Minor changes inspired by testing with latest + GCC. The latest GCC (what will become 4.5.x) has a few new + warnings, that in these cases found some either downright buggy + code, or at least seriously poorly designed code that could be + improved. ........ + +2009-07-31 21:53 +0000 [r209711] Russell Bryant + + * main/event.c: Fix some places where ast_event_type was used + instead of ast_event_ie_type. + +2009-07-31 17:57 +0000 [r209673-209674] Mark Michelson + + * configs/sip.conf.sample: Add configuration sample code for + previous commit. + + * channels/chan_sip.c: Improve chan_sip's ability to determine what + methods should and should not be used in a dialog. The previous + effort here was to store what a peer is capable of receiving by + parsing REGISTER requests from the peer and keeping that + information for as long as the registration was active. The + problem with this is that there are a great number of SIP devices + which give no indication of the methods allowed in their REGISTER + requests, and it is unreasonable to try to guess what the device + may or may not support. In addition, some SIP devices have been + found to claim support for a specific method, but their handling + the method is less than ideal, or they are actually lying. With + this patch, we now determine what methods a device supports by + parsing the Allow header we receive from them, and we do this + with each new dialog. In addition, a configuration option has + been added so that an administrator can essentially blacklist + certain methods from being used with certain peers if the admin + knows that support for a specific method is dodgy or nonexistent. + ABE-1822 + +2009-07-30 23:37 +0000 [r209623] Sean Bright + + * configure, configure.ac, makeopts.in: Allow passing 'noisy' to + configure's --enable-dev-mode argument to turn on verbose builds. + (closes issue #15607) Reported by: mvanbaak Patches: + 20090730_issue15607.patch uploaded by seanbright (license 71) + Tested by: seanbright + +2009-07-30 23:31 +0000 [r209619] Jeff Peeler + + * channels/sig_pri.h, channels/sig_pri.c: Add missing ifdef-s for + service maintenance message functionality (closes issue #15614) + Reported by: fabled + +2009-07-30 16:07 +0000 [r209554] David Brooks + + * channels/sig_pri.h, apps/app_forkcdr.c, channels/chan_dahdi.c, + contrib/init.d/rc.debian.asterisk, addons/chan_ooh323.c, + addons/ooh323c/src/ooGkClient.h, funcs/func_math.c, + apps/app_sms.c, codecs/lpc10/pitsyn.c, channels/chan_console.c, + include/asterisk/abstract_jb.h: Fixes numerous spelling errors. + Patch submitted by alecdavis. (closes issue #15595) Reported by: + alecdavis + +2009-07-30 14:38 +0000 [r209516] Mark Michelson + + * channels/chan_sip.c: Fix a crash that can result if text codecs + are allowed but textsupport is disabled. (closes issue #15596) + Reported by: fabled Patches: sip-red.patch uploaded by fabled + (license 448) + +2009-07-29 21:46 +0000 [r209453-209484] Matthew Nicholson + + * addons/chan_mobile.c: This patch adds the ability to send a CUSD + command to a bluetooth device. (closes issue #15278) Reported by: + Artem Patches: cusd5.patch uploaded by Artem (license 800) Tested + by: mnicholson, Artem Review: + https://reviewboard.asterisk.org/r/274/ + + * addons/chan_mobile.c: Fixed a comment for hfp_parse_clip + +2009-07-28 13:49 +0000 [r209400] Kevin P. Fleming + + * channels/chan_usbradio.c, include/asterisk/utils.h, + channels/chan_sip.c, channels/chan_alsa.c, + channels/chan_console.c, channels/chan_oss.c, main/poll.c: Define + side-effect-safe MIN and MAX macros and remove duplicate + definitions from various files. + +2009-07-28 00:20 +0000 [r209317-209331] Tilghman Lesher + + * sounds/sounds.xml: Regex FTL + + * /, sounds/sounds.xml: Merged revisions 209315 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009) + | 2 lines Publish French extra sounds ........ + +2009-07-27 21:43 +0000 [r209256-209279] Kevin P. Fleming + + * apps/app_fax.c: Cleanup T.38 negotiation changes. Convert + LOG_NOTICE messages about T.38 negotiation in debug level 1 + messages, clean up some looping logic, and correct an improper + use of ast_free() for freeing an ast_frame. + + * apps/app_fax.c: Make T.38 switchover in ReceiveFAX synchronous. + In receive mode, if the channel that ReceiveFAX is running on + supports T.38, we should *always* attempt to switch T.38, rather + than listening for an incoming CNG tone and only triggering on + that. The channel may be using a low-bitrate codec that distorts + the CNG tone, the sending FAX endpoint may not send CNG at all, + or there could be a variety of other reasons that we don't detect + it, but in all those cases if T.38 is available we certainly want + to use it. + +2009-07-27 20:54 +0000 [r209132-209235] Mark Michelson + + * res/res_rtp_asterisk.c: Gracefully handle malformed RTP text + packets. AST-2009-004 + + * res/res_musiconhold.c: Honor channel's music class when using + realtime music on hold. (closes issue #15051) Reported by: alexh + Patches: 15051.patch uploaded by mmichelson (license 60) Tested + by: alexh + + * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions + 209131 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul + 2009) | 18 lines Allow for UDPTL to use only even-numbered ports + if desired. There are some VoIP providers out there that will not + accept SDP offers with odd numbered UDPTL ports. While it is my + personal opinion that these VoIP providers are misinterpreting + RFC 2327, it really is not a big deal to play along with their + silly little games. Of course, since restricting UDPTL ports to + only even numbers reduces the range of available ports by half, + so the option to use only even port numbers is off by default. A + user can enable the behavior by setting use_even_ports=yes in + udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: + 15182.patch uploaded by mmichelson (license 60) Tested by: + CGMChris ........ + +2009-07-27 16:33 +0000 [r209098] David Brooks + + * channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, + include/asterisk/module.h, main/features.c, pbx/pbx_dundi.c, + res/res_jabber.c, addons/chan_mobile.c, apps/app_rpt.c, + main/loader.c: Fixing typos. Replaces "recieved" with "received" + and "initilize" with "initialize" (closes issue #15571) Reported + by: alecdavis + +2009-07-27 15:38 +0000 [r209056] Kevin P. Fleming + + * Makefile: Restore explicit export of ASTCFLAGS/ASTLDFLAGS and + underscore-variants to sub-makes. During the recent Makefile + improvements I made, it seemed the 'make' was automatically + carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so + I removed the explict export of them. However, there are some + circumstances where make does this, and some where it does not, + so I've brought them back to ensure they are always exported. I + also removed an extraneous double setting of _ASTLDFLAGS on *BSD + platforms. + +2009-07-27 01:20 +0000 [r208924] Jeff Peeler + + * /, main/translate.c, channels/chan_iax2.c: Merged revisions + 208923 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) + | 2 lines Fix logic errors from 208746 ........ + +2009-07-26 14:00 +0000 [r208886] Michiel van Baak + + * contrib/scripts/install_prereq: add OpenBSD to the install_prereq + script + +2009-07-25 12:28 +0000 [r208813-208848] Michiel van Baak + + * contrib/scripts/install_prereq: libxml2-dev is needed as well by + default. + + * configs/cli_aliases.conf.sample, main/cli.c: add default alias + reload to run module reload. Requiring 'module reload' to reload + everything, including core etc makes russell very unhappy. The + default configuration already loads the 'friendly' aliases + template. Added 'reload=module reload' to that template. Also + removed the comment in main/cli.c that reload should come back. + +2009-07-25 06:23 +0000 [r208749] Jeff Peeler + + * /, channels/chan_skinny.c, main/translate.c, + channels/chan_iax2.c: Merged revisions 208746 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) + | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly + trivial changes, but I did not know of any other way to fix the + "dereferencing type-punned pointer will break strict-aliasing + rules" error without creating a tmp variable in chan_skinny. + ........ + +2009-07-24 21:12 +0000 [r208593-208709] Russell Bryant + + * pbx/pbx_dundi.c: Remove trailing whitespace. + + * main/cli.c: Note that "reload" needs to be added back. I keep + getting annoyed at having to type "module reload" to reload + everything, so I'm adding a note that we need to add "reload" + back. "module reload" doesn't really make sense as the command to + reload everything, including the core. + + * main/cli.c: Don't log a warning for something that does not + affect operation. + + * apps/app_dial.c, /: Merged revisions 208592 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) + | 7 lines Do not log an ERROR if autoservice_stop() returns -1. + This does not indicate an error. A return of -1 just means that + the channel has been hung up. (reported in #asterisk-dev) + ........ + +2009-07-24 18:31 +0000 [r208588] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 208587 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul + 2009) | 10 lines Only send a BYE when hanging up a channel that + is up. For cases where Asterisk sends an INVITE and receives a + non 2XX final response, Asterisk would follow the INVITE + transaction by immediately sending a BYE, which was unnecessary. + (closes issue #14575) Reported by: chris-mac ........ + +2009-07-24 15:02 +0000 [r208548] Kevin P. Fleming + + * main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h: + Resolve a T.38 negotiation issue left over from the udptl-updates + merge. The udptl-updates branch that was merged yesterday failed + to properly send back T.38 SDP responses with the correct error + correction mode, if the incoming SDP from the other end caused us + to change error correction modes. This patch corrects that + situation. + +2009-07-24 14:35 +0000 [r208542] Michiel van Baak + + * contrib/scripts/install_prereq: use aptitude for debian based + systems The function to check wether we need to install packages + was using dpkg-query which was gives wrong output on Debian 5 + Also, the apt-get has been replaced with aptitude because + aptitude is now the preferred way to handle packages on Debian + (closes issue #15570) Reported by: mvanbaak Patches: + 2009072400_installprereq-aptitude.diff uploaded by mvanbaak + (license 7) + +2009-07-23 22:32 +0000 [r208464-208504] Kevin P. Fleming + + * UPGRADE.txt: T.38 change note is not necessary in this branch + + * main/channel.c, main/udptl.c, main/frame.c, main/rtp_engine.c, + channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt, + include/asterisk/udptl.h, include/asterisk/frame.h: Rework of + T.38 negotiation and UDPTL API to address interoperability + problems Over the past couple of months, a number of issues with + Asterisk negotiating (and successfully completing) T.38 sessions + with various endpoints have been found. This patch attempts to + address many of them, primarily focused around ensuring that the + endpoints' MaxDatagram size is honored, and in addition by + ensuring that T.38 session parameter negotiation is performed + correctly according to the ITU T.38 Recommendation. The major + changes here are: 1) T.38 applications in Asterisk (app_fax) only + generate/receive IFP packets, they do not ever work with UDPTL + packets. As a result of this, they cannot be allowed to generate + packets that would overflow the other endpoints' MaxDatagram size + after the UDPTL stack adds any error correction information. With + this patch, the application is told the maximum *IFP* size it can + generate, based on a calculation using the far end MaxDatagram + size and the active error correction mode on the T.38 session. + The same is true for sending *our* MaxDatagram size to the remote + endpoint; it is computed from the value that the application says + it can accept (for a single IFP packet) combined with the active + error correction mode. 2) All treatment of T.38 session + parameters as 'capabilities' in chan_sip has been removed; these + parameters are not at all like audio/video stream capabilities. + There are strict rules to follow for computing an answer to a + T.38 offer, and chan_sip now follows those rules, using the + desired parameters from the application (or channel) that wants + to accept the T.38 negotiation. 3) chan_sip now stores and + forwards ast_control_t38_parameters structures for tracking 'our' + and 'their' T.38 session parameters; this greatly simplifies + negotiation, especially for pass-through calls. 4) Since T.38 + negotiation without specifying parameters or receiving the final + negotiated parameters is not very worthwhile, the AST_CONTROL_T38 + control frame has been removed. A note has been added to + UPGRADE.txt about this removal, since any out-of-tree + applications that use it will no longer function properly until + they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: + https://reviewboard.asterisk.org/r/310/ + +2009-07-23 19:34 +0000 [r208388] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 208386 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul + 2009) | 17 lines Fix a problem where a 491 response could be sent + out of dialog. This generalizes the fix for issue 13849. The + initial fix corrected the problem that Asterisk would reply with + a 491 if a reinvite were received from an endpoint and we had not + yet received an ACK from that endpoint for the initial INVITE it + had sent us. This expansion also allows Asterisk to appropriately + handle an INVITE with authorization credentials if Asterisk had + not received an ACK from the previous transaction in which + Asterisk had responded to an unauthorized INVITE with a 407. + (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch + uploaded by mmichelson (license 60) Tested by: klaus3000 ........ + +2009-07-23 19:21 +0000 [r208383] Jeff Peeler + + * channels/chan_dahdi.c, /: Merged revisions 208380 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 + Jul 2009) | 6 lines Only set the priindication setting when not + performing a reload (closes issue #14696) Reported by: fdecher + ........ + +2009-07-23 16:29 +0000 [r208314] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 208312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul + 2009) | 3 lines Remove inaccurate XXX comment. ........ + +2009-07-23 15:59 +0000 [r208267] Jeff Peeler + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Fix sending of interface identifier unconditionally in sig_pri + The wrong logic was being used in chan_dahdi to convert a + sig_pri_chan to the proper libpri channel number. The most + significant bit must only be set only when trunk groups are being + used. (closes issue #15452) Reported by: alecdavis Patches: + bug15452.patch uploaded by jpeeler (license 325) Tested by: + alecdavis + +2009-07-23 15:46 +0000 [r208229-208263] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 208262 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul + 2009) | 8 lines Properly handle 183 responses which do not + contain an SDP. (closes issue #15442) Reported by: ffloimair + Patches: 15442.patch uploaded by mmichelson (license 60) Tested + by: tkarl, ffloimair ........ + + * channels/chan_sip.c: Fix potential crash if p->owner is NULL. + Problem was observed when a call-forwarding loop was accidentally + configured. ABE-1906 + +2009-07-23 01:31 +0000 [r208193] Russell Bryant + + * main/cel.c: Resolve compiler warning on mac. + +2009-07-22 22:42 +0000 [r208155] Jeff Peeler + + * channels/chan_dahdi.c: Reset the fax buffers back to default + settings regardless of signaling in use - Pointed out by Matt F. + Also in the case of not using a signaling module, set the law + back to the default as well. + +2009-07-22 22:35 +0000 [r208151] Tilghman Lesher + + * /, include/asterisk/compat.h, main/strcompat.c, + main/asterisk.exports: Merged revisions 208083 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r208083 | tilghman | 2009-07-22 15:23:53 -0500 (Wed, 22 Jul 2009) + | 4 lines Export symbols for functions included in our + compatibility headers. (closes issue #15556) Reported by: smw1218 + ........ + +2009-07-22 21:43 +0000 [r208113] Jason Parker + + * apps/app_festival.c: Restore an int declaration on PPC platforms. + This x is one crafty little bugger... It was used for 2 different + things (one of which was only done on PPC) in 1.4. One of the + uses were removed in trunk, and with it went the declaration. + (closes issue #14038) Reported by: ffloimair + +2009-07-22 16:49 +0000 [r208052] Tilghman Lesher + + * res/res_realtime.c: Clarify documentation on 'realtime update2' + to show more than one condition. (closes issue #15357) Reported + by: snuffy Patches: bug_fix_doc_update2.diff uploaded by snuffy + (license 35) (slightly modified by me) + +2009-07-22 14:35 +0000 [r208018] Russell Bryant + + * include/asterisk/channel.h: Remove trailing whitespace. + +2009-07-22 14:35 +0000 [r208017] Mark Michelson + + * apps/app_directed_pickup.c: Fix the crash in directed pickups. + For real this time. A shallow pointer copy was causing an + ast_party_connected_line structure to be freed multiple times, + thus causing a crash. (closes issue #15441) Reported by: + lmsteffan Patches: 15441.patch uploaded by mmichelson (license + 60) Tested by: lmsteffan + +2009-07-21 22:51 +0000 [r207950] Jeff Peeler + + * channels/sig_pri.c: Do not dial digits when none were specified + for sig_pri based calls (closes issue #15524) Reported by: + elguero Patches: pri-sig-no-dest-set.patch uploaded by elguero + (license 37) + +2009-07-21 22:45 +0000 [r207946] Tilghman Lesher + + * /, funcs/func_strings.c: Merged revisions 207945 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 + Jul 2009) | 8 lines Force an error if a blank is passed to QUOTE + (because the documentation states the argument is not optional). + This change makes URIENCODE and QUOTE behave similarly, since the + documentation states that the argument is not optional, for both. + (closes issue #15439) Reported by: pkempgen Patches: + 20090706__issue15439.diff.txt uploaded by tilghman (license 14) + ........ + +2009-07-21 22:24 +0000 [r207934] Jeff Peeler + + * channels/chan_dahdi.c: whitespace fix only + +2009-07-21 22:22 +0000 [r207925] Russell Bryant + + * doc/CODING-GUIDELINES: Note that we use tabs instead of spaces + for indentation. I'm surprised this was never actually in here... + +2009-07-21 22:02 +0000 [r207854-207902] Jeff Peeler + + * channels/chan_dahdi.c: Fix my_is_off_hook to check rxbits only + for FXS signaling + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 207827 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) + | 9 lines Wait for wink before dialing when using E&M wink + signaling There was already code for other signaling types in + dahdi_handle_event to handle dialing if a dial operation dial + string was present. Simply add SIG_EMWINK to the list. (closes + issue #14434) Reported by: araasch ........ + +2009-07-21 14:29 +0000 [r207723] Mark Michelson + + * main/manager.c, /: Merged revisions 207714 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul + 2009) | 5 lines Document default timeout for AMI originations. + AST-224 ........ + +2009-07-21 13:28 +0000 [r207680] Kevin P. Fleming + + * /, main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, + res/Makefile, pbx/Makefile, Makefile.rules, channels/Makefile, + doc/video_console.txt, Makefile, utils/Makefile, codecs/Makefile, + agi/Makefile, addons/Makefile, funcs/Makefile, + codecs/lpc10/Makefile, main/db1-ast/Makefile: Merged revisions + 207647 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul + 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are + honored. This commit changes the build system so that + user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to + the compiler/linker *after* all flags provided by the build + system itself, so that the user can effectively override the + build system's flags if desired. In addition, ASTCFLAGS and + ASTLDFLAGS can now be provided *either* in the environment before + running 'make', or as variable assignments on the 'make' command + line. As a result, the use of COPTS and LDOPTS is no longer + necessary, so they are no longer documented, but are still + supported so as not to break existing build systems that supply + them when building Asterisk. ........ + +2009-07-20 23:08 +0000 [r207522-207551] Mark Michelson + + * apps/app_directed_pickup.c: Okay, that didn't fix the crash. It + didn't really do anything useful. + + * apps/app_directed_pickup.c: Initialize connected line instance + when doing a directed pickup. This helps to prevent a crash which + may occur due to our freeing garbage due to a struct being + uninitialized. + +2009-07-20 20:45 +0000 [r207484] David Vossel + + * channels/chan_sip.c: reg->username is parsed only once on sip + reload The registration string can contain an expanded user + portion of the form user@domain. This expanded user portion was + stored in reg->username and parsed each time there is a + registration refresh. Now, the domain portion of the user is + parsed and stored separately in the regdomain field. (closes + issue #14331) Reported by: Nick_Lewis Patches: + chan_sip.c.domainparse3.patch uploaded by Nick (license 657) + Tested by: Nick_Lewis, dvossel + +2009-07-20 19:48 +0000 [r207424] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 207423 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul + 2009) | 33 lines Answer video SDP offers properly when + videosupport is not enabled. Copied from Review board: In issue + 12434, the reporter describes a situation in which audio and + video is offered on the call, but because videosupport is + disabled in sip.conf, Asterisk gives no response at all to the + video offer. According to RFC 3264, all media offers should have + a corresponding answer. For offers we do not intend to actually + reply to with meaningful values, we should still reply with the + port for the media stream set to 0. In this patch, we take note + of what types of media have been offered and save the information + on the sip_pvt. The SDP in the response will take into account + whether media was offered. If we are not otherwise going to + answer a media offer, we will insert an appropriate m= line with + the port set to 0. It is important to note that this patch is + pretty much a bandage being applied to a broken bone. The patch + *only* helps for situations where video is offered but + videosupport is disabled and when udptl_pt is disabled but T.38 + is offered. Asterisk is not guaranteed to respond to every media + offer. Notable cases are when multiple streams of the same type + are offered. The 2 media stream limit is still present with this + patch, too. In trunk and the 1.6.X branches, things will be a bit + different since Asterisk also supports text in SDPs as well. + (closes issue #12434) Reported by: mnnojd Review: + https://reviewboard.asterisk.org/r/311 Review: + https://reviewboard.asterisk.org/r/313 ........ + +2009-07-20 16:36 +0000 [r207361] Russell Bryant + + * main/channel.c, /: Merged revisions 207360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) + | 9 lines Only do the chan->fdno check in ast_read() in a + developer build. I changed this check to only happen in a + dev-mode build. I also added a comment explaining what is going + on. I also made it so that detection of this situation does not + affect ast_read() operation. (closes issue #14723) Reported by: + seadweller ........ + +2009-07-18 04:17 +0000 [r207318] Richard Mudgett + + * channels/chan_misdn.c, CHANGES: Merged 207316 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... + .......... r207316 | rmudgett | 2009-07-17 23:05:05 -0500 (Fri, + 17 Jul 2009) | 20 lines Fixed incoming calls being matched to + MSNs without type-of-number prefix added. For an incoming ISDN + call the dialed.number is incorrectly matched against the + configured MSNs in misdn.conf. The numbers passed to the dialplan + include the configured prefix for the dialed.number_type, whereas + the check against the configured MSNs (to decide if the call is + accepted at all), is executed without the configured prefix. + e.g., dialed.number = 241168020, TON = national, configured + national prefix is "0". (This is the TON which is used by ISDN + providers in the Netherlands.) In chan_misdn.c:cb_events() in + case EVENT_SETUP the call to misdn_cfg_is_msn_valid() uses the + unnormalized number 241168020, but 57 lines later the call to + read_config() adds the prefix, and the dialed.number is now + 0241168020, which is then used in the dialplan. + misdn_cfg_is_msn_valid() must use the normalized number, too. + JIRA ABE-1912 + +2009-07-18 04:16 +0000 [r207317] Tilghman Lesher + + * apps/app_voicemail.c: Flag field in wrong position. Reported by + "Hoggins!" on asterisk-dev list. + +2009-07-18 01:31 +0000 [r207285] Richard Mudgett + + * /: Recorded merge of revisions 145293,158010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 + (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c + channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk + to make merging easier later. ........ r145200 | rmudgett | + 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * + Miscellaneous formatting changes to make v1.4 and trunk more + merge compatible in the mISDN area. channels/chan_misdn.c * + Eliminated redundant code in cb_events() EVENT_SETUP ........ + r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) + | 9 lines improved helptext of misdn_set_opt. ........ r142181 | + rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line + Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 + 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines + channels/chan_misdn.c * Made bearer2str() use + allowed_bearers_array[] * Made use the causes.h defines instead + of hardcoded numbers. * Made use Asterisk presentation indicator + values if either of the mISDN presentation or screen options are + negative. * Updated the misdn_set_opt application option + descriptions. * Renamed the awkward Caller ID presentation + misdn_set_opt application option value not_screened to + restricted. Deprecated the not_screened option value. + channels/misdn/isdn_lib.c * Made use the causes.h defines instead + of hardcoded numbers. * Fixed some spelling errors and typos. * + Added all defined facility code strings to fac2str(). + channels/misdn/isdn_lib.h * Added doxygen comments to struct + misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen + comments to struct misdn_stack. channels/misdn_config.c + configs/misdn.conf.sample * Updated the mISDN presentation and + screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) + * Updated the misdn_set_opt application option descriptions. * + Fixed some spelling errors and typos. ................ r158010 | + rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines + Merged revision 157977 from + https://origsvn.digium.com/svn/asterisk/team/group/issue8824 + ........ Fixes JIRA ABE-1726 The dial extension could be empty if + you are using MISDN_KEYPAD to control ISDN provider features. + ................ + +2009-07-17 22:29 +0000 [r207255] Tilghman Lesher + + * doc/voicemail_odbc_postgresql.txt: Add flag here, too (as + requested by jsmith) + +2009-07-17 22:07 +0000 [r207225] David Vossel + + * channels/chan_iax2.c: fixes an error in r203638 CEL commit + (closes issue #15525) Reported by: elguero Patches: + iax2-double-unlock.patch uploaded by elguero (license 37) + 15525.diff uploaded by dvossel (license 671) Tested by: dvossel + +2009-07-17 22:04 +0000 [r207224] Tilghman Lesher + + * doc/tex/odbcstorage.tex, UPGRADE.txt: Document the "flag" field + in the voicemessages table. + +2009-07-17 19:37 +0000 [r207095-207156] Jeff Peeler + + * channels/chan_dahdi.c, /: Merged revisions 207155 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 + Jul 2009) | 7 lines Fix format specifier to print out an unsigned + long long. Yep, it's even ifdefed out code. But it made it to the + RR list... (closes issue #14726) Reported by: lmadsen ........ + + * configs/chan_dahdi.conf.sample: Update some missing allowed + options for overlapdial + +2009-07-17 17:51 +0000 [r207029] David Vossel + + * channels/chan_sip.c: sip option flags handled incorrectly (closes + issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, + Takehiko_Ooshima + +2009-07-17 17:02 +0000 [r206998] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_analog.c: Fix segfault in + sig_analog when using callwaiting, respect callwaiting options + Sig_analog handles allocating the sub channel for callwaiting, so + no longer try to do it in chan_dahdi. Modified analog_alloc_sub + to only mark the sub as allocated upon success of the alloc_sub + callback, which was responsible for the segfault. Also, the + callwaiting and callwaitingcallerid options were being + unconditionally set to true. Now, the options are properly set + from chan_dahdi.conf. (closes issue #15508) Reported by: elguero + Tested by: elguero + +2009-07-17 16:13 +0000 [r206868-206939] David Vossel + + * /, channels/chan_sip.c: Merged revisions 206938 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) + | 14 lines SIP incorrect From: header information when callpres + is prohib Some ITSP make use of the "Anonymous" display name to + detect a requirement to withhold caller id across the PSTN. This + does not work if the display name is "Unknown". (closes issue + #14465) Reported by: Nick_Lewis Patches: + chan_sip.c-callerpres.patch uploaded by Nick (license 657) + chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license + 671) Tested by: Nick_Lewis, dvossel ........ + + * funcs/func_timeout.c: TIMEOUT(absolute) returned negative value. + (closes issue #15513) Reported by: ys + + * configs/iax.conf.sample, /: Merged revisions 206872 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 + Jul 2009) | 6 lines error in iax.conf related IP-based access + control (closes issue #15518) Reported by: pkempgen ........ + + * /, main/callerid.c: Merged revisions 206867 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) + | 8 lines avoid segfault caused by user error If the CALLERPRES() + dialplan function is set to nothing, a segfault occurs. This is + user error to begin with, but I'd rather see a cli warning + message than have Asterisk crash on me. ........ + +2009-07-16 16:51 +0000 [r206808] Tilghman Lesher + + * /, funcs/func_realtime.c: Merged revisions 206807 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 + Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517) + Reported by: adomjan Patches: + func_realtime.c-ast_variable_destroy.diff uploaded by adomjan + (license 487) ........ + +2009-07-15 22:04 +0000 [r206768] David Vossel + + * channels/chan_sip.c: Session timer were not activated if + Supported header field in INVITE had both "timer" and other + options. (closes issue #15403) Reported by: makoto Patches: + sip-session-timer.patch uploaded by makoto (license 38) + +2009-07-15 22:02 +0000 [r206767] Jeff Peeler + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h, channels/sig_pri.c: The dialing flag was + mistakingly removed from sig_pri. This readds the proper setting + of the flag and is really a continuation of r205731. The flag was + being set properly in sig_analog, but use of the newly added + set_dialing callback allowed for some simplification in + chan_dahdi. (closes issue #15486) Reported by: rmudgett + +2009-07-15 21:14 +0000 [r206707] Richard Mudgett + + * channels/misdn/isdn_lib_intern.h, /, channels/misdn/isdn_lib.c: + Merged revisions 206706 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500 + (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... + .......... Fixed chan_misdn crash because mISDNuser library is + not thread safe. With Asterisk the mISDNuser library is driven by + two threads concurrently: 1. + channels/misdn/isdn_lib.c::manager_event_handler() 2. + channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls + into the library are done concurrently and recursively from + isdn_lib.c. Both threads can fiddle with the master/child + layer3_proc_t lists. One thread may traverse the list when the + other interrupts it and then removes the list element which the + first thread was currently handling. This is exactly what caused + the crash. About 60 calls were needed to a Gigaset CX475 before + it occurred once. This patch adds locking when calling into the + mISDNuser library. This also fixes some cb_log calls with wrong + port parameter. JIRA ABE-1913 Patches: misdn-locking.patch + (Modified with mostly cosmetic changes) .......... + ................ + +2009-07-15 20:20 +0000 [r206702] David Vossel + + * channels/chan_sip.c: callerid(num) is wrong when username is + missing A domain only sip uri would return + 123.123.123.123 as callid num. Now, if the username is missing + from a uri, the callerid num field is left empty. (closes issue + #15476) Reported by: viraptor + +2009-07-15 16:00 +0000 [r206636] Sean Bright + + * /, codecs/codec_dahdi.c: Merged revisions 206635 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, + 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we + are asking for it. ........ + +2009-07-14 20:38 +0000 [r206603] Jeff Peeler + + * configs/chan_dahdi.conf.sample: fix a typo in sample config file + for option change + +2009-07-14 20:14 +0000 [r206567] Tilghman Lesher + + * apps/app_meetme.c, contrib/scripts/meetme.sql: Document all + meetme realtime fields, and in the process, make some field + lengths more consistent. (closes issue #15493) Reported by: lasko + Patches: meetme.diff uploaded by lasko (license 833) + +2009-07-14 20:01 +0000 [r206566] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h: Restore some missing functionality to + sig_analog. The main purpose of this commit is to restore missing + functionality present in the ss_thread before all the sig related + work was done. Two of the biggest missing things were distinctive + ring detection and cid handling for V23. fxsoffhookstate and + associated mwi variables have been moved inside sig_analog as + they were not being set properly as well. + +2009-07-14 17:03 +0000 [r206490] Mark Michelson + + * apps/app_dial.c: I AM A TERRIBLE PERSON + +2009-07-14 17:01 +0000 [r206489] Richard Mudgett + + * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, + channels/misdn/isdn_lib.c: Merged revisions 206487 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 + Jul 2009) | 28 lines Fixes several call transfer issues with + chan_misdn. * issue #14355 - Crash if attempt to transfer a call + to an application. Masquerade the other pair of the four asterisk + channels involved in the two calls. The held call already must be + a bridged call (not an applicaton) or it would have been + rejected. * issue #14692 - Held calls are not automatically + cleared after transfer. Allow the core to initate disconnect of + held calls to the ISDN port. This also fixes a similar case where + the party on hold hangs up before being transferred or taken off + hold. * JIRA ABE-1903 - Orphaned held calls left in + music-on-hold. Do not simply block passing the hangup event on + held calls to asterisk core. * Fixed to allow held calls to be + transferred to ringing calls. Previously, held calls could only + be transferred to connected calls. * Eliminated unused call + states to simplify hangup code. * Eliminated most uses of + "holded" because it is not a word. (closes issue #14355) (closes + issue #14692) Reported by: sodom Patches: + misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) + Tested by: rmudgett ........ + +2009-07-14 16:09 +0000 [r206455] Mark Michelson + + * apps/app_dial.c: Reset the sentringing indication when redirects + occur. If a redirecting control frame is processed or a call + forward occurs, we need to reset the sentringing flag so that we + can send another ringing indication to the phone that may contain + a connected line update. AST-164 + +2009-07-14 14:51 +0000 [r206386] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 206385 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r206385 | russell | 2009-07-14 09:48:00 -0500 + (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) + | 6 lines Ensure apathetic replies are sent out on the proper + socket. chan_iax2 supports multiple address bindings. The + send_apathetic_reply() function did not attempt to send its + response on the same socket that the incoming message came in on. + ........ ................ + +2009-07-14 00:48 +0000 [r206341] Richard Mudgett + + * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged + revisions 206284 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) + | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 + ........ + +2009-07-13 23:26 +0000 [r206280] David Vossel + + * channels/chan_sip.c: dns lookup of peername rather than peer's + host in transmit_register() (closes issue #15052) Reported by: + fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch + uploaded by fsantulli (license 818) Tested by: fsantulli + +2009-07-13 18:46 +0000 [r206225] Sean Bright + + * contrib/upstart/asterisk.upstart-0.3.9: Make sure that since we + are passing -c to asterisk that we have a console. Without this + line, Asterisk will busy-loop trying to read and write to + /dev/null (woops... my bad). + +2009-07-13 16:23 +0000 [r206185] Tilghman Lesher + + * apps/app_voicemail.c: Remove reference to non-existent help file + (closes issue #15427) Reported by: brushtyler Patches: + app_voicemail.c.diff uploaded by brushtyler (license 821) + +2009-07-13 14:06 +0000 [r206092-206094] Kevin P. Fleming + + * .cleancount: Bump up cleancount so that existing checkouts will + update themselves properly for the 'Addons' -> 'ADDONS' change. + + * addons/Makefile: Make the menuselect category for Add-Ons + consistent with the other directories (uppercase). + +2009-07-11 19:30 +0000 [r206021-206049] Russell Bryant + + * CHANGES: note the security events API in CHANGES + + * doc/tex/security-events.tex (added), tests/test_security_events.c + (added), main/manager.c, main/security_events.c (added), + include/asterisk/event_defs.h, main/event.c, + include/asterisk/security_events.h (added), doc/tex/asterisk.tex, + include/asterisk/security_events_defs.h (added), + res/res_security_log.c (added), tests/test_ami_security_events.sh + (added): Add an API for reporting security events, and a security + event logging module. This commit introduces the security events + API. This API is to be used by Asterisk components to report + events that have security implications. A simple example is when + a connection is made but fails authentication. These events can + be used by external tools manipulate firewall rules or something + similar after detecting unusual activity based on security + events. Inside of Asterisk, the events go through the ast_event + API. This means that they have a binary encoding, and it is easy + to write code to subscribe to these events and do something with + them. One module is provided that is a subscriber to these events + - res_security_log. This module turns security events into a + parseable text format and sends them to the "security" logger + level. Using logger.conf, these log entries may be sent to a + file, or to syslog. One service, AMI, has been fully updated for + reporting security events. AMI was chosen as it was a fairly + straight forward service to convert. The next target will be + chan_sip. That will be more complicated and will be done as its + own project as the next phase of security events work. For more + information on the security events framework, see the + documentation generated from doc/tex/. "make asterisk.pdf" + Review: https://reviewboard.asterisk.org/r/273/ + +2009-07-10 21:42 +0000 [r205985] David Vossel + + * channels/chan_sip.c: SIP register not using peer's outbound proxy + If callbackextension is defined for a peer it successfully causes + a registration to occur, but the registration ignores the + outboundproxy settings for the peer. This patch allows the peer + to be passed to obproxy_get() in transmit_register(). (closes + issue #14344) Reported by: Nick_Lewis Patches: + callbackextension_peer_trunk.diff uploaded by dvossel (license + 671) Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/294/ + +2009-07-10 18:44 +0000 [r205939] Kevin P. Fleming + + * main/udptl.c: Update comments about the level of T.38 support in + Asterisk. + +2009-07-10 17:39 +0000 [r205878] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 205877 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 + (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 + (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul + 2009) | 10 lines Ensure that outbound NOTIFY requests are + properly routed through stateful proxies. With this change, we + make note of Record-Route headers present in any SUBSCRIBE + request that we receive so that our outbound NOTIFY requests will + have the proper Route headers in them. (closes issue #14725) + Reported by: ibc ........ ................ ................ + +2009-07-10 16:42 +0000 [r205840] David Vossel + + * /, channels/chan_sip.c: Merged revisions 205804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) + | 31 lines SIP registration auth loop caused by stale nonce If an + endpoint sends two registration requests in a very short period + of time with the same nonce, both receive 401 responses from + Asterisk, each with a different nonce (the second 401 containing + the current nonce and the first one being stale). If the endpoint + responds to the first 401, it does not match the current nonce so + Asterisk sends a third 401 with a newly generated nonce (which + updates the current nonce)... Now if the endpoint responds to the + second 401, it does not match the current nonce either and + Asterisk sends a fourth 401 with a newly generated nonce... This + loop goes on and on. There appears to be a simple fix for this. + If the nonce from the request does not match our nonce, but is a + good response to a previous nonce, instead of sending a 401 with + a newly generated nonce, use the current one instead. This breaks + the loop as the nonce is not updated until a response is + received. Additional logic has been added to make sure no nonce + can be responded to twice though. (closes issue #15102) Reported + by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license + 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: + Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ + +2009-07-10 16:00 +0000 [r205780] Kevin P. Fleming + + * apps/app_fax.c: Eliminate extraneous LOG_DEBUG messages generated + by app_fax. The transmit_audio() and transmit_t38() functions in + app_fax have processing loops that are supposed to wait for + frames to arrive on the channel and then handle them, but they + also have short timeouts so that the loops can have watchdog + timers and do other required processing. This commit changes the + loops to not actually call ast_read() and attempt to process the + returned frame unless a frame actually arrived, eliminating + hundreds of LOG_DEBUG messages and slightly improving + performance. + +2009-07-10 15:56 +0000 [r205776] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 205775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul + 2009) | 10 lines Ensure that outbound NOTIFY requests are + properly routed through stateful proxies. With this change, we + make note of Record-Route headers present in any SUBSCRIBE + request that we receive so that our outbound NOTIFY requests will + have the proper Route headers in them. (closes issue #14725) + Reported by: ibc ........ + +2009-07-10 15:28 +0000 [r205770] Kevin P. Fleming + + * apps/app_fax.c: Fix some remaining T.38 negotiation problems in + app_fax. Revision 205696 did not quite fix all the issues with + the T.38 negotiation changes and app_fax; this patch corrects + them, along with a couple of other minor issues. (closes issue + #15480) Reported by: dimas Patches: test2-15480.patch uploaded by + dimas (license 88) + +2009-07-09 21:32 +0000 [r205700] Matthew Nicholson + + * addons/chan_mobile.c: Fix mbl_fixup() in chan_mobile to update + newchan->tech_pvt instead of oldchan. (closes issue #15299) + Reported by: nikkk + +2009-07-09 21:20 +0000 [r205696] Kevin P. Fleming + + * channels/chan_sip.c, apps/app_fax.c, include/asterisk/frame.h: + Repair ability of SendFAX/ReceiveFAX to respond to T.38 + switchover. Recent changes in T.38 negotiation in Asterisk caused + these applications to not respond when the other endpoint + initiated a switchover to T.38; this resulted in the T.38 + switchover failing, and the FAX attempt to be made using an audio + connection, instead of T.38 (which would usually cause the FAX to + fail completely). This patch corrects this problem, and the + applications will now correctly respond to the T.38 switchover + request. In addition, the response will include the appopriate + T.38 session parameters based on what the other end offered and + what our end is capable of. (closes issue #14849) Reported by: + afosorio + +2009-07-09 20:04 +0000 [r205666] Matthew Nicholson + + * funcs/func_odbc.c: Convert func_odbc to use + ast_dummy_alloc_channel() Review: + https://reviewboard.asterisk.org/r/290/ + +2009-07-09 16:19 +0000 [r205600] David Vossel + + * /, include/asterisk/time.h: Merged revisions 205599 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 + Jul 2009) | 2 lines Changing ast_samp2tv to not use floating + point. ........ + +2009-07-09 14:10 +0000 [r205532-205562] Michiel van Baak + + * main/cel.c: make this compile again under devmode + + * main/ssl.c: pthread_self returns a pthread_t which is not an + unsigned int on all pthread implementations. Casting it to an + unsigned int fixes compiler warnings. Tested on OpenBSD and Linux + both 32 and 64 bit + +2009-07-08 23:19 +0000 [r205479] David Vossel + + * res/res_rtp_asterisk.c, /, channels/chan_iax2.c, + include/asterisk/frame.h: Merged revisions 205471 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 + Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations + assume 8khz is the codec rate. This is not always the case. This + patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am + sure there are other areas that make this assumption as well. + Review: https://reviewboard.asterisk.org/r/306/ ........ + +2009-07-08 23:07 +0000 [r205469] Matthew Nicholson + + * main/pbx.c: Fix a CEL related regression with hints updating by + subscribing to AST_DEVICE_STATE instead of + AST_DEVICE_STATE_CHANGED. (closes issue #15440) Reported by: + lmsteffan + +2009-07-08 22:15 +0000 [r205410-205412] David Vossel + + * include/asterisk/devicestate.h, main/pbx.c, /, + main/devicestate.c, include/asterisk/pbx.h: Merged revisions + 205409 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) + | 6 lines moving ast_devstate_to_extenstate to pbx.c from + devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This + change fixes a compile time error with chan_vpb as well. ........ + + * main/devicestate.c: missing comma in devstatestring array + +2009-07-08 19:26 +0000 [r205350] Mark Michelson + + * /, apps/app_queue.c: Merged revisions 205349 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul + 2009) | 14 lines Prevent phantom calls to queue members. If a + caller were to hang up while a periodic announcement or position + were being said, the return value for those functions would + incorrectly indicate that the caller was still in the queue. With + these changes, the problem does not occur. (closes issue #14631) + Reported by: latinsud Patches: queue_announce_ghost_call2.diff + uploaded by latinsud (license 745) (with small modification from + me) ........ + +2009-07-08 18:19 +0000 [r205291] Jason Parker + + * config.sub, /, config.guess: Merged revisions 205288 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul + 2009) | 1 line Update config.guess and config.sub from the + savannah.gnu.org git repo. ........ + +2009-07-08 17:26 +0000 [r205254] David Brooks + + * main/features.c: Fixes Park() argument handling Park() was not + respecting the arguments passed to it. Any + extension/context/priority given to it was being ignored. This + patch remedies this. (closes issue #15380) Reported by: DLNoah + +2009-07-08 16:59 +0000 [r205221] Tilghman Lesher + + * main/say.c: Oops, fixing build + +2009-07-08 16:54 +0000 [r205216] David Vossel + + * /, include/asterisk/time.h: Merged revisions 205215 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 + Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz + audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is + 16000. The .5 is currently stripped off because we don't + calculate using floating points. This causes madness with 16khz + audio. (issue ABE-1899) Review: + https://reviewboard.asterisk.org/r/305/ ........ + +2009-07-08 16:43 +0000 [r205214] Sean Bright + + * utils/muted.c, configure, include/asterisk/autoconfig.h.in, + configure.ac, main/dns.c: Fix a few compilation problems found + when building Asterisk against uClibc. + +2009-07-08 16:27 +0000 [r205196] Tilghman Lesher + + * /, main/say.c: Merged revisions 205188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) + | 2 lines Add redirection warnings for the invalid language codes + previously removed. ........ + +2009-07-08 15:56 +0000 [r205120-205151] Russell Bryant + + * main/ssl.c: Use tabs instead of spaces for indentation. + + * res/res_crypto.c, main/ssl.c (added), + include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c: + Move OpenSSL initialization to a single place, make library usage + thread-safe. While doing some reading about OpenSSL, I noticed a + couple of things that needed to be improved with our usage of + OpenSSL. 1) We had initialization of the library done in multiple + modules. This has now been moved to a core function that gets + executed during Asterisk startup. We already link OpenSSL into + the core for TCP/TLS functionality, so this was the most logical + place to do it. 2) OpenSSL is not thread-safe by default. + However, making it thread safe is very easy. We just have to + provide a couple of callbacks. One callback returns a thread ID. + The other handles locking. For more information, start with the + "Is OpenSSL thread-safe?" question on the FAQ page of + openssl.org. + +2009-07-08 14:45 +0000 [r205118] Luigi Rizzo + + * bootstrap.sh: FreeBSD now has autoconf 2.62 in the ports, 2.61 + has disappeared. + +2009-07-07 21:10 +0000 [r205086] Tilghman Lesher + + * channels/chan_sip.c: Permit setting custom headers from the peer + definition. (closes issue #14059) Reported by: fnordian + +2009-07-07 18:24 +0000 [r205014-205047] Matthew Nicholson + + * channels/sig_analog.c: Fix a deadlock in sig_analog + + * channels/sig_analog.c: Add CEL transfer events to analog + (chan_dahdi) transfers. + +2009-07-06 21:37 +0000 [r204986] Tilghman Lesher + + * addons/res_config_mysql.c: Merged revisions 981 via svnmerge from + https://origsvn.digium.com/svn/asterisk-addons/branches/1.4 + ........ r981 | tilghman | 2009-07-06 16:30:13 -0500 (Mon, 06 Jul + 2009) | 7 lines Don't reset reconnect time, unless a reconnect + really occurred. (closes issue #15375) Reported by: kowalma + Patches: 20090628__issue15375.diff.txt uploaded by tilghman + (license 14) Tested by: kowalma, jacco ........ + +2009-07-06 13:38 +0000 [r204948] Kevin P. Fleming + + * main/channel.c: Improve handling of AST_CONTROL_T38 and + AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This + change allows applications that request T.38 negotiation on a + channel that does not support it to get the proper indication + that it is not supported, rather than thinking that negotiation + was started when it was not. + +2009-07-03 15:44 +0000 [r204893-204919] Sean Bright + + * channels/sig_pri.h, channels/chan_dahdi.c, configure, + include/asterisk/autoconfig.h.in, configure.ac, + channels/sig_pri.c: Add a configure check for Reverse Charging + Indication support in LibPRI. Also go back and wrap all of the + places that use the specific reverse charge APIs with + preprocessor conditionals. + + * include/asterisk/rtp_engine.h: Wrap rtp_engine.h header comments + to 80 characters. + +2009-07-02 22:01 +0000 [r204835] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 204834 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 + Jul 2009) | 10 lines Removed confusing warning message "Got Busy + in Connected State" If an incoming mISDN call is answered with + the Answer application and a subsequent Dial gets a busy endpoint + then it is valid for that already connected channel to get the + busy indication. Asterisk will play the busy tones until the + dialplan plays something else or hangs up the call. (closes issue + #11974) Reported by: fvdb ........ + +2009-07-02 20:37 +0000 [r204807] Matthew Nicholson + + * main/channel.c, main/features.c: Moved trigger for BRIDGE_END CEL + event so that it is more accurate. + +2009-07-02 17:46 +0000 [r204749] Sean Bright + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, funcs/func_channel.c, CHANGES, + channels/sig_pri.c: Support setting and receiving Reverse + Charging Indication over ISDN PRI. This is a continuation of + revision 885 to LibPRI (Capture and expose the Reverse Charging + Indication IE on ISDN PRI) which added the ability to get/set + Reverse Charging Indication in LibPRI. This patch adds the + ability to specify RCI on the outbound leg of a PRI call from + within Asterisk, by prefixing the dialed number with a capital + 'C' like: ...,Dial(DAHDI/g1/C4445556666) And to read it off an + inbound channel: exten => s,1,Set(RCI=${CHANNEL(reversecharge)}) + Thanks again to rmudgett for the thorough review. (closes issue + #13760) Reported by: mrgabu Review: + https://reviewboard.asterisk.org/r/303/ + +2009-07-02 16:03 +0000 [r204710] David Vossel + + * include/asterisk/devicestate.h, main/pbx.c, /, + main/devicestate.c: Merged revisions 204681 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) + | 14 lines Improved mapping of extension states from combined + device states. This fixes a few issues with incorrect extension + states and adds a cli command, core show device2extenstate, to + display all possible state mappings. (closes issue #15413) + Reported by: legart Patches: exten_helper.diff uploaded by + dvossel (license 671) Tested by: dvossel, legart, amilcar Review: + https://reviewboard.asterisk.org/r/301/ ........ + +2009-07-01 19:47 +0000 [r204654] Ryan Brindley + + * configs/http.conf.sample: - cfgbasic.html has been replaced by + index.html in the GUI for some time now + +2009-07-01 16:06 +0000 [r204622] Sean Bright + + * apps/app_voicemail.c: A bunch of CODING_GUIDELINES related fixes. + Not even close to done. + +2009-06-30 20:41 +0000 [r204563] Tilghman Lesher + + * /, main/say.c, UPGRADE.txt: Merged revisions 204556 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 + Jun 2009) | 6 lines More incorrect language codes, plus ensuring + that regionalizations use the specified language, and not English + for grammar. (closes issue #15022) Reported by: greenfieldtech + Patches: 20090519__issue15022.diff.txt uploaded by tilghman + (license 14) ........ + +2009-06-30 20:39 +0000 [r204561] Sean Bright + + * apps/app_voicemail.c: Remove an unnecessary #ifdef + +2009-06-30 19:59 +0000 [r204530-204532] Mark Michelson + + * channels/chan_sip.c: Move the masquerade in + local_attended_transfer to a point where we hold the channel + lock. Masquerading without the channel's lock held is a + *horrible* idea. + + * channels/chan_sip.c: Remove some bogus deadlock avoidance code + from local_attended_transfer. First of all, the code was + unnecessary. The goal was to lock a channel which was already + locked. Second, the assumption of the deadlock avoidance loop was + that the sip_pvt was already locked and we were trying to get the + channel lock. The problem is that the sip_pvt was unlocked a few + lines above. Basically, I'm removing 5 lines of no-op. + +2009-06-30 18:48 +0000 [r204475] Jason Parker + + * /, main/say.c: Merged revisions 204474 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | + 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a + comment typo in passing. ........ + +2009-06-30 18:36 +0000 [r204470] Tilghman Lesher + + * /, main/say.c, UPGRADE.txt, apps/app_voicemail.c: Recorded merge + of revisions 204469 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) + | 11 lines "tw" is the language specification for Twi (from + Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier + Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman + (license 14) 20090617__issue15346__trunk.diff.txt uploaded by + tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt + uploaded by tilghman (license 14) + 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman + (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by + tilghman (license 14) Tested by: volivier ........ + +2009-06-30 17:22 +0000 [r204417-204440] Russell Bryant + + * configs/res_config_sqlite.conf (removed), + configs/res_config_sqlite.conf.sample (added): Rename + res_config_sqlite.conf to res_config_sqlite.conf.sample (missing + .sample). + + * addons/chan_ooh323.c, configs/chan_ooh323.conf.sample (added), + configs/ooh323.conf.sample (removed): Rename ooh323.conf to + chan_ooh323.conf, make module support both names + + * configs/mobile.conf.sample (removed), addons/chan_mobile.c, + configs/chan_mobile.conf.sample (added): Rename mobile.conf to + chan_mobile.conf, make module support old name, too + + * configs/res_config_mysql.conf.sample (added), + configs/res_mysql.conf.sample (removed), + addons/res_config_mysql.c: Rename res_mysql.conf to + res_config_mysql.conf, make module support both + + * Makefile: Make addons build last - this is for Qwell. + + * addons/app_mysql.c, configs/app_mysql.conf.sample (added), + configs/mysql.conf.sample (removed): Rename mysql.conf to + app_mysql.conf, make module support both names + + * addons/Makefile, addons/cdr_mysql.c (added), + addons/cdr_addon_mysql.c (removed): Rename cdr_addon_mysql to + cdr_mysql + + * addons/app_mysql.c (added), addons/app_addon_sql_mysql.c + (removed), addons/Makefile: Rename app_addon_sql_mysql to + app_mysql + +2009-06-30 17:04 +0000 [r204415] Kevin P. Fleming + + * build_tools/embed_modules.xml, Makefile.moddir_rules, + addons/Makefile: Add-ons related build system improvements. + Ensure that add-on modules can be embedded, fix up + Makefile.moddir_rules to allow module directory Makefiles to more + easily specify the modules to be built, and explicitly list the + addons modules in its Makefile, since the module names don't + follow any pattern. + +2009-06-30 16:40 +0000 [r204413] Russell Bryant + + * autoconf/ast_ext_tool_check.m4, addons/ooh323c/src/oochannels.h, + addons/ooh323c/src/printHandler.h, addons/chan_ooh323.c, + addons/ooh323c/src/ooq931.h, include/asterisk/autoconfig.h.in, + addons/ooh323c/src/ootrace.h, addons/chan_ooh323.h, + addons/ooh323c/src/ooasn1.h, configs/res_mysql.conf.sample + (added), addons/ooh323c/src/ooStackCmds.c, + addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooStackCmds.h, + addons/ooh323c/src/eventHandler.c, + addons/ooh323c/src/h323/H235-SECURITY-MESSAGES.h, + addons/mp3/huffman.h, configure, + addons/ooh323c/src/eventHandler.h, addons/ooh323cDriver.c, + include/asterisk/mod_format.h, addons/mp3/interface.c, + doc/tex/asterisk.tex, addons/ooh323cDriver.h, + addons/cdr_addon_mysql.c, addons/ooh323c/src/encode.c, + addons/mp3/MPGLIB_README, + addons/ooh323c/src/h323/H235-SECURITY-MESSAGESEnc.c, + configure.ac, doc/tex/chan_mobile.tex (added), + addons/ooh323c/src/ooports.c, addons/mp3/mpg123.h, + addons/mp3/mpglib.h, addons (added), + addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROL.c, + addons/ooh323c/src/ooports.h, addons/ooh323c/src/memheap.c, + Makefile, addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROL.h, + addons/ooh323c/src/ooh245.c, addons/mp3/common.c, + addons/ooh323c/src/memheap.h, addons/ooh323c/src/perutil.c, + addons/mp3/decode_i386.c, addons/ooh323c/src/ooh245.h, + addons/mp3/dct64_i386.c, addons/ooh323c/src/ooSocket.c, + addons/ooh323c/src/h323/H235-SECURITY-MESSAGESDec.c, + addons/mp3/layer3.c, addons/ooh323c/src/ooper.h, + addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooSocket.h, + addons/ooh323c/src/ooCommon.h, addons/ooh323c/src/ooCmdChannel.h, + addons/ooh323c/COPYING, addons/format_mp3.c, + addons/ooh323c/src/Makefile.in, configs/mobile.conf.sample + (added), addons/ooh323c/src/ootypes.h, addons/mp3, + addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooTimer.c, + addons/ooh323c/src/ooLogChan.h, addons/ooh323c/src/dlist.c, + addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/oohdr.h, + README-addons.txt (added), addons/app_addon_sql_mysql.c, + addons/ooh323c/src/ooTimer.h, addons/ooh323c/src/ooCapability.h, + addons/ooh323c/src/dlist.h, addons/mp3/Makefile, addons/Makefile, + addons/ooh323c/README, addons/ooh323c, doc/tex/cdrdriver.tex, + addons/ooh323c/src/h323/H323-MESSAGESEnc.c, addons/chan_mobile.c, + configs/cdr_mysql.conf.sample (added), + addons/ooh323c/src/ooDateTime.c, addons/ooh323c/src/rtctype.c, + addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/ooGkClient.c, + addons/ooh323c/src/h323, addons/ooh323c/src/ooUtils.c, + addons/ooh323c/src/ooDateTime.h, + addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLEnc.c, + addons/ooh323c/src/rtctype.h, addons/ooh323c/src/ooCalls.h, + configs/mysql.conf.sample (added), addons/ooh323c/src/ooh323ep.c, + addons/ooh323c/src/ooGkClient.h, + addons/ooh323c/src/h323/H323-MESSAGES.c, + addons/ooh323c/src/ooUtils.h, addons/mp3/README, UPGRADE.txt, + addons/mp3/MPGLIB_TODO, addons/ooh323c/src/ooh323ep.h, + addons/ooh323c/src/h323/H323-MESSAGES.h, + addons/mp3/decode_ntom.c, configs/ooh323.conf.sample (added), + addons/ooh323c/src/ooh323.c, + addons/ooh323c/src/h323/H323-MESSAGESDec.c, addons/ooh323c/src, + build_tools/menuselect-deps.in, addons/mp3/tabinit.c, + addons/ooh323c/src/ooh323.h, doc/tex/Makefile, + addons/ooh323c/src/decode.c, addons/ooh323c/src/context.c, + main/file.c, + addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLDec.c, + makeopts.in, addons/ooh323c/src/oochannels.c, + addons/app_saycountpl.c, addons/ooh323c/src/printHandler.c, + addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ootrace.c, + addons/res_config_mysql.c: Move Asterisk-addons modules into the + main Asterisk source tree. Someone asked yesterday, "is there a + good reason why we can't just put these modules in Asterisk?". + After a brief discussion, as long as the modules are clearly set + aside in their own directory and not enabled by default, it is + perfectly fine. For more information about why a module goes in + addons, see README-addons.txt. chan_ooh323 does not currently + compile as it is behind some trunk API updates. However, it will + not build by default, so it should be okay for now. + +2009-06-29 23:50 +0000 [r204355] Sean Bright + + * apps/app_meetme.c: A few const changes in app_meetme.c that I + noticed while browsing the source. + +2009-06-29 22:50 +0000 [r204247-204301] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 204300 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun + 2009) | 9 lines Add error message so that it is clear why a SIP + peer was not processed when a DNS lookup fails on a host or + outboundproxy. (closes issue #13432) Reported by: p_lindheimer + Patches: outboundproxy.patch uploaded by p (license 558) ........ + + * /, channels/chan_sip.c: Merged revisions 204243,204246 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun + 2009) | 22 lines Fix a problem where chan_sip would ignore "old" + but valid responses. chan_sip has had a problem for quite a long + time that would manifest when Asterisk would send multiple SIP + responses on the same dialog before receiving a response. The + problem occurred because chan_sip only kept track of the highest + outgoing sequence number used on the dialog. If Asterisk sent two + requests out, and a response arrived for the first request sent, + then Asterisk would ignore the response. The result was that + Asterisk would continue retransmitting the requests and ignoring + the responses until the maximum number of retransmissions had + been reached. The fix here is to rearrange the code a bit so that + instead of simply comparing the sequence number of the response + to our latest outgoing sequence number, we walk our list of + outstanding packets and determine if there is a match. If there + is, we continue. If not, then we ignore the response. In doing + this, I found a few completely useless variables that I have now + removed. (closes issue #11231) Reported by: flefoll Review: + https://reviewboard.asterisk.org/r/298 ........ r204246 | + mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 + lines Fix build oops. ........ + +2009-06-29 20:29 +0000 [r204119-204217] Sean Bright + + * configs/cel_adaptive_odbc.conf.sample: Reorganize this adaptive + CEL config a bit. + + * apps/app_rpt.c: Get app_rpt compiling again. I doubt seriously + that it actually works. Also, the code in this module is + horrendous and we should remove it from the tree. I'm not sure + who is supposed to be maintaning this thing, but they clearly are + not. I don't see the sense of leaving it in the main tree. If it + lives *anywhere* it should be in addons. + + * configs/cel_sqlite3_custom.conf.sample, configs/cel.conf.sample, + configs/cel_adaptive_odbc.conf.sample, + configs/cel_pgsql.conf.sample, configs/cel_custom.conf.sample: + Add common headers to CEL related configs. + +2009-06-29 17:56 +0000 [r204069-204118] Tilghman Lesher + + * main/channel.c, include/asterisk/channel.h: Allow trunk to once + again compile under MALLOC_DEBUG + + * configs/cel_adaptive_odbc.conf.sample: Remove invalid entries in + the config. This might seem like a legitimate comment that merely + needed semicolon prefixes, but in reality, the adaptive layer is + designed to allow arbitrary CDR variables, without needing the + use of a userfield to store multiple items. It's therefore not + only invalid syntax but also goes against the intent of the + adaptive method. + +2009-06-27 20:26 +0000 [r203985] Sean Bright + + * CHANGES: Another CHANGES spelling fix. + +2009-06-27 10:04 +0000 [r203960-203962] Russell Bryant + + * main/app.c: Only update total silence counter after a counter + reset. (closes issue #2264) Reported by: pfn Patches: + silent-vm-1.6.2-fix2.txt uploaded by pfn (license 810) Tested by: + pfn + + * UPGRADE.txt, CHANGES: Minor tweaks and spelling fixes for CHANGES + and UPGRADE.txt. + +2009-06-27 01:07 +0000 [r203909] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 203908 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) + | 16 lines The ISDN CPE side should not exclusively pick B + channels normally. Before this patch, Asterisk unconditionally + picked B channels exclusively on the CPE side and normally + allowed alternative B channels on the network side. Now Asterisk + does the opposite. Reasons for the CPE side to normally not pick + B channels exclusively: * For CPE point-to-multipoint mode (i.e. + phone side), the CPE side does not have enough information to + exclusively pick B channels. (There may be other devices on the + line.) * Q.931 gives preference to the network side picking B + channels. * Some telcos require the CPE side to not pick B + channels exclusively. (closes issue #14383) Reported by: + mbrancaleoni ........ + +2009-06-26 22:11 +0000 [r203853] Jeff Peeler + + * channels/chan_dahdi.c, /: Merged revisions 203848 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 + Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo + channel after dahdi restart (closes issue #14477) Reported by: + timking ........ + +2009-06-26 22:08 +0000 [r203846] Sean Bright + + * cdr/cdr_syslog.c (added), build_tools/menuselect-deps.in, + configure, configure.ac, configs/cdr_syslog.conf.sample (added), + CHANGES: Add a new module, cdr_syslog, which allows writing CDRs + to syslog. The original patch for this was written by Brett + Bryant, and I split it out into it's own module. (closes issue + #12876) Reported by: bbryant Patches: + 06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36) + 05212009_cdr_syslog.patch uploaded by seanbright (license 71) + Tested by: seanbright Review: + https://reviewboard.asterisk.org/r/297/ + +2009-06-26 21:48 +0000 [r203802-203842] Russell Bryant + + * CHANGES, apps/app_chanspy.c: Add 's' option to ChanSpy, which + makes the app exit when no channels are left to spy on. (closes + issue #14594) Reported by: JimDickenson Patches: chanspy.diff + uploaded by JimDickenson (license 710) + + * /, main/file.c: Merged revisions 203785 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) + | 15 lines Don't fast forward past the end of a message. This is + nice change for users of the voicemail application. If someone + gets a little carried away with fast forwarding through a + message, they can easily get to the end and accidentally exit the + voicemail application by hitting the fast forward key during the + following prompt. This adds some safety by not allowing a fast + forward past the end of a message. (closes issue #14554) Reported + by: lacoursj Patches: 21761.patch uploaded by lacoursj (license + 707) Tested by: lacoursj ........ + +2009-06-26 20:52 +0000 [r203783] Mark Michelson + + * doc/manager_1_1.txt, main/manager.c: Add timestamp to response to + "Ping" manager action. (closes issue #14596) Reported by: + JimDickenson Patches: pong2.diff uploaded by JimDickenson + (license 710) + +2009-06-26 20:45 +0000 [r203779] Russell Bryant + + * channels/chan_sip.c: Ensure the TCP read buffer is fully + initialized before handling each packet. (closes issue #14452) + Reported by: umberto71 + +2009-06-26 20:19 +0000 [r203735] Joshua Colp + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Fix the + 'nat' option to actually do RFC3581 as expected and extend the + configurable values for finer control. (closes issue #8855) + Reported by: mikma Tested by: klaus3000, file + +2009-06-26 20:13 +0000 [r203721] David Brooks + + * apps/app_voicemail.c: Fixing voicemail's error in checking max + silence vs min message length Max silence was represented in + milliseconds, yet vmminsecs (minmessage) was represented as + seconds. Also, the inequality was reversed. The warning, if + triggered, was "Max silence should be less than minmessage or you + may get empty messages", which should have been logged if max + silence was greater than minmessage, but the check was for less + than. Also, conforming if statement to coding guidelines. closes + issue #15331) Reported by: markd Review: + https://reviewboard.asterisk.org/r/293/ + +2009-06-26 19:47 +0000 [r203710] David Vossel + + * channels/chan_iax2.c: moving debug message from level 0 to 1. + (closes issue #15404) Reported by: leobrown Patches: + iax_codec_debug.patch uploaded by leobrown (license 541) + +2009-06-26 19:31 +0000 [r203702] Russell Bryant + + * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c: + Make invalid hints report Unavailable instead of Idle. (closes + issue #14413) Reported by: pj + +2009-06-26 19:27 +0000 [r203699] Joshua Colp + + * main/channel.c, main/frame.c, main/rtp_engine.c, + channels/chan_sip.c, apps/app_fax.c, configs/sip.conf.sample, + include/asterisk/frame.h: Improve T.38 negotiation by exchanging + session parameters between application and channel. + +2009-06-26 19:03 +0000 [r203672] Jeff Peeler + + * channels/sig_analog.c: Check if polarityonanswerdelay has elapsed + before setting a channel as answered after a polarity reversal. + Previously on a polarity switch event chan_dahdi would set the + channel immediately as answered. This would cause problems if a + polarity reversal occurred when the line was picked up as the + dial would not have yet occurred. Now if the polarity reversal + occurs before delay has elapsed after coming off hook or an + answer, it is ignored. Also, some refactoring was done in + _handle_event. (closes issue #13917) Reported by: alecdavis + Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by + alecdavis (license 585) Tested by: alecdavis + +2009-06-26 15:42 +0000 [r203638-203640] Russell Bryant + + * include/asterisk/doxyref.h, include/asterisk/channel.h: Note a + new API call, and one that changed in doxygen. + + * cel/cel_pgsql.c, configs/cel_sqlite3_custom.conf.sample (added), + cdr/cdr_sqlite3_custom.c, configs/cel.conf.sample (added), + channels/chan_local.c, include/asterisk/cel.h (added), + main/devicestate.c, apps/app_chanisavail.c, channels/chan_iax2.c, + doc/tex/cel-doc.tex (added), main/loader.c, main/cli.c, + channels/chan_dahdi.c, channels/sig_analog.c, + channels/chan_skinny.c, include/asterisk/event_defs.h, + main/features.c, res/ais/evt.c, channels/sig_analog.h, + channels/chan_alsa.c, doc/tex/asterisk.tex, cdr/cdr_manager.c, + apps/app_dial.c, main/pbx.c, include/asterisk/utils.h, + channels/chan_bridge.c, cel/cel_tds.c, channels/chan_agent.c, + configs/cel_adaptive_odbc.conf.sample (added), + include/asterisk/cdr.h, include/asterisk/channel.h, CHANGES, + main/cel.c (added), Makefile, channels/chan_misdn.c, + funcs/func_channel.c, funcs/func_cdr.c, doc/tex/celdriver.tex + (added), main/asterisk.c, cel/cel_adaptive_odbc.c, + apps/app_voicemail.c, res/res_calendar.c, + channels/chan_unistim.c, tests/test_substitution.c, + cel/cel_radius.c, channels/chan_multicast_rtp.c, + channels/chan_vpb.cc, apps/app_meetme.c, channels/chan_gtalk.c, + apps/app_followme.c, configs/cel_tds.conf.sample (added), + main/channel.c, main/cdr.c, channels/chan_phone.c, main/dial.c, + main/manager.c, include/asterisk/event.h, + bridges/bridge_builtin_features.c, funcs/func_odbc.c, + cel/cel_custom.c, cel/cel_manager.c, cdr/cdr_sqlite.c, + res/res_agi.c, apps/app_minivm.c, main/logger.c, + apps/app_confbridge.c, configs/cel_custom.conf.sample (added), + channels/chan_mgcp.c, apps/app_parkandannounce.c, + cdr/cdr_custom.c, channels/chan_sip.c, cel (added), + configs/cel_pgsql.conf.sample (added), channels/chan_console.c, + include/asterisk/_private.h, channels/sig_pri.c, + apps/app_queue.c, channels/chan_oss.c, channels/sig_pri.h, + channels/chan_usbradio.c, channels/chan_jingle.c, cel/Makefile, + apps/app_celgenuserevent.c (added), apps/app_directed_pickup.c, + channels/chan_h323.c, cel/cel_sqlite3_custom.c, main/event.c, + channels/chan_nbs.c: Merge the new Channel Event Logging (CEL) + subsystem. CEL is the new system for logging channel events. This + was inspired after facing many problems trying to represent what + is possible to happen to a call in Asterisk using CDR records. + For more information on CEL, see the built in HTML or PDF + documentation generated from the files in doc/tex/. Many thanks + to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard + work developing this code. Also, thanks to Matt Nicholson + (mnicholson) and Sean Bright (seanbright) for their assistance in + the final push to get this code ready for Asterisk trunk. Review: + https://reviewboard.asterisk.org/r/239/ + +2009-06-26 13:00 +0000 [r203569-203605] Sean Bright + + * include/asterisk/syslog.h, main/syslog.c: Add functions to map + syslog facilities and priorities constants to strings. Also + change the default casing of the string contants to lowercase. + This really just saves us from have to lowercase them later when + displaying them. + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/syslog.c: Add checks in configure for non-POSIX syslog + facilities. + +2009-06-26 00:23 +0000 [r203525-203534] Russell Bryant + + * main/syslog.c: One more formatting nit ... use spaces for inline + indentation. + + * main/syslog.c: Convert spaces to tabs for indentation. + +2009-06-25 23:54 +0000 [r203508] Sean Bright + + * include/asterisk/syslog.h (added), main/logger.c, main/syslog.c + (added): Move syslog utility functions into a separate file so + they can be re-used. This has the pleasant side effect of + cleaning up the header inclusion process in logger.c. + +2009-06-25 22:48 +0000 [r203479] Jeff Peeler + + * channels/chan_dahdi.c: make sure chan_dahdi compiles with only + libss7 and not libpri installed + +2009-06-25 21:45 +0000 [r203444] David Vossel + + * main/ast_expr2.fl, main/ast_expr2.c: fixes a few redundant + conditions (issue #15269) + +2009-06-25 21:34 +0000 [r203443] Richard Mudgett + + * channels/chan_dahdi.c: Picking nits + +2009-06-25 21:22 +0000 [r203402] Jeff Peeler + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Remove + some unnecessary code and update sample config file with respect + to GR-303. + +2009-06-25 21:15 +0000 [r203381] Terry Wilson + + * /, main/cli.c: Merged revisions 203380 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) + | 4 lines I didn't see that Mark already fixed the underlying + issue! Yay for removing useless code. ........ + +2009-06-25 21:04 +0000 [r203376] Russell Bryant + + * /, main/features.c: Merged revisions 203375 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) + | 9 lines Fix a case where CDR answer time could be before the + start time involving parking. (closes issue #13794) Reported by: + davidw Patches: 13794.patch uploaded by murf (license 17) + 13794.patch.160 uploaded by murf (license 17) Tested by: murf, + dbrooks ........ + +2009-06-25 20:25 +0000 [r203338] Terry Wilson + + * /, main/cli.c: Merged revisions 203311 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009) + | 2 lines Don't try to free NULL ........ + +2009-06-25 19:54 +0000 [r203304] Jeff Peeler + + * channels/sig_pri.h (added), channels/chan_dahdi.c, + channels/sig_analog.c, channels/sig_analog.h, channels/sig_pri.c + (added), channels/Makefile: New signaling module to handle + PRI/BRI operations in chan_dahdi This merge splits the PRI/BRI + signaling logic out of chan_dahdi.c into sig_pri.c. Functionality + in theory should not change (mostly). A few trivial changes were + made in sig_analog with verbose messages and commenting. + +2009-06-25 19:22 +0000 [r203258] Jason Parker + + * channels/chan_dahdi.c: Unmute when we get a dtmfup (we muted on + dtmfdown) event. This would occasionally cause one-way audio when + using hardware DTMF detection. (closes issue #14761) Reported by: + tzafrir Patches: v1-14761.patch uploaded by dimas (license 88) + Tested by: tzafrir, dimas + +2009-06-25 18:25 +0000 [r203227] Joshua Colp + + * res/res_rtp_multicast.c (added), channels/chan_multicast_rtp.c + (added), CHANGES: Add support for multicast RTP paging. (closes + issue #11797) Reported by: macbrody Review: + https://reviewboard.asterisk.org/r/270/ + +2009-06-25 17:01 +0000 [r203188] Sean Bright + + * main/logger.c: Pass a logmsg to ast_log_vsyslog instead of + separate arguments. + +2009-06-25 16:18 +0000 [r203126] Doug Bailey + + * channels/chan_dahdi.c: Insure ring cadence is set for fxs ports + Moved SETCADENCE ioctl call to before call into new analog signal + module to insure that it gets set. (closes issue #15381) Reported + by: alecdavis Patches: fix15381.diff uploaded by dbailey (license + 819) Tested by: dbailey + +2009-06-25 16:04 +0000 [r203116] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 203115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) + | 11 lines Resolve a crash related to a T.38 reinvite race + condition. This change resolves a crash observed locally during + some T.38 testing. A call was set up using a call file, and when + the T.38 reinvite came in, the channel state was still + AST_STATE_DOWN. The reason is explained by a comment in the code + that previously lived in the handling of AST_STATE_RINGING. This + change modifies the logic to handle the same race condition for + any channel state that is not UP. (closes ABE-1895) ........ + +2009-06-24 21:08 +0000 [r203037] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 203036 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 + Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error + checking. Valid format is: pritimer=timer_name,timer_value * + Fixed segfault if the ',' is missing. * Completely check the + range returned by pri_timer2idx() to prevent possible access + outside array bounds. ........ + +2009-06-24 18:29 +0000 [r202967] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 202966 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun + 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding + the same thing in-line. ........ + +2009-06-24 18:08 +0000 [r202925] Joshua Colp + + * channels/chan_sip.c: Ensure the default settings are applied for + T.38 when we set it up for a peer. + +2009-06-24 13:53 +0000 [r202840-202889] Sean Bright + + * doc/tex: Ignore some files generated when asterisk.pdf is + created. + + * configs/cdr_tds.conf.sample, cdr/cdr_tds.c: Update sample cdr_tds + configuration to try and eliminate some confusion. Also change + the preferred configuration option from 'hostname' (which was + misleading because it didn't actually treat the value as a + hostname) to 'connection' and added some verbage explaining that + the user would need to refer to their freetds.conf file for those + settings. 'hostname' was kept as a backwards compatible + configuration parameter. + + * doc/tex/billing.tex, doc/tex/cdrdriver.tex: Change some section + names in the CDR tex documentation. + + * doc/tex/cdrdriver.tex: Remove some trailing whitespace before + making content changes. + +2009-06-23 22:47 +0000 [r202804] Russell Bryant + + * doc/tex/cdrdriver.tex: Clean up section hierarchy for the CDR + chapter. + +2009-06-23 22:08 +0000 [r202761] Matthew Fredrickson + + * channels/chan_dahdi.c: I could have sworn I committed this patch + ages ago, but... bug fix with setting NAI properly on linksets in + certain situations. + +2009-06-23 21:38 +0000 [r202755] Richard Mudgett + + * channels/chan_misdn.c: Make outgoing_colp=2 misdn.conf port + parameter not send redirecting or transfer messages. If the + outgoing_colp parameter is set to not send COLP information, then + it does not make sense to send redirecting or transfer messages + announcing new COLP information that is blocked. The service + provider may supply the listed number for that line when it + passes the messages to the next hop. Why tell the switch that + these events happened when the information is otherwise + suppressed? Also blocked the number of previous redirects that + may have occurred to calls going out the port when outgoing_colp + is 2. Follow on to JIRA ABE-1853. + +2009-06-23 21:25 +0000 [r202753] Ryan Brindley + + * main/config.c: If we delete the info, lets also delete the lines + (closes issue #14509) Reported by: timeshell Patches: + 20090504__bug14509.diff.txt uploaded by tilghman (license 14) + Tested by: awk, timeshell + +2009-06-23 16:31 +0000 [r202672] David Vossel + + * /, channels/chan_sip.c: Merged revisions 202671 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) + | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to + non-standard port and transport (closes issue #14659) Reported + by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded + by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded + by dvossel (license 671) Tested by: dvossel, klaus3000 Review: + https://reviewboard.asterisk.org/r/288/ ........ + +2009-06-23 14:54 +0000 [r202497-202570] Russell Bryant + + * main/app.c, CHANGES: Ignore voicemail messages that are just + silence. (closes issue #2264) Reported by: pfn Patches: + silent-vm-1.6.2.txt uploaded by pfn (license 810) + + * main/channel.c, /: Merged revisions 202496 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) + | 4 lines Report CallerID change during a masquerade. Reported + by: markster ........ + +2009-06-22 16:09 +0000 [r202417] Sean Bright + + * cdr/cdr_sqlite3_custom.c: Fix lock usage in cdr_sqlite3_custom to + avoid potential crashes during reload. Pointed out by Russell + while working on the CEL branch. + +2009-06-22 16:05 +0000 [r202415] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 202414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) + | 2 lines Make Polycom subscription type override check more + explicit. ........ + +2009-06-22 15:33 +0000 [r202410] David Vossel + + * include/asterisk/module.h, main/loader.c: attempting to load + running modules Modules placed in the priority heap for loading + were not properly removed from the linked list. This resulted in + some modules attempting to load twice. + +2009-06-22 14:58 +0000 [r202337-202343] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 202341-202342 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun + 2009) | 26 lines Fix a situation in which Asterisk would not stop + retransmitting 487s. If a CANCEL were received by Asterisk, we + would send a 487 in response to the original INVITE and a 200 OK + for the CANCEL. If there were a network hiccup which caused the + 200 OK and the 487 to be lost, then the UA communicating with + Asterisk may try to retransmit its CANCEL. Asterisk's response to + this used to be to try sending another 487 to the canceled INVITE + and another 200 OK to the CANCEL. The problem here is that the + originally-sent 487 was sent "reliably" meaning that it will be + retransmitted until it is received properly. So when we receive + the second CANCEL it is likely that the first batch of 487s we + sent is still going strong and reaches the UA. The result was + that the second set of 487s would be retransmitted constantly + until the maximum number of retries had been reached. The fix for + this is that if we receive a second CANCEL for an INVITE, then we + cancel the retransmission of the first set of 487s and start a + second set. This causes the dialog to be terminated reasonably. + (closes issue #14584) Reported by: klaus3000 Patches: + 14584_v2.patch uploaded by mmichelson (license 60) Tested by: + klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58 + -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line + left from previous commit. ........ + + * /, channels/chan_sip.c: Merged revisions 202336 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun + 2009) | 25 lines Fix a possible infinite loop in SDP parsing + during glare situation. There was a while loop in + get_ip_and_port_from_sdp which was controlled by a call to + get_sdp_iterate. The loop would exit either if what we were + searching for was found or if the return was NULL. The problem is + that get_sdp_iterate never returns NULL. This means that if what + we were searching for was not present, the loop would run + infinitely. This modification of the loop fixes the problem. + (closes issue #15213) Reported by: schmidts (closes issue #15349) + Reported by: samy (closes issue #14464) Reported by: pj (closes + issue #15345) Reported by: aragon Patches: sip_inf_loop.patch + uploaded by mmichelson (license 60) Tested by: aragon ........ + +2009-06-21 16:36 +0000 [r202223-202301] Russell Bryant + + * cdr/cdr_sqlite3_custom.c: Note a bug in cdr_sqlite3_custom so I + don't forget about it. + + * cdr/cdr_manager.c: Fix possibility of crashiness during reload in + custom fields handling. + + * cdr/cdr_manager.c: Standardize return values of load_config() so + reload() doesn't report an error on success. + + * cdr/cdr_manager.c: Leave a note about some unsafe code in + cdr_manager + +2009-06-20 19:09 +0000 [r202183] Sean Bright + + * apps/app_fax.c: Fix version detection for API changes in spandsp. + (closes issue #15355) Reported by: deuffy + +2009-06-20 14:09 +0000 [r202109] Russell Bryant + + * main/cdr.c, cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c: Remove + unnecessary usleep() from a couple of module unload callbacks. In + passing, also tweak cdr_unregister() to hold the list lock a bit + less time. + +2009-06-19 21:25 +0000 [r202039] Matthew Nicholson + + * channels/chan_sip.c: Use sched_yield() instead of usleep(1) + +2009-06-19 20:24 +0000 [r201994] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 201993 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 + Jun 2009) | 8 lines timestamp was being converted to host order + as a short rather than a long (closes issue #15361) Reported by: + ffloimair Patches: ts_issue.diff uploaded by dvossel (license + 671) ........ + +2009-06-19 17:40 +0000 [r201944] Terry Wilson + + * CHANGES: Add note about the addition of calendar support + +2009-06-19 15:47 +0000 [r201904] Tilghman Lesher + + * res/res_config_odbc.c: Fix 2 typos and add support for wide + character types. Reported by Benny Amorsen via the asterisk-users + mailing list. + http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html + +2009-06-19 15:41 +0000 [r201902] Joshua Colp + + * main/rtp_engine.c, channels/chan_sip.c, + include/asterisk/rtp_engine.h: Add support for allowing an RTP + engine to decide on whether it is possible for specific formats + to be transcoded for an RTP instance. + +2009-06-19 00:43 +0000 [r201745-201829] Tilghman Lesher + + * /, main/features.c: Merged revisions 201828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) + | 6 lines If the "h" extension fails, give it another chance in + main/pbx.c. If the "h" extension fails, give it another chance in + main/pbx.c, when it returns from the bridge code. Fixes an issue + where the "h" extension may occasionally not fire, when a Dial is + executed from a Macro. Debugged in #asterisk with user tompaw. + ........ + + * apps/Makefile: One of the changes in 1.6.1 was to allow + app_directory to use functionality within app_voicemail for + directory functions. It is therefore no longer necessary for + app_directory to be linked against the ODBC libraries (and it + never was necessary for app_directory to be linked against IMAP, + though it was). + + * funcs/func_cut.c: Clarify CUT code, and in the process, fix a bug + in trunk only (closes issue #15320) Reported by: chappell + Patches: cut_fix.patch uploaded by chappell (license 8) + cut_clarify.patch uploaded by chappell (license 8) + +2009-06-18 17:41 +0000 [r201717] Matthew Nicholson + + * channels/chan_sip.c: Added deadlock protection to + try_suggested_sip_codec in chan_sip.c. Review: + https://reviewboard.asterisk.org/r/285/ + +2009-06-18 16:37 +0000 [r201678] David Vossel + + * codecs/gsm/src/gsm_destroy.c, channels/h323/ast_h323.cxx, + main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c, + utils/extconf.c, channels/xpmr/xpmr.c, pbx/pbx_config.c, + res/res_config_ldap.c, apps/app_rpt.c, channels/misdn/isdn_lib.c, + main/asterisk.c, utils/conf2ael.c, main/ast_expr2.c, + utils/stereorize.c: fixes some memory leaks and redundant + conditions (closes issue #15269) Reported by: contactmayankjain + Patches: patch.txt uploaded by contactmayankjain (license 740) + memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) + Tested by: contactmayankjain, dvossel + +2009-06-18 15:27 +0000 [r201610] Russell Bryant + + * /, res/res_musiconhold.c: Merged revisions 201600 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 + Jun 2009) | 29 lines Fix memory corruption and leakage related + reloads of non files mode MoH classes. For Music on Hold classes + that are not files mode, meaning that we are executing an + application that will feed us audio data, we use a thread to + monitor the external application and read audio from it. This + thread also makes use of the MoH class object. In the MoH class + destructor, we used pthread_cancel() to ask the thread to exit. + Unfortunately, the code did not wait to ensure that the thread + actually went away. What needed to be done is a pthread_join() to + ensure that the thread fully cleans up before we proceed. By + adding this one line, we resolve two significant problems: 1) + Since the thread was never joined, it never fully goes away. So, + on every reload of non-files mode MoH, an unused thread was + sticking around. 2) There was a race condition here where the + application monitoring thread could still try to access the MoH + class, even though the thread executing the MoH reload has + already destroyed it. (issue #15109) Reported by: jvandal (issue + #15123) Reported by: axisinternet (issue #15195) Reported by: + amorsen (issue AST-208) ........ + +2009-06-18 15:20 +0000 [r201583] Mark Michelson + + * res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c, + include/asterisk/rtp_engine.h: Trunk implementation of setting an + alternate RTP source. This contains the interface by which we can + let an rtp instance know that it might start receiving audio from + a new source. This is similar in nature to revision 197588 of + Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276 + +2009-06-18 15:16 +0000 [r201534-201570] David Vossel + + * channels/chan_sip.c: parsing extension correctly from sip + register lines If a transport type was specified, but no + extension, parsing of the extension would return whatever was + after the transport rather than defaulting to 's'. (closes issue + #15111) Reported by: ffs Patches: + chan_sip.c_register-parser.patch uploaded by ffs (license 730) + Tested by: ffs, dvossel + + * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Add + rtsavesysname to chan_iax chan_sip has an option to save the + sysname on rtupdate. This patch copies that same logic to + chan_iax. (closes issue #14837) Reported by: barthpbx Patches: + iax2-rtsavesysname.patch uploaded by barthpbx (license 744) + rt_iax.diff uploaded by dvossel (license 671) + +2009-06-17 21:31 +0000 [r201531] Tilghman Lesher + + * apps/app_voicemail.c: Initialize additional variables, to prevent + a possible crash. (closes issue #15186) Reported by: ajohnson + Patches: 20090528__issue15186.diff.txt uploaded by tilghman + (license 14) Tested by: ajohnson + +2009-06-17 20:10 +0000 [r201458-201462] Mark Michelson + + * channels/chan_sip.c: Fix problem with no audio due to ignoring + the SDP. A recent change to our SDP version comparison made audio + not function on some calls. This was because of a test wherein we + were trying to see if an unsigned value was less than 0. This is + a dumb comparison and arguably the compiler should have warned + about it. Alas, though, it slipped past. Now it's fixed by + changing the variable to be a signed type. Found by several + developers. Tested by mnicholson and dbrooks. + + * main/channel.c, /: Merged revisions 201450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun + 2009) | 9 lines Change the datastore traversal in + ast_do_masquerade to use a safe list traversal. It is possible + for datastore fixup functions to remove the datastore from the + list and free it. In particular, the queue_transfer_fixup in + app_queue does this. While I don't yet know of this causing any + crashes, it certainly could. Found while discussing a separate + issue with Brian Degenhardt. ........ + +2009-06-17 20:00 +0000 [r201445-201453] David Vossel + + * doc/datastores.txt: ast_channel_datastore_alloc is no longer + used. updating datastores.txt to reflect that. + + * /, apps/app_mixmonitor.c: Merged revisions 201423 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 + Jun 2009) | 19 lines StopMixMonitor race condition (not giving up + file immediately) StopMixMonitor only indicates to the MixMonitor + thread to stop writing to the file. It does not guarantee that + the recording's file handle is available to the dialplan + immediately after execution. This results in a race condition. To + resolve this, the filestream pointer is placed in a datastore on + the channel. When StopMixMonitor is called, the datastore is + retrieved from the channel and the filestream is closed + immediately before returning to the dialplan. Documentation + indicating the use of StopMixMonitor to free files has been + updated as well. (closes issue #15259) Reported by: travisghansen + Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/283/ ........ + +2009-06-17 19:15 +0000 [r201381] David Brooks + + * /, channels/chan_sip.c: Merged revisions 201380 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) + | 9 lines Checks for NULL sip_pvt pointer in + chan_sip.c->acf_channel_read() Zombie channels could be passed, + and chan_sip.c wasn't checking for it. Could crash Asterisk. Now + checking for NULL pointer. (closes issue #15330) Reported by: + okrief Tested by: dbrooks ........ + +2009-06-17 15:20 +0000 [r201331-201344] David Vossel + + * channels/chan_sip.c: SIP registry ref count error During a sip + reload, the list of sip_registry objects are supposed to be + traversed, unlinked, and destroyed, but destruction never takes + place due to a ref counting error. This causes a memory leak when + registry items are removed from sip.conf and reloaded. While the + registries are removed from the global list, they are not removed + from the scheduler. Because of this, SIP register attempts + continue to be sent out for the item even though it may no longer + be in the .conf. (closes issue #15295) Reported by: amorsen + Review: https://reviewboard.asterisk.org/r/282/ + + * channels/chan_iax2.c: update chan_iax to use 64bit feature flags. + (closes issue #15335) Reported by: lmadsen Review: + https://reviewboard.asterisk.org/r/284/ + +2009-06-17 12:04 +0000 [r201262] Kevin P. Fleming + + * /, include/asterisk/linkedlists.h: Merged revisions 201261 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun + 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list + to be appended is empty. When the list to be appended is empty, + and the list to be appended to is *not*, AST_LIST_APPEND_LIST + would actually cause the target list to become broken, and no + longer have a pointer to its last entry. This patch fixes the + problem. (reported by Stanislaw Pitucha on the asterisk-dev + mailing list) ........ + +2009-06-16 22:29 +0000 [r201223] David Vossel + + * channels/chan_sip.c: fix issue with build_contact introduced by + the "SIP trasnport type issues" commit + +2009-06-16 22:11 +0000 [r201190] Sean Bright + + * CREDITS: Update my e-mail address (thanks for the props, russell + :)) + +2009-06-16 21:10 +0000 [r200985-201139] Kevin P. Fleming + + * channels/chan_dahdi.c, channels/chan_sip.c, apps/app_fax.c, + include/asterisk/frame.h: Enable applications to enable/disable + digit and tone detection. Some applications (notably app_fax) do + not need digit detection nor FAX tone detection while they are + running, and if Asterisk is using software DSPs to provide the + detection, this consumes extra CPU cycles that could be better + spent on the actual application. This patch allows applications + to query and control the state of digit and tone detection on a + channel, and modifies app_fax to disable them while the FAX + operations are occurring (and re-enable digit detection + afterwards). + + * configure, configure.ac: Explicitly test for 'static weakref' + support. Since we use 'static' weakref symbols, and not all GCC + versions support them, test for that combination explicitly. + + * Makefile: When compiling in an Emacs-spawned shell, always print + directory names. This change ensures that Emacs can find the + proper source files when parsing compiler error messages, since + it uses the 'make' output including directory names to do it. + + * configure, autoconf/ast_gcc_attribute.m4, configure.ac: Another + minor fix to compiler attribute checking. Defaulting to 'static' + for the function scope was bad... so remove it. + + * main/channel.c, main/autoservice.c, main/frame.c, /, + apps/app_meetme.c, main/slinfactory.c, + include/asterisk/linkedlists.h, main/file.c, + include/asterisk/channel.h, include/asterisk/frame.h, + apps/app_chanspy.c, apps/app_mixmonitor.c: Merged revisions + 200991 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun + 2009) | 11 lines Improve support for media paths that can + generate multiple frames at once. There are various media paths + in Asterisk (codec translators and UDPTL, primarily) that can + generate more than one frame to be generated when the application + calling them expects only a single frame. This patch addresses a + number of those cases, at least the primary ones to solve the + known problems. In addition it removes the broken TRACE_FRAMES + support, fixes a number of bugs in various frame-related API + functions, and cleans up various code paths affected by these + changes. https://reviewboard.asterisk.org/r/175/ ........ + + * configure, autoconf/ast_gcc_attribute.m4, configure.ac: Fix + problems with new compiler attribute checking in configure + script. The last changes to ast_gcc_attribute.m4 caused some + problems checking for various attributes, because the scope of + the symbol the attribute is applied to can be important; this + patch allows the scope to be specified for the check. + +2009-06-16 16:03 +0000 [r200946] David Vossel + + * channels/chan_sip.c: SIP transport type issues What this patch + addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP + address/port reguardless if the sip->pvt is of type UDP or not. + Now when no remapping is required, ast_sip_ouraddrfor() checks + the sip_pvt's transport type, attempting to set the address and + port to the correct TCP/TLS bindings if necessary. 2. It is not + necessary to send the port number in the Contact header unless + the port is non-standard for the transport type. This patch fixes + this and removes the todo note. 3. In sip_alloc(), the default + dialog built always uses transport type UDP. Now sip_alloc() + looks at the sip_request (if present) and determines what + transport type to use by default. 4. When changing the transport + type of a sip_socket, the file descriptor must be set to -1 and + in some cases the tcptls_session's ref count must be decremented + and set to NULL. I've encountered several issues associated with + this process and have created a function, set_socket_transport(), + to handle the setting of the socket type. (closes issue #13865) + Reported by: st Patches: dont_add_port_if_tls.patch uploaded by + Kristijan (license 753) 13865.patch uploaded by mmichelson + (license 60) tls_port_v5.patch uploaded by vrban (license 756) + transport_issues.diff uploaded by dvossel (license 671) Tested + by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: + https://reviewboard.asterisk.org/r/278/ + +2009-06-16 15:51 +0000 [r200943] Michiel van Baak + + * apps/app_voicemail.c: add FILE_STORAGE to Voicemail Build Options + Voicemail can only use one storage module at the moment. Because + it's unclear that selecting one of the storage modules in + menuselect will disable filesystem storage we now have a + FILE_STORAGE option that conflicts with the other modules. + (closes issue #15333) + +2009-06-16 15:26 +0000 [r200942] Russell Bryant + + * CREDITS: Add Sean Bright to CREDITS - Thanks, Sean! + +2009-06-16 14:12 +0000 [r200841-200878] Eliel C. Sardanons + + * /: Recorded merge of revisions 200875 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r200875 | eliel | 2009-06-16 09:25:51 -0400 (Tue, 16 Jun 2009) | + 5 lines Show the interface name on error, if it is not found. If + the smdiport specified is not found, show the interface name + instead of '(null)'. ........ + + * res/res_smdi.c: Show the interface name on error, if it is not + found. If the smdiport specified is not found, show the interface + name instead of '(null)'. + +2009-06-16 02:32 +0000 [r200805] Russell Bryant + + * main/manager.c: Don't claim a char * is a mansession object. + Since there was only 1 bucket, and no hash function was + specified, the code actually worked perfectly fine. However, in + theory, this was invalid use of the OBJ_POINTER flag, so remove + it so the code provides a better usage example. + +2009-06-16 02:24 +0000 [r200799] Moises Silva + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: keep + backwards compatible chan_dahdi with older openr2 versions by not + using the new skip category feature unless supported + +2009-06-16 01:28 +0000 [r200764] Kevin P. Fleming + + * configure, autoconf/ast_gcc_attribute.m4: Ensure that + configure-script testing for compiler attributes actually works. + The configure script tests for compiler attributes didn't + actually enable enough warnings or provide a proper test harness + to determine whether the compiler supports the attribute in + question or not; this caused gcc 4.1 to report that it supports + 'weakref', but it doesn't actually support it in the way that is + needed for our optional API mechanism. The new configure script + test will properly distinguish between full support and partial + support for this attribute, among others. + +2009-06-16 01:26 +0000 [r200762] Russell Bryant + + * doc/tex/channelvariables.tex: Add missing closure of verbatim + environment. + +2009-06-16 01:03 +0000 [r200519-200726] Kevin P. Fleming + + * CHANGES: Document the new automatic 'ignoresdpversion' behavior. + Asterisk will now automatically ignore incorrect incoming SDP + version numbers when necessary to complete a T.38 re-INVITE + operation. + + * channels/chan_sip.c: Accept T.38 re-INVITE responses with invalid + SDP versions. This commit changes the 'incoming SDP version' + check logic a bit more; when 'ignoresdpversion' is *not* set for + a peer, if we initiate a re-INVITE to switch to T.38, we'll + always accept the peer's SDP response, even if they don't + properly increment the SDP version number as they should. If this + situation occurs, a warning message will be generated suggesting + that the peer's configuration be changed to include the + 'ignoresdpversion' configuration option (although ideally they'd + fix their SIP implementation to be RFC compliant). AST-221 + + * doc/CODING-GUIDELINES, apps/app_read.c, apps/app_page.c, + apps/app_fax.c, apps/app_readexten.c, apps/app_queue.c, + include/asterisk/app.h, apps/app_skel.c, apps/app_minivm.c, + apps/app_macro.c, apps/app_url.c, apps/app_sms.c, + apps/app_externalivr.c, apps/app_stack.c, apps/app_mixmonitor.c, + apps/app_voicemail.c: Last batch of 'static' qualifiers for + module-level global variables. Fix up modules in the 'apps' + directory, and also correct the bad example of enum definitions + in include/asterisk/app.h, which many developers followed (thanks + for reading the documentation!). In addition, add some basic + usage examples of the 'pahole' and 'pglobal' tools to the coding + guidelines. + + * res/res_snmp.c, main/devicestate.c, funcs/func_vmcount.c, + res/res_calendar_caldav.c, formats/format_wav_gsm.c, + res/res_jabber.c, main/loader.c, main/cli.c, funcs/func_enum.c, + main/manager.c, res/res_smdi.c, funcs/func_odbc.c, + main/features.c, main/logger.c, main/http.c, pbx/pbx_realtime.c, + main/image.c, main/db.c, cdr/cdr_manager.c, + res/res_calendar_exchange.c, res/res_calendar_icalendar.c, + res/res_config_pgsql.c, funcs/func_lock.c, pbx/pbx_lua.c, + funcs/func_cut.c, include/asterisk/calendar.h, + funcs/func_realtime.c, funcs/func_curl.c, funcs/func_cdr.c, + funcs/func_channel.c, main/file.c, main/event.c, pbx/pbx_dundi.c, + main/xmldoc.c, res/res_calendar.c: More 'static' qualifiers on + module global variables. The 'pglobal' tool is quite handy indeed + :-) + + * channels/chan_dahdi.c, channels/chan_misdn.c, + channels/chan_sip.c, channels/chan_skinny.c, + channels/chan_agent.c, channels/chan_h323.c, + channels/chan_iax2.c: Convert a number of global module variables + to 'static'. These modules all contained variables that are + module-global but not system-global, but were not marked + 'static'. + + * channels/chan_sip.c: Some minor structure size improvements in + sip_pvt and sip_peer. Using the 'pahole' tool, it is now quite + easy to see where structure fields could be organized differently + to keep the compiler from having to add padding to satisfy + alignment requirements. These changes reduced the sizes of + sip_pvt and sip_peer by a few bytes each (on 64-bit platforms), + and also fixed a spelling error in a field name. + + * include/asterisk/agi.h, main/Makefile, + include/asterisk/autoconfig.h.in, res/res_smdi.exports, + configure.ac, res/res_agi.exports, include/asterisk/compiler.h, + apps/app_queue.c, res/res_monitor.c, + include/asterisk/optional_api.h, Makefile, res/res_smdi.c, + configure, res/res_agi.c, include/asterisk/monitor.h, + apps/app_stack.c, include/asterisk/smdi.h, + res/res_monitor.exports, apps/app_voicemail.c: Redesigned + 'optional API' support. This patch provides a new implementation + of the optional API support defined in asterisk/optional_api.h; + this new version provides solves compatibility issues with the + use of linker version scripts for suppressing global symbols. In + addition, there is now a functional (and tested!) implementation + for Mac OS/X, so module writers no longer need to use special + tests before calling optional API functions. All future + implementations must provide these same semantics, so that module + writers can rely on them. + +2009-06-15 15:22 +0000 [r200514] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 200513 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun + 2009) | 5 lines Add INFO to our allowed methods so that endpoints + know they may send it to us. AST-223 ........ + +2009-06-14 06:13 +0000 [r200477] Moises Silva + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + build_tools/menuselect-deps.in: added openr2 to + menuselect-deps.in, recent commit in menuselect made me realize + this was never done but was working anyways also added support + for skip category request feature of openr2 and updated + chan_dahdi.conf.sample + +2009-06-12 19:46 +0000 [r200428-200430] Sean Bright + + * contrib/upstart/asterisk.upstart-0.3.9: Include basic + installation and usage instructions for upstart script. + + * contrib/upstart/asterisk.upstart-0.3.9 (added), contrib/upstart + (added): First shot at an upstart script for asterisk on Ubuntu. + This works relatively well (assuming you are using + /var/run/asterisk) as your run directory and upstart 0.3.9. Needs + to be generalized and eventually added to the 'make install' + target for Ubuntu. + +2009-06-12 19:07 +0000 [r200290-200361] Mark Michelson + + * main/channel.c, /: Merged revisions 200360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun + 2009) | 10 lines Suppress a warning message and give a better + return code when generating inband ringing after a call is + answered. (closes issue #15158) Reported by: madkins Patches: + 15158.patch uploaded by mmichelson (license 60) Tested by: + madkins ........ + + * channels/chan_local.c, apps/app_queue.c: Fix some bad locking + stemming from trying to forward a call to a non-existent + extension from a queue. + + * apps/app_queue.c: Fix a potential crash from trying to access a + NULL channel pointer. + +2009-06-12 02:20 +0000 [r200254] Sean Bright + + * contrib/init.d/rc.debian.asterisk: Call chgrp instead of chown + when setting run directory group ownership. (issue #13153) + Reported by: pabelanger + +2009-06-11 21:17 +0000 [r200146] Mark Michelson + + * channels/chan_sip.c: Fix a crash due to a potentially NULL + p->options. Thanks to mnicholson for pointing it out. + +2009-06-11 15:40 +0000 [r200108] Eliel C. Sardanons + + * main/channel.c: Release the allocated channel decreasing the + reference counter. When allocating the channel use ao2_ref(-1) to + release it, instead of calling ast_free(). Also avoid freeing + structures inside that channel (on error) if they will be + released by the channel destructor being called if the reference + counter reachs 0. + +2009-06-11 12:15 +0000 [r200039] Leif Madsen + + * build_tools/make_version_c, build_tools/make_version_h: Fix path + for .flavor and .version (issue #14737) Reported by: davidw + Patches: flavor.patch uploaded by davidw (license 780) Tested by: + davidw + +2009-06-10 20:40 +0000 [r200000] Sean Bright + + * sample.call: Remove some trailing whitespace and steal revision + 200000. + +2009-06-10 20:15 +0000 [r199958] Mark Michelson + + * channels/chan_sip.c: Only try to use the invite_branch on + outgoing INVITEs with auth credentials. I have added a comment to + the code to help ease understanding of the logic here as well. + +2009-06-10 20:00 +0000 [r199957] David Brooks + + * main/pbx.c: Fixes the argument order in definition of + new_find_extension(). In the definition of new_find_extension(), + the arguments 'callerid' and 'label' were swapped. The prototype + declaration and all calls to the function are ordered 'callerid' + then 'label', but the function itself was ordered 'label' then + 'callerid'. (closes issue #15303) Reported by: JimDickenson + +2009-06-10 18:58 +0000 [r199923] Mark Michelson + + * main/channel.c: Use ast_channel_unref to instead of ast_free on a + newly created channel. Also I removed an unnecessary free of a + cid_name. This will be freed properly in the channel destructor. + Reported by mnicholson in #asterisk-dev. + +2009-06-10 16:10 +0000 [r199857] Sean Bright + + * include/asterisk/utils.h, /: Merged revisions 199856 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, + 10 Jun 2009) | 2 lines __WORDSIZE is not available on all + platforms, so use sizeof(void *) instead. ........ + +2009-06-09 20:47 +0000 [r199818] David Vossel + + * channels/chan_sip.c: CLI NOTIFY sending wrong transport type. + SIP's cli NOTIFY command only used UDP rather than copying the + transport type from the peer. (closes issue #15283) Reported by: + jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by + jthurman (license 614) Tested by: jthurman, dvossel + +2009-06-09 18:08 +0000 [r199781] Sean Bright + + * Makefile: Fix all of the parallel build warnings issued when + running make -j#. + +2009-06-09 16:22 +0000 [r199743] David Vossel + + * res/res_timing_pthread.c, include/asterisk/module.h, + res/res_timing_dahdi.c, res/res_timing_timerfd.c, main/loader.c: + module load priority This patch adds the option to give a module + a load priority. The value represents the order in which a + module's load() function is initialized. The lower the value, the + higher the priority. The value is only checked if the + AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER + flag is not set, the value will never be read and the module will + be given the lowest possible priority on load. Since some modules + are reliant on a timing interface, the timing modules have been + given a high load priorty. (closes issue #15191) Reported by: + alecdavis Tested by: dvossel Review: + https://reviewboard.asterisk.org/r/262/ + +2009-06-08 22:08 +0000 [r199696] Tilghman Lesher + + * doc/janitor-projects.txt: Add sigaction janitor + +2009-06-08 19:33 +0000 [r199630] Sean Bright + + * include/asterisk/utils.h, /: Merged revisions 199626,199628 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun + 2009) | 21 lines Increase the size of our thread stack on 64 bit + processors. We were setting the stack size for each thread to + 240KB regardless of architecture, which meant that in some + scenarios we actually had less available stack space on 64 bit + processors (pointers use 8 bytes instead of 4). So now we + calculate the stack size we reserve based on the platform's + __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 + bit -> 1008KB (that's right, we're ready for 128 bit processors) + Patch typed by me but written by several members of + #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes + issue #14932) Reported by: jpiszcz Patches: + 06052009_issue14932.patch uploaded by seanbright (license 71) + Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 + 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the + stack size calculation just introduced. ........ + +2009-06-08 17:32 +0000 [r199588] Mark Michelson + + * channels/chan_sip.c: Fix a deadlock that could occur when setting + rtp stats on SIP calls. (closes issue #15143) Reported by: + cristiandimache Patches: 15143.patch uploaded by mmichelson + (license 60) Tested by: cristiandimache + +2009-06-07 19:15 +0000 [r199514-199547] Eliel C. Sardanons + + * apps/app_osplookup.c: Move OSP* applications static documentation + to XML. Move OSP* applications static documentation to the new + AstXML form. (closes issue #15245) Reported by: eliel Patches: + app_osplookup_static_conversion.txt uploaded by lmadsen (license + 10) + + * apps/app_externalivr.c: Move application ExternalIVR static + documentation to XML. Move application ExternalIVR static + documentation to the new AstXML form. (issue #15245) Reported by: + eliel Patches: app_externalivr.diff uploaded by eliel (license + 64) + +2009-06-07 14:55 +0000 [r199479] Russell Bryant + + * apps/app_dial.c, apps/app_dahdibarge.c, apps/app_dictate.c, + apps/app_authenticate.c, apps/app_echo.c, apps/app_fax.c, + apps/app_dahdiras.c, apps/app_disa.c, apps/app_alarmreceiver.c, + apps/app_chanisavail.c, apps/app_exec.c, apps/app_db.c, + apps/app_controlplayback.c, apps/app_channelredirect.c, + apps/app_directed_pickup.c, apps/app_dumpchan.c, apps/app_amd.c, + apps/app_confbridge.c, apps/app_directory.c, apps/app_chanspy.c, + apps/app_adsiprog.c: Global var cleanup - constification and + removing unused vars. + +2009-06-06 23:28 +0000 [r199374-199446] Eliel C. Sardanons + + * apps/app_stack.c: Move AGI command 'gosub' static documentation + to XML. Move AGI command 'gosub' statis documentation to the new + AstXML form. (issue #15245) Reported by: eliel Patches: + app_stack_static_conversion.txt uploaded by lmadsen (license 10) + (with minor changes by me) + + * res/res_musiconhold.c: Move music on hold related applications + documentation to XML. Move MusicOnHold, SetMusicOnHold, + StartMusicOnHold, StopMusicOnHold static documentation to the new + AstXML form. (issue #15245) Reported by: eliel Patches: + res_musiconhold_static_conversion.txt uploaded by lmadsen + (license 10) (with some fixes and formatting by me) + + * res/res_phoneprov.c: Move function PP_EACH_USER and + PP_EACH_EXTENSION documentation to XML. Move function + PP_EACH_USER and PP_EACH_EXTENSION documentation to the new + AstXML form. (issue #15245) Reported by: eliel Patches: + res_phoneprov_static_conversion.txt uploaded by lmadsen (license + 10) (with PP_EACH_USER add by me) + + * apps/app_meetme.c: Move function MEETME_INFO documentation to + XML. Move function MEETME_INFO static documentation to the new + AstXML form. (issue #15245) Reported by: eliel Patches: + app_meetme_static_conversion.txt uploaded by lmadsen (license 10) + + * apps/app_minivm.c: Move function MINIVMACCOUNT and MINIVMCOUNTER + static documentation to XML. Move function MINIVMACCOUNT and + MINIVMCOUNTER statis documentation to the new AstXML form. (issue + #15245) Reported by: eliel Patches: + app_minivm_static_conversion.txt uploaded by lmadsen (license 10) + (with minor changes by me) + + * funcs/func_sysinfo.c: Move function SYSINFO documentation to XML. + Move function SYSINFO static documentation to the new AstXML + form. (issue #15245) Reported by: eliel Patches: + func_sysinfo_static_conversion.txt uploaded by lmadsen (license + 10) + +2009-06-06 21:42 +0000 [r199368-199372] Russell Bryant + + * apps/app_jack.c: minor tweak + + * apps/app_jack.c: Constify a string and strip trailing whitespace. + + * Makefile: Switch from "echo -n" to printf. On my mac, the -n was + just getting printed out. + +2009-06-05 21:21 +0000 [r199298] David Vossel + + * include/asterisk/devicestate.h, /, main/devicestate.c: Merged + revisions 199297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) + | 14 lines Fixes issue with hints giving unexpected results. + Hints with two or more devices that include ONHOLD gave + unexpected results. (closes issue #15057) Reported by: + p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel + (license 671) pbx.c.1.4.patch uploaded by p (license 558) + devicestate.c.trunk.patch uploaded by p (license 671) Tested by: + p_lindheimer, dvossel Review: + https://reviewboard.asterisk.org/r/254/ ........ + +2009-06-05 13:51 +0000 [r199227] Mark Michelson + + * channels/chan_dahdi.c: Correct "dahdi show channels" output when + specifying a group. Since a DAHDI channel may belong to multiple + groups, we need to use a bitwise and instead of equivalence to + determine whether to display the channel information. (closes + issue #15248) Reported by: gentian Patches: 15248.patch uploaded + by mmichelson (license 60) Tested by: gentian + +2009-06-04 19:10 +0000 [r199139] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 199138 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 + Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ + +2009-06-04 16:29 +0000 [r199091] Eliel C. Sardanons + + * res/res_smdi.c: Move static docs to the new AstXML form. Move + SMDI_MSG and SMDI_MSG_RETRIEVE functions statis documentation to + XML. (issue #15245) Reported by: eliel Patches: + res_smdi_static_conversion.txt uploaded by lmadsen (license 10) + +2009-06-04 14:31 +0000 [r199051] Sean Bright + + * /, include/asterisk/_private.h, main/asterisk.c, main/loader.c: + Merged revisions 199022 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun + 2009) | 40 lines Safely handle AMI connections/reload requests + that occur during startup. During asterisk startup, a lock on the + list of modules is obtained by the primary thread while each + module is initialized. Issue 13778 pointed out a problem with + this approach, however. Because the AMI is loaded before other + modules, it is possible for a module reload to be issued by a + connected client (via Action: Command), causing a deadlock. The + resolution for 13778 was to move initialization of the manager to + happen after the other modules had already been lodaded. While + this fixed this particular issue, it caused a problem for users + (like FreePBX) who call AMI scripts via an #exec in a + configuration file (See issue 15189). The solution I have come up + with is to defer any reload requests that come in until after the + server is fully booted. When a call comes in to ast_module_reload + (from wherever) before we are fully booted, the request is added + to a queue of pending requests. Once we are done booting up, we + then execute these deferred requests in turn. Note that I have + tried to make this a bit more intelligent in that it will not + queue up more than 1 request for the same module to be reloaded, + and if a general reload request comes in ('module reload') the + queue is flushed and we only issue a single deferred reload for + the entire system. As for how this will impact existing + installations - Before 13778, a reload issued before module + initialization was completed would result in a deadlock. After + 13778, you simply couldn't connect to the manager during startup + (which causes problems with #exec-that-calls-AMI configuration + files). I believe this is a good general purpose solution that + won't negatively impact existing installations. (closes issue + #15189) (closes issue #13778) Reported by: p_lindheimer Patches: + 06032009_15189_deferred_reloads.diff uploaded by seanbright + (license 71) Tested by: p_lindheimer, seanbright Review: + https://reviewboard.asterisk.org/r/272/ ........ + +2009-06-03 20:30 +0000 [r198824-198954] David Vossel + + * apps/app_dial.c, main/channel.c, apps/app_queue.c: + ast_call_forward() todo notes and originate flag copy. + + * main/channel.c, main/features.c, include/asterisk/channel.h: + Generic call forward api, ast_call_forward() The function + ast_call_forward() forwards a call to an extension specified in + an ast_channel's call_forward string. After an ast_channel is + called, if the channel's call_forward string is set this function + can be used to forward the call to a new channel and terminate + the original one. I have included this api call in both + channel.c's ast_request_and_dial() and feature.c's + feature_request_and_dial(). App_dial and app_queue already + contain call forward logic specific for their application and + options. (closes issue #13630) Reported by: festr Review: + https://reviewboard.asterisk.org/r/271/ + + * channels/chan_iax2.c: fixes issue with channels not going down + after transfer Iax2 currently does not support native bridging if + the timeoutms value is set. We check for that in iax2_bridge, but + then set timeoutms to 0 by default. If the timeoutms is not + provided it is set to -1. By setting timeoutms to 0 it is + processed causing a bridging retry loop. (closes issue #15216) + Reported by: oxymoron Tested by: dvossel + +2009-06-02 13:48 +0000 [r198762-198791] Joshua Colp + + * channels/chan_sip.c, configs/sip.conf.sample: Correct + documentation for the register line, specifically where the + domain should be specified. (closes issue #14367) Reported by: + Nick_Lewis + + * main/rtp_engine.c: Fix a bug where we were passing in address + information that should remain unmodified to a function that may + modify it. (closes issue #15243) Reported by: pj + +2009-06-01 21:03 +0000 [r198729] Russell Bryant + + * channels/iax2-parser.c: Tell the IAX2 parser about more control + frame types. + +2009-06-01 20:57 +0000 [r198727] Mark Michelson + + * apps/app_dial.c, main/channel.c, include/asterisk/app.h, + main/dial.c, channels/chan_sip.c, apps/app_directed_pickup.c, + main/features.c, apps/app_macro.c, doc/tex/channelvariables.tex, + main/app.c, include/asterisk/channel.h, apps/app_queue.c: Add the + ability to execute connected line interception macros. When + connected line updates are received or generated in the middle of + an application call, it is now possible to execute a macro to + manipulate the connected line data. This way, phone numbers may + be manipulated to be more presentable to users, names may be + changed for...whatever reason, or whatever else needs to be done + may be. Review: https://reviewboard.asterisk.org/r/256 AST-165 + +2009-06-01 20:33 +0000 [r198725] Tilghman Lesher + + * funcs/func_math.c: Add INCrement and DECrement functions (closes + issue #15025) Reported by: greenfieldtech Patches: + func_math.c.patch_v4 uploaded by greenfieldtech (license 369) + slightly modified by me Tested by: greenfieldtech, lmadsen + +2009-06-01 20:17 +0000 [r198670] Russell Bryant + + * include/asterisk/frame.h: Minor whitespace fix. + +2009-06-01 19:37 +0000 [r198661] Eliel C. Sardanons + + * res/res_monitor.c: Moved more static documentation to the new + AstXML form. Moved more static docs to XML (pplications and + manager actions): Monitor, StopMonitor, ChangeMonitor, + PauseMonitor, UnpauseMonitor. + +2009-06-01 18:40 +0000 [r198626] Tilghman Lesher + + * contrib/scripts/meetme.sql: Add information for new meetme + realtime fields + +2009-06-01 17:53 +0000 [r198561-198597] Eliel C. Sardanons + + * main/Makefile: Do not add say.o in a separate line. + + * res/res_jabber.c: Move JabberSend manager action from static docs + to the AstXML form. + + * res/res_agi.c: Move static documentation of E|Dead|AGI() + application and manager action to XML. + +2009-06-01 15:23 +0000 [r198558] David Vossel + + * main/threadstorage.c: Fixed an issue in the threadstorage cli + functions resulting from the constification of struct + ast_cli_args in r196072. + +2009-06-01 14:45 +0000 [r198500-198530] Mark Michelson + + * apps/app_queue.c: Remove extra lock from app_queue. + + * channels/chan_local.c: Remove extra lock from local_indicate in + connected line case. Oh, and this fixes a deadlock I was seeing. + + * channels/chan_local.c: Add missing unlock of local pvt. + + * channels/chan_agent.c: Remove documentation for the 'exten' + argument to the AGENT function. Since AgentCallbackLogin has been + removed, this should not be documented any more. + +2009-06-01 13:31 +0000 [r198498] Joshua Colp + + * channels/chan_sip.c: Fix a bug where the Event and Content-Type + headers were added twice to outgoing SIP NOTIFY messages. (closes + issue #15239) Reported by: pj + +2009-05-31 17:52 +0000 [r198470] Tilghman Lesher + + * funcs/func_strings.c: Fix documentation for FIELDQTY. + +2009-05-31 02:09 +0000 [r198442] Eliel C. Sardanons + + * main/Makefile: Filter the say.o object, it is being added later. + +2009-05-31 01:40 +0000 [r198438] Russell Bryant + + * main/manager.c: Constification and remove some unused code. + +2009-05-31 01:22 +0000 [r198437] Eliel C. Sardanons + + * res/res_timing_dahdi.c: Avoid a crash when res_timing_dahdi is + unloaded but wasn't properly loaded. if dahdi_test_timer() fails, + timing_funcs_handle remains NULL causing a crash when calling + ast_unregister_timing_interface() with a NULL pointer. (closes + issue #15234) Reported by: eliel Patches: timing_dahdi1.diff + uploaded by eliel (license 64) + +2009-05-31 01:19 +0000 [r198434] Russell Bryant + + * main/channel.c, include/asterisk/channel.h: Constify the + ast_frame arg to ast_queue_frame(). + +2009-05-30 20:11 +0000 [r198371-198375] Sean Bright + + * res/res_jabber.c: Properly terminate the receive buffer before + sending to iksemel. aji_io_recv takes the maximum number of bytes + to read (instead of the total buffer size), so we have to + subtract 1 from our buffer size. Without this, when we receive + packets that are larger than our buffer, iksemel will choke and + things get wonky. (closes issue #15232) Reported by: lp0 Patches: + 05302009_res_jabber.c.patch uploaded by seanbright (license 71) + Tested by: seanbright, lp0 + + * /, res/res_jabber.c: Merged revisions 198370 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May + 2009) | 12 lines Properly terminate AMI JabberSend response + messages. The response message (either Error or Success) needs an + extra trailing \r\n after the fields to inform the client that + the message is complete. (closes issue #14876) Reported by: srt + Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright + (license 71) asterisk_14876.patch uploaded by srt (license 378) + trunk-14876-2.diff uploaded by phsultan (license 73) ........ + +2009-05-30 03:43 +0000 [r198312] Russell Bryant + + * res/res_smdi.c, /: Merged revisions 198311 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) + | 5 lines Fix a crash that occurred when MWI SMDI messages + expired. (closes issue #14561) Reported by: cmoss28 ........ + +2009-05-30 03:26 +0000 [r198285] Sean Bright + + * apps/app_dial.c, /: Merged revisions 198251 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May + 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we + treat a missing one. (closes issue #15056) Reported by: + p_lindheimer Patches: 05292009_bug15056.diff uploaded by + seanbright (license 71) Tested by: p_lindheimer ........ + +2009-05-30 02:31 +0000 [r198248] Joshua Colp + + * channels/chan_sip.c: When removing all packets from a dialog we + also need to free the data if present. + +2009-05-30 01:04 +0000 [r198217] Eliel C. Sardanons + + * configs/agents.conf.sample, channels/chan_agent.c: Remove not + used code in the Agent channel. This code was there because of + the AgentCallbackLogin() application. ->loginchan[] member was + only used by AgentCallbackLogin(). Agent where dumped to astdb if + they where logged in using AgentCallbacklogin() so they are not + being dumper anymore. Review: + https://reviewboard.asterisk.org/r/267/ + +2009-05-29 23:04 +0000 [r198183-198186] Russell Bryant + + * configs/modules.conf.sample: Suggesting that only a single timing + module be loaded is no longer necessary. + + * res/res_timing_pthread.c: Improve handling of trying to ACK too + many timer expirations. + +2009-05-29 22:21 +0000 [r198182] Terry Wilson + + * res/res_calendar.c: Add a couple of TODO items so I don't forget + +2009-05-29 20:06 +0000 [r198146] Russell Bryant + + * res/res_timing_pthread.c: Resolve issues with choppy sound when + using res_timing_pthread. The situation that caused this problem + was when continuous mode was being turned on and off while a rate + was set for a timing interface. A very easy way to replicate this + bug was to do a Playback() from behind a Local channel. In this + scenario, a rate gets set on the channel for doing file playback. + At the same time, continuous mode gets turned on and off about + every 20 ms as frames get queued on to the PBX side channel from + the other side of the Local channel. Essentially, this module + treated continuous mode and a set rate as mutually exclusive + states for the timer to be in. When I dug deep enough, I observed + the following pattern: 1) Set timer to tick every 20 ms. 2) Wait + almost 20 ms ... 3) Continuous mode gets turned on for a queued + up frame 4) Continuous mode gets turned off 5) The timer goes + back to its tick per 20 ms. state but starts counting at 0 ms. 6) + Goto step 2. Sometimes, res_timing_pthread would make it 20 ms + and produce a timer tick, but not most of the time. This is what + produced the choppy sound (or sometimes no sound at all). Now, + the module treats continuous mode and a set rate as completely + independent timer modes. They can be enabled and disabled + independently of each other and things work as expected. (closes + issue #14412) Reported by: dome Patches: issue14412.diff.txt + uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt + uploaded by russell (license 2) Tested by: DennisD, russell + +2009-05-29 19:46 +0000 [r198139] Eliel C. Sardanons + + * main/Makefile: Simplify the Makefile and avoid needing to specify + each object file. Instead of specifying every object file, use + make's magic to generate it. This will generate less conflicts in + team branches when a new file is added in trunk. (closes issue + #15226) Reported by: eliel Patches: makefile uploaded by eliel + (license 64) Review: http://reviewboard.asterisk.org/r/269/ + +2009-05-29 19:19 +0000 [r198088] Jeff Peeler + + * channels/chan_dahdi.c, channels/sig_analog.c (added), + channels/sig_analog.h (added), channels/Makefile: New signaling + module to handle analog operations in chan_dahdi This branch + splits all the analog signaling logic out of chan_dahdi.c into + sig_analog.c. Functionality in theory should not change at all. + As noted in the code, there is still some unused code remaining + that will be cleaned up in a later commit. Review: + https://reviewboard.asterisk.org/r/253/ + +2009-05-29 19:18 +0000 [r198083] Eliel C. Sardanons + + * CREDITS: Apply anti-spam obfuscation to an email address. + +2009-05-29 19:04 +0000 [r198072] Matthew Nicholson + + * main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged + revisions 198068 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May + 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as + the default CDR disposition. This change also involves the + addition of an AST_CDR_FLAG_ORIGINATED flag that is used on + originated channels to distinguish: them from dialed channels. + (closes issue #12946) Reported by: meral Patches: null-cdr2.diff + uploaded by mnicholson (license 96) Tested by: mnicholson, + dbrooks (closes issue #15122) Reported by: sum Tested by: sum + ........ + +2009-05-29 18:39 +0000 [r198064] Joshua Colp + + * main/file.c: Fix a memory leak of the write buffer when writing a + file. + +2009-05-29 18:15 +0000 [r198000] Sean Bright + + * Makefile, /: Merged revisions 197998 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May + 2009) | 8 lines Fix 'make config' target for Slackware. There was + a missing semi-colon after the echo statement in the Makefile + that was causing problems for some users. Fix suggested by + reporter. (closes issue #15225) Reported by: pdavis ........ + +2009-05-29 17:51 +0000 [r197996] Joshua Colp + + * channels/chan_sip.c: Fix a bug where the default setting did not + perform a remote bridge when it should have. + +2009-05-29 16:15 +0000 [r197960] Russell Bryant + + * res/res_timing_pthread.c: Trim trailing whitespace so that I can + work on this bug without it bothering me. :-) + +2009-05-29 15:48 +0000 [r197959] Mark Michelson + + * channels/chan_sip.c: A few fixes to SIP with regards to connected + line updates during transfers. * Set the invitestate to + INV_CALLING when we send a connected line reinvite. This prevents + us from potentially rapid-firing reinvites to a single peer. * + Use the astdb to store a peer's allowed methods. This prevents us + from sending an UPDATE during the interval between startup and + the peer's first registration if the peer does not support the + UPDATE method. * Handle Polycom's method of indicating allowed + methods in REGISTER. Instead of using an Allow header, they place + the allowed methods in a methods= parameter in the Contact + header. ABE-1873 + +2009-05-29 05:15 +0000 [r197926] Terry Wilson + + * doc/tex/asterisk.tex, doc/tex/calendaring.tex (added): Add some + TeX docs for calendaring. I still need to set up tests to make + sure my examples are completely correct, but I ran out of time + tonight and felt that they at least would give an idea as to how + to use calendaring. I will try to test the examples and do some + cleanup on the docs tomorrow night. + +2009-05-28 22:42 +0000 [r197861] Sean Bright + + * include/asterisk/doxygen/releases.h, sounds/Makefile: Update + references to downloads.digium.com to its new URL. + +2009-05-28 22:04 +0000 [r197828] Leif Madsen + + * apps/app_mixmonitor.c: Update documentation in MixMonitor. + Updated the MixMonitor documentation for the 'b' option so that + it is more obvious that you must not optimize away the Local + channel when using this option. (closes issue #14829) Reported + by: licedey Tested by: mmichelson, licedey, lmadsen + +2009-05-28 21:50 +0000 [r197824] Sean Bright + + * doc/CODING-GUIDELINES, doc/asterisk.8, BUGS, doc/backtrace.txt, + doc/tex/mp3.tex, channels/h323/README, main/enum.c, + doc/tex/misdn.tex, include/asterisk/doxyref.h, + contrib/scripts/ast_grab_core, doc/tex/backtrace.tex, + include/asterisk/doxygen/reviewboard.h, + include/asterisk/doxygen/commits.h, + contrib/scripts/asterisk.ldif, + contrib/scripts/asterisk.ldap-schema, + configs/extensions.conf.sample, doc/asterisk.sgml: Update + references to bugs.digium.com and reviewboard.digium.com to the + new URLs. + +2009-05-28 20:43 +0000 [r197777] Terry Wilson + + * configs/calendar.conf.sample: Make note of Exchange calendar + support limitations + +2009-05-28 20:36 +0000 [r197775] Kevin P. Fleming + + * main/utils.c: Ensure that accidental calls to + ast_string_field_free_memory() on embedded stringfield pools are + safe. It is possible for a stringfield manager structure (and + pool) structure to be allocated as part of a larger structure + allocation (using ast_calloc_with_strinfields()); when this is + done, the stringfield pool cannot be separately freed, but users + of the tructure may not be aware (and shouldn't have to be aware) + of whether the pool was embedded. This patch modifies the + behavior so that they can always call + ast_string_field_free_memory() and the function will do the right + thing for both embedded and non-embedded situations. + +2009-05-28 20:17 +0000 [r197740] Mark Michelson + + * channels/chan_sip.c: Treat 405 responses the same way we would a + 501. This makes sure that we mark a method as being unallowed if + we receive a 405 response so that we don't continue to try to + send that same type of message. + +2009-05-28 19:57 +0000 [r197738] Terry Wilson + + * res/res_calendar.exports (added), res/res_calendar_exchange.c + (added), res/res_calendar_icalendar.c (added), + build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, + configs/calendar.conf.sample (added), res/res_calendar_caldav.c + (added), include/asterisk/calendar.h (added), makeopts.in, + res/res_calendar.c (added): Add Calendaring support for Asterisk + This commit add Calendaring support to Asterisk for iCalendar, + CalDAV, and MS Exchange calendars. Exchange support has only been + tested on Exchange Server 2k3 and does not support forms-based + authentication at this time (patches *very* welcome). Exchange + support is also currently missing the ability to return a list of + a meting's attendees (again, patches are very, very welcome). + Features include: Querying a calendar for events over a specific + time range Checking a calendar's busy status via the dialplan + Writing calendar events via the dialplan (CalDAV and Exchange + only) Handling calendar event notifications through the dialplan + (closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash + Review: https://reviewboard.asterisk.org/r/58 + +2009-05-28 18:48 +0000 [r197701] Mark Michelson + + * channels/chan_local.c: Add missing lock to local_indicate + function for connected line frames. + +2009-05-28 18:45 +0000 [r197697] Joshua Colp + + * channels/chan_iax2.c: Fix a bug where the trunkmtu setting was + not set to the default value of 1240 on load but was on reload. + +2009-05-28 16:01 +0000 [r197621] Eliel C. Sardanons + + * /, channels/chan_sip.c: Merged revisions 197562 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | + 13 lines Use the address we already know when reloading a peer + with nat=yes. If we already have an address for a peer, and we + are reloading the sip configuration, try to use that address to + contact the peer, instead of getting it from the Contact. (closes + issue #15194) Reported by: ibc Patches: sip.patch uploaded by + eliel (license 64) Tested by: manwe ........ + +2009-05-28 15:35 +0000 [r197616] Tilghman Lesher + + * channels/chan_dahdi.c, channels/chan_console.c, apps/app_rpt.c, + main/astobj2.c, main/cli.c: Eliminate several needless checks and + fix a few memory leaks (closes issue #14833) Reported by: + contactmayankjain Patches: all_changes.patch uploaded by + contactmayankjain (license 740) slightly modified by me + +2009-05-28 15:32 +0000 [r197606] Mark Michelson + + * /: Recorded merge of revisions 197588 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May + 2009) | 16 lines Allow for media to arrive from an alternate + source when responding to a reinvite with 491. When we receive a + SIP reinvite, it is possible that we may not be able to process + the reinvite immediately since we have also sent a reinvite out + ourselves. The problem is that whoever sent us the reinvite may + have also sent a reinvite out to another party, and that reinvite + may have succeeded. As a result, even though we are not going to + accept the reinvite we just received, it is important for us to + not have problems if we suddenly start receiving RTP from a new + source. The fix for this is to grab the media source information + from the SDP of the reinvite that we receive. This information is + passed to the RTP layer so that it will know about the alternate + source for media. Review: https://reviewboard.asterisk.org/r/252 + ........ + +2009-05-28 15:23 +0000 [r197570] Joshua Colp + + * main/logger.c: Fix an incorrect call to + ast_string_field_free_memory which caused a crash in the logger. + Since the message structure is allocated using + ast_calloc_with_stringfields we do not need to free the memory + used for the stringfields as it will get freed when the message + structure is. + +2009-05-28 14:58 +0000 [r197543] Mark Michelson + + * /, include/asterisk/audiohook.h, main/audiohook.c, + apps/app_chanspy.c: Merged revisions 197537 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May + 2009) | 21 lines Add flags to chanspy audiohook so that audio + stays in sync. There are two flags being added to the chanspy + audiohook here. One is the pre-existing + AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that + the read and write slinfactories on the audiohook do not skew + beyond a certain tolerance. In addition, there is a new audiohook + flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, + we do not allow for a slinfactory to build up a substantial + amount of audio before flushing it. For this particular issue, + this means that the person spying on the call will hear the + conversations in real time with very little delay in the audio. + (closes issue #13745) Reported by: geoffs Patches: 13745.patch + uploaded by mmichelson (license 60) Tested by: snblitz ........ + +2009-05-28 14:51 +0000 [r197538] Joshua Colp + + * main/utils.c: Fix a bug in stringfields where it did not actually + free the pools of memory. (closes issue #15074) Reported by: pj + +2009-05-28 14:39 +0000 [r197528-197535] Sean Bright + + * configs/amd.conf.sample, configs/users.conf.sample, + configs/gtalk.conf.sample, configs/rpt.conf.sample, + configs/rtp.conf.sample, configs/cli_aliases.conf.sample, + configs/modules.conf.sample, configs/phone.conf.sample, + configs/extensions.ael.sample, configs/skinny.conf.sample, + configs/ais.conf.sample, configs/meetme.conf.sample, + configs/extensions_minivm.conf.sample, configs/telcordia-1.adsi, + configs/alsa.conf.sample, configs/iax.conf.sample, + configs/followme.conf.sample, configs/mgcp.conf.sample, + configs/sip.conf.sample, configs/extensions.lua.sample, + configs/say.conf.sample, configs/queuerules.conf.sample, + configs/minivm.conf.sample, configs/osp.conf.sample, + configs/chan_dahdi.conf.sample, + configs/cli_permissions.conf.sample, configs/console.conf.sample, + configs/dundi.conf.sample, configs/indications.conf.sample, + configs/oss.conf.sample, configs/queues.conf.sample, + configs/voicemail.conf.sample, configs/usbradio.conf.sample, + configs/cdr.conf.sample, configs/jingle.conf.sample, + configs/misdn.conf.sample, configs/manager.conf.sample, + configs/festival.conf.sample, configs/features.conf.sample, + configs/logger.conf.sample, configs/http.conf.sample, + configs/h323.conf.sample, configs/sla.conf.sample, + configs/phoneprov.conf.sample, configs/res_odbc.conf.sample, + configs/agents.conf.sample, configs/alarmreceiver.conf.sample, + configs/func_odbc.conf.sample, configs/musiconhold.conf.sample, + configs/jabber.conf.sample, configs/extconfig.conf.sample, + configs/res_snmp.conf.sample, configs/iaxprov.conf.sample, + configs/unistim.conf.sample, configs/dnsmgr.conf.sample, + configs/extensions.conf.sample, configs/asterisk.adsi: Remove a + bunch of trailing whitespace in preparation for + reformatting/cleanup. Let's try that again, this time removing + trailing whitespace and not leading whitespace. I can't believe + no one noticed. + + * configs/amd.conf.sample, configs/gtalk.conf.sample, + configs/rtp.conf.sample, configs/rpt.conf.sample, + configs/cli_aliases.conf.sample, configs/extensions.ael.sample, + configs/skinny.conf.sample, configs/meetme.conf.sample, + configs/telcordia-1.adsi, configs/alsa.conf.sample, + configs/iax.conf.sample, configs/mgcp.conf.sample, + configs/extensions.lua.sample, configs/sip.conf.sample, + configs/say.conf.sample, configs/minivm.conf.sample, + configs/console.conf.sample, configs/cli_permissions.conf.sample, + configs/chan_dahdi.conf.sample, configs/oss.conf.sample, + configs/queues.conf.sample, configs/jingle.conf.sample, + configs/usbradio.conf.sample, configs/voicemail.conf.sample, + configs/misdn.conf.sample, configs/manager.conf.sample, + configs/features.conf.sample, configs/h323.conf.sample, + configs/sla.conf.sample, configs/res_odbc.conf.sample, + configs/phoneprov.conf.sample, configs/alarmreceiver.conf.sample, + configs/func_odbc.conf.sample, configs/musiconhold.conf.sample, + configs/jabber.conf.sample, configs/unistim.conf.sample, + configs/dnsmgr.conf.sample, configs/extensions.conf.sample, + configs/asterisk.adsi: Remove a bunch of trailing whitespace in + preparation for reformatting/cleanup. + +2009-05-28 13:47 +0000 [r197467] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 197466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 + lines Fix a bug where the flag indicating the presence of rport + would get overwritten by the nat setting. The presence of rport + is now stored as a separate flag. Once the dialog is setup and + authenticated (or it passes through unauthenticated) the proper + nat flag is set. (closes issue #13823) Reported by: dimas + ........ + +2009-05-28 11:25 +0000 [r197406-197431] Gavin Henry + + * contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif: Added AstVoicemailContext Added + AstVoicemailContext (closes issue #15155) Reported by: scramatte + Tested by: suretec + + * contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif: New objectclass AsteriskVoiceMail + and AstAccountCallLimit attribute Added new ObjectClass + AsteriskVoiceMail, and AstAccountCallLimit attribute and cleaned + up formatting and tested with OpenLDAP (closes issue #15155) + Reported by: scramatte Patches: asterisk.schema uploaded by + scramatte (license 796) Tested by: suretec Review: [full review + board URL with trailing slash] + + * doc/ldap.txt, configs/res_ldap.conf.sample, + contrib/scripts/asterisk.ldap-schema, + contrib/scripts/asterisk.ldif: closes issue #15156 + +2009-05-27 23:48 +0000 [r197374] Tilghman Lesher + + * main/xml.c: Revert commit 192032. This define is needed on Mac OS + X. + +2009-05-27 22:42 +0000 [r197338] Russell Bryant + + * main/rtp_engine.c: Don't do a pointer comparison before setting + the remote address. + +2009-05-27 22:21 +0000 [r197335] Kevin P. Fleming + + * include/asterisk/agi.h: Ensure that this header includes + xmldoc.h, since it depends on it. + +2009-05-27 20:14 +0000 [r197266] Olle Johansson + + * channels/chan_sip.c: Adding some generic handling of error codes + sent to us in replys to requests. Previously they always set + hangupcause 0, which is generally wrong. With this change, we're + setting some generic hangup causes. For 5xx errors, which + indicate some sort of problem with the remote server, we're now + setting CONGESTION. EDVX002 + +2009-05-27 20:08 +0000 [r197260] Sean Bright + + * Makefile: Use bash explicitly when calling + build_tools/mkpkgconfig from the Makefile. Since we use bashisms + in build_tools/mkpkgconfig, we should call on bash explicitly + when running from the Makefile, otherwise we get errors during a + 'make install.' (closes issue #15209) Reported by: seandarcy + +2009-05-27 19:20 +0000 [r197209] Tilghman Lesher + + * /, funcs/func_cut.c: Recorded merge of revisions 197194 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) + | 5 lines Use a different determinator on whether to print the + delimiter, since leading fields may be blank. (closes issue + #15208) Reported by: ramonpeek Patch by me, though inspired in + part by a patch from ramonpeek ........ + +2009-05-27 18:25 +0000 [r196948-197189] Sean Bright + + * configs/adtranvofr.conf.sample (removed): Remove a file sample + configuration file that is no longer used. + + * configs/chan_dahdi.conf.sample, configs/vpb.conf.sample, + configs/smdi.conf.sample, configs/extensions.conf.sample, + configs/sla.conf.sample: Fix references to /etc/dahdi/system.conf + and /etc/asterisk/chan_dahdi.conf in the sample configuration + files. (closes issue #15207) Reported by: seandarcy + + * channels/chan_alsa.c: Display an error message when chan_alsa + fails to load due to a missing or inaccessible configuration + file. Before this change, when chan_alsa failed to load due to a + missing or inaccessible configuration file, no message would be + displayed. With this change, when chan_alsa fails to load due to + a missing or inaccessible configuration file, a message will be + displayed. (closes issue #14760) Reported by: Nick_Lewis Patches: + chan_alsa.c-confload.patch uploaded by Nick (license 657) + + * main/xmldoc.c: Reset the terminal to the correct fg/bg after XML + documenation is rendered. (closes issue #15200) Reported by: + ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright + (license 71) Tested by: ajohnson + +2009-05-26 22:40 +0000 [r196946] Russell Bryant + + * autoconf/ast_check_osptk.m4 (added), configure, + include/asterisk/autoconfig.h.in, configure.ac: Update configure + script to check for OSP toolkit 3.5.0. (closes issue #14988) + Reported by: tzafrir Patches: configure.ac.diff uploaded by + homesick (license 91) new_ast_check_osptk.m4 uploaded by homesick + (license 91) + +2009-05-26 22:38 +0000 [r196907-196945] Sean Bright + + * main/manager.c: Add ActionID to CoreShowChannel event. There is + inconsistency in how we handle manager responses that are lists + of items and, unfortunately, third parties have come to rely on + ActionID being on every event within those lists instead of just + keeping track of the ActionID for the current response. This + change makes CoreShowChannels include the ActionID with each + CoreShowChannel event generated as a result of it being called. + (closes issue #15001) Reported by: sum Patches: + patchactionid2.patch uploaded by sum (license 766) + + * main/manager.c: Include startup and reload date in the CoreStatus + manager message. The CoreStartupTime and CoreReloadTime + name/value pairs in the CoreStatus response message only included + the time and not the date. This patch, inspired by the reporter's + patch, adds 2 new fields - CoreStartupDate and CoreReloadDate - + which contain the date portion of these values. (closes issue + #15000) Reported by: sum + +2009-05-26 19:50 +0000 [r196893] Mark Michelson + + * channels/chan_sip.c, apps/app_directed_pickup.c: Remove some + redundant or unnecessary connected line-related function calls. + +2009-05-26 18:20 +0000 [r196843] Russell Bryant + + * /, res/res_convert.c: Merged revisions 196826 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) + | 9 lines Resolve a file handle leak. The frames here should have + always been freed. However, out of luck, there was never any + memory leaked. However, after file streams became reference + counted, this code would leak the file stream for the file being + read. (closes issue #15181) Reported by: jkroon ........ + +2009-05-26 16:38 +0000 [r196725-196792] Sean Bright + + * apps/app_queue.c: Add a missing unref for queues in + handle_statechange. + + * main/pbx.c, include/asterisk/pbx.h, res/res_clioriginate.c: Add + new ast_complete_applications function so that we can use it with + the 'channel originate ... application ' CLI command. (And + yeah, I cleaned up some whitespace in res_clioriginate.c... big + whoop, wanna fight about it!?) + + * cdr/cdr_sqlite3_custom.c: Use a properly allocated channel for + substitution in cdr_sqlite3_custom. + +2009-05-26 13:43 +0000 [r196658-196721] Joshua Colp + + * channels/chan_sip.c: Fix a bug where the sip unregister CLI + command did not completely unregister the peer. (closes issue + #15118) Reported by: alecdavis Patches: + chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585) + + * /, contrib/scripts/safe_asterisk: Merged revisions 196657 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7 + lines Remove some bash specific stuff from safe_asterisk. (closes + issue #10812) Reported by: paravoid Patches: + safe_asterisk_bashism.diff uploaded by tzafrir (license 46) + ........ + +2009-05-26 12:14 +0000 [r196622] Sean Bright + + * cdr/cdr_manager.c: Use a properly allocated channel for + substitution in cdr_manager. + +2009-05-24 16:17 +0000 [r196554-196585] Eliel C. Sardanons + + * res/res_agi.c: Move AGI static documentation to the new AstXML + form. Move AGI commands documentation to XML docs: 'set priority' + 'set variable' 'stream file' 'control stream file' 'tdd mode' + 'verbose' 'wait for digit' 'speech create' 'speech set' 'speech + destroy' 'speech load grammar' 'speech unload grammar' 'speech + activate grammar' 'speech deactivate grammar' 'speech recognize' + + * res/res_agi.c: Move static AGI commands documentation to XML. + Move AGI commands ('say datetime', 'send image', 'send text', + 'set autohangup', 'set callerid', 'set context', 'set extension') + documentation to the AstXML form. + +2009-05-23 15:16 +0000 [r196520] Sean Bright + + * cdr/cdr_custom.c: Fix errors in cdr_custom that cause reference + errors when non-CDR variable substitution is done. cdr_custom was + creating a ast_channel struct directly and passing it into the + core for variable substition. This was fine as long as the format + string contained only calls to the CDR() function. Doing + something like ${EPOCH} on the other hand tried to lock the + channel, which would fail and throw an error because the passed + channel hadn't been allocated as an ao2 object. So now we create + the dummy channel with ast_channel_alloc, and everything works as + expected. + +2009-05-23 13:31 +0000 [r196488] Kevin P. Fleming + + * include/asterisk/cli.h: Correct example for CLI autocompletion + (generation) Reported by Atis on #asterisk-dev + +2009-05-23 04:27 +0000 [r196456] Moises Silva + + * channels/chan_dahdi.c: set MFCR2_CATEGORY just when starting the + pbx + +2009-05-22 21:11 +0000 [r196417] Sean Bright + + * main/asterisk.c: Call ast_stun_init() when we're initializing to + get the 'stun debug set' commands. + +2009-05-22 21:09 +0000 [r196416] David Vossel + + * channels/chan_sip.c, configs/sip.conf.sample: SIP set outbound + transport type from Registration In sip.conf the transport option + allows for the configuration of what transport types (udp, tcp, + and tls) a peer will accept, but only the first type listed was + used for outbound connections. This patch changes this. Now the + default transport type is only used until the peer registers. + When registration takes place the transport type is parsed out of + the Contact header. If the Contact header's transport type is + equal to one that the peer supports, the peer's default transport + type for outbound connections is set to match the Contact + header's type. If the Contact header's transport type is not + present, then the peer's default transport type is set to match + the one the peer registered with. When a peer unregisters or the + registration expires, the default transport type for that peer is + reset. (closes issue #12282) Reported by: rjain Patches: + reg_patch_1.diff uploaded by dvossel (license 671) Tested by: + dvossel (closes issue #14727) Reported by: pj Patches: + reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, + dvossel Review: https://reviewboard.asterisk.org/r/249/ + +2009-05-22 20:01 +0000 [r196381] Sean Bright + + * channels/chan_gtalk.c: Don't crash if an RTP instance can't be + created. This could occur when an invalid bindaddr was specified + in gtalk.conf. + +2009-05-22 19:38 +0000 [r196308-196377] Eliel C. Sardanons + + * apps/app_minivm.c: Unregister every registered application by + MiniVM. The MinivmMWI application was not being unregistered on + unload and we were not able to load again the module or reload + it. (closes issue #15174) Reported by: junky Patches: + unregister_minivm_mwi.diff uploaded by junky (license 177) + + * res/res_agi.c: Moved static documentation to the AstXML form. + Moved AGI commands static documentation to XML docs ('say alpha', + 'say digits', 'say number', 'say phonetic', 'say date' and 'say + time'). + + * main/pbx.c, channels/chan_sip.c, apps/app_meetme.c, + channels/chan_agent.c, apps/app_queue.c, channels/chan_iax2.c, + include/asterisk/manager.h, channels/chan_dahdi.c, + main/manager.c, channels/chan_skinny.c, main/features.c, + res/res_agi.c, include/asterisk/xmldoc.h, include/asterisk/pbx.h, + apps/app_senddtmf.c, doc/appdocsxml.dtd, main/db.c, + main/xmldoc.c, apps/app_voicemail.c: Implement a new element in + AstXML for AMI actions documentation. A new xml element was + created to manage the AMI actions documentation, using AstXML. To + register a manager action using XML documentation it is now + possible using ast_manager_register_xml(). The CLI command + 'manager show command' can be used to show the parsed + documentation. Example manager xml documentation: AMI action + synopsis. <-- for ActionID Description + ... AMI action + description ... + + +2009-05-22 16:53 +0000 [r196272] Tilghman Lesher + + * main/astmm.c: Two more minor fixes due to constification + +2009-05-22 16:51 +0000 [r196270] Sean Bright + + * res/res_agi.c: Fix res_agi compilation after the const-ify the + world merge. Since we are dealing with a 'const char * const' + now, we have to create a temporary copy of the string to work on + rather than the original. Fix inspired by reporter. Reviewed by + everyone-and-their-mother in #asterisk-dev. (closes issue #15184) + Reported by: andrew + +2009-05-22 16:50 +0000 [r196268] Mark Michelson + + * channels/chan_sip.c: s/it's/its/ + +2009-05-22 16:20 +0000 [r196246] Russell Bryant + + * channels/chan_dahdi.c: resolve compiler warning + +2009-05-22 16:10 +0000 [r196227] Sean Bright + + * channels/chan_dahdi.c, main/pbx.c, res/res_jabber.c, + res/res_monitor.c: Fix build under dev mode and remove some casts + that are no longer necessary as a result of the const-ify the + world patch. + +2009-05-22 15:07 +0000 [r196187-196188] Richard Mudgett + + * apps/app_mp3.c: Fix constify the world compile problem. + + * channels/chan_misdn.c: Make chan_misdn compile. + +2009-05-22 13:56 +0000 [r196117] Joshua Colp + + * channels/chan_misdn.c, /: Merged revisions 196116 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May + 2009) | 5 lines Fix a bug where using immediate with mISDN caused + a cause code of 16 to get sent back instead of 1 if the 's' + extension did not exist. (closes issue #12286) Reported by: + lmamane ........ + +2009-05-22 13:34 +0000 [r196114] Eliel C. Sardanons + + * main/pbx.c: Avoid using prototypes when not necessary (it is + already defined in the header file). Make log_match_char_tree() + static to main/pbx.c (only used there). + +2009-05-21 21:13 +0000 [r196072] Kevin P. Fleming + + * apps/app_dahdibarge.c, main/frame.c, apps/app_record.c, + apps/app_playtones.c, funcs/func_strings.c, + include/asterisk/extconf.h, apps/app_alarmreceiver.c, + apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c, + channels/chan_iax2.c, main/astobj2.c, channels/chan_dahdi.c, + channels/chan_skinny.c, apps/app_dumpchan.c, pbx/pbx_ael.c, + main/pbx.c, channels/vcodecs.c, apps/app_softhangup.c, + apps/app_morsecode.c, apps/app_talkdetect.c, + channels/iax2-parser.c, apps/app_db.c, apps/app_speech_utils.c, + apps/app_sendtext.c, pbx/pbx_config.c, apps/app_mixmonitor.c, + main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c, + apps/app_dictate.c, apps/app_authenticate.c, + apps/app_readexten.c, apps/app_userevent.c, res/res_jabber.c, + include/asterisk/abstract_jb.h, main/channel.c, + apps/app_setcallerid.c, apps/app_osplookup.c, funcs/func_odbc.c, + apps/app_mp3.c, apps/app_minivm.c, apps/app_directory.c, + apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, + apps/app_read.c, channels/chan_sip.c, + include/asterisk/taskprocessor.h, include/asterisk/cli.h, + apps/app_originate.c, utils/conf2ael.c, + apps/app_channelredirect.c, apps/app_forkcdr.c, + main/abstract_jb.c, channels/misdn/chan_misdn_config.h, + apps/app_sms.c, utils/extconf.c, funcs/func_devstate.c, + apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c, + include/asterisk/agi.h, cdr/cdr_sqlite3_custom.c, + apps/app_readfile.c, apps/app_sayunixtime.c, apps/app_test.c, + include/asterisk/speech.h, cdr/cdr_adaptive_odbc.c, + apps/app_image.c, main/taskprocessor.c, main/loader.c, + main/cli.c, apps/app_skel.c, include/asterisk/module.h, + main/features.c, apps/app_amd.c, channels/chan_alsa.c, + apps/app_url.c, apps/app_externalivr.c, formats/format_gsm.c, + apps/app_milliwatt.c, res/res_speech.c, main/ast_expr2.fl, + apps/app_dial.c, include/asterisk/utils.h, apps/app_page.c, + apps/app_privacy.c, apps/app_fax.c, apps/app_echo.c, + channels/chan_agent.c, apps/app_dahdiras.c, apps/app_disa.c, + pbx/dundi-parser.c, apps/app_transfer.c, res/res_monitor.c, + apps/app_playback.c, include/asterisk/app.h, + channels/chan_misdn.c, apps/app_waitforring.c, + include/asterisk/image.h, apps/app_macro.c, + apps/app_zapateller.c, apps/app_chanspy.c, apps/app_cdr.c, + channels/chan_unistim.c, apps/app_meetme.c, main/utils.c, + res/res_musiconhold.c, apps/app_followme.c, + channels/misdn_config.c, apps/app_controlplayback.c, main/ulaw.c, + main/cdr.c, main/manager.c, channels/console_gui.c, + cdr/cdr_sqlite.c, res/res_agi.c, main/app.c, + apps/app_confbridge.c, main/image.c, apps/app_ivrdemo.c, + apps/app_parkandannounce.c, res/res_clioriginate.c, + apps/app_jack.c, apps/app_while.c, res/res_rtp_asterisk.c, + apps/app_nbscat.c, apps/app_festival.c, res/res_limit.c, + apps/app_waitforsilence.c, apps/app_waituntil.c, + channels/chan_console.c, apps/app_queue.c, apps/app_system.c, + apps/app_getcpeid.c, channels/chan_oss.c, + include/asterisk/features.h, apps/app_flash.c, + apps/app_directed_pickup.c, channels/chan_nbs.c, + include/asterisk/strings.h, include/asterisk/pbx.h, + apps/app_senddtmf.c: Const-ify the world (or at least a good part + of it) This patch adds 'const' tags to a number of Asterisk APIs + where they are appropriate (where the API already demanded that + the function argument not be modified, but the compiler was not + informed of that fact). The list includes: - CLI command handlers + - CLI command handler arguments - AGI command handlers - AGI + command handler arguments - Dialplan application handler + arguments - Speech engine API function arguments In addition, + various file-scope and function-scope constant arrays got 'const' + and/or 'static' qualifiers where they were missing. Review: + https://reviewboard.asterisk.org/r/251/ + +2009-05-21 19:11 +0000 [r195995] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 195991 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 + May 2009) | 14 lines Sign problem calculating timestamp for iax + frame leads to no audio on the receiving peer. There are rare + cases in which a frame's delivery timestamp is slightly less than + the iax2_pvt's offset. This causes the pvt's timestamp to be a + small negative number, but since the timestamp value is unsigned + it looks like a huge positive number. This patch checks for this + negative case and sets the ms to zero. A similar check is already + done right below this one in the 'else' statement. (closes issue + #15032) Reported by: guillecabeza Patches: + chan_iax2.c.patch_timestamp uploaded by guillecabeza (license + 380) Tested by: guillecabeza (closes issue #14216) Reported by: + Andrey Sofronov ........ + +2009-05-21 19:06 +0000 [r195992] Mark Michelson + + * main/features.c: Pass connected line updates along during a + bridge. + +2009-05-21 17:15 +0000 [r195949] Sean Bright + + * configs/cdr_custom.conf.sample: Rework the cdr_custom.conf.sample + header a bit to reflect the changes in functionality (allowing + multiple mappings). + +2009-05-21 15:33 +0000 [r195882] Matthew Nicholson + + * main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195881 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May + 2009) | 13 lines This commit prevents cdr records with + AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated + in certain cases. This is accomplished by adding two functions to + update the answer time and disposition of calls that checks for + the proper lock flags. These functions are used in the + ast_bridge_call() function so that ForkCDR(A) calls are + respected. This patch also modifies the way ast_bridge_call() + chooses the cdr record to base the bridged_cdr on. Previously the + first unlocked cdr record would be chosen, now instead the first + cdr record is chosen and forked cdr records are moved to the + bridge_cdr. This allows the original cdr record and any forked + cdr records to be properly updated with answer and end times. + (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes + issue #14744) Reported by: deepesh ........ + +2009-05-20 23:30 +0000 [r195839] Tilghman Lesher + + * apps/app_stack.c: If a variable had a blank value upon the + initial setting, then it would do nothing. Identified by Dmitry + Andrianov via private email, fixed by me. + +2009-05-20 20:45 +0000 [r195763-195798] Mark Michelson + + * channels/chan_sip.c: Get rid of some duplicated code and correct + a connected line error. When receiving a 200 OK response to an + INVITE, it was possible to transmit two connected line updates + instead of a single one. Furthermore, the second did not have the + proper information present. Now the two have been combined into a + single update and the correct information is presented. + + * apps/app_dial.c: Plug a memory leak in app_dial. Since we may + have copied connected line info into the chanlist struct prior to + placing an outbound call, we need to be sure to free the + allocated data when we hang the call up. + +2009-05-20 17:33 +0000 [r195636-195698] Joshua Colp + + * /, main/features.c: Merged revisions 195688 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 + lines Fix some code that wrongly assumed a pointer would always + be non-NULL when dealing with CDRs after a bridge. (closes issue + #15079) Reported by: barryf ........ + + * /, apps/app_meetme.c: Merged revisions 195635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 + lines Fix a bug where the MeetMe option 'D' did not actually + prompt for the pin. (closes issue #15050) Reported by: pmhaddad + ........ + +2009-05-19 20:59 +0000 [r195589] Mark Michelson + + * channels/chan_sip.c, configs/sip.conf.sample: Add basic support + for handling connected line-related UPDATE requests. SIP purists + may want to look the other way... When COLP/CONP support for SIP + was committed, there was a condition under which Asterisk may + transmit a SIP UPDATE in order to communicate the change in + connected line information. The issue here is that while we could + send a SIP UPDATE message, we were not prepared to receive such + an UPDATE and would always responde with a 501 when we received + an UPDATE. The situation was a bit rough. We really want to be + able to receive UPDATEs having to do with connected line changes, + but the amount of effort involved in properly supporting RFC 3311 + was staggering. This commit represents a compromise. First, it + was decided that it is important to only send a SIP UPDATE to an + endpoint that is able to handle one. So, now we have added + parsing of the Allow header into SIP. We store the allowed + methods on SIP peers so that when we communicate with them, we + already will know what we can and cannot send to them. We will + parse the peer's allowed methods when he registers with us. If + the peer is not the type to register with us, but the qualify + option is enabled, then we will use the response to the OPTIONS + request we send the peer to determine the peer's allowed methods. + When the peer's registration expires, or when qualify deems the + peer to be unreachable, we clear the allowed methods from the + peer. For an actual call, we will copy the peer's allowed methods + to the sip_pvt representing the call leg. If we are communicating + with an endpoint which is not a peer, then we will just parse the + Allow header from the first message we receive during the call + and store the information in the sip_pvt. If, during + communication with a peer, we receive a 501 response, then we + will make sure to save the fact that we cannot use that method + when communicating with that peer. Now, with all that + infrastructure in place, the only actual place we use this + information currently is when attempting to send a connected line + change using an UPDATE request. If we cannot send the change + immediately using an UPDATE, we will set the SIP_NEEDREINVITE + flag so that we can send a REINVITE as soon as it is allowed. The + second part of the changes here is for Asterisk to accept UPDATE + requests that have connected line changes. Since we are not fully + supporting RFC 3311, Asterisk will NOT place the UPDATE method in + Allow headers it sends. Instead, if you are communicating with + what you know to be another Asterisk box, you may set the + rpid_update parameter in sip.conf so that we will send UPDATEs to + that Asterisk box. When we send a connected line update, we set a + custom header called "X-Asterisk-rpid-update." On the receiving + end, if Asterisk receives an UPDATE that does not have the + "X-Asterisk-rpid-update" header present, then Asterisk will + respond with a 501 since media-changing UPDATEs are not + supported. We should never get such UPDATEs, since as was stated + earlier, Asterisk does not put UPDATE in its Allow header. If the + custom header is present in the received UPDATE, though, then we + will check the incoming request for connected line updates and + queue the update on the channel where the change occurred. + ABE-1840 ABE-1822 + +2009-05-19 20:16 +0000 [r195521] Tilghman Lesher + + * /, apps/app_voicemail.c: Merged revisions 195520 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 + May 2009) | 7 lines Ensure thread keys are initialized before + attempting to access them. (closes issue #14889) Reported by: + jaroth Patches: app_voicemail.c.patch uploaded by msirota + (license 758) Tested by: msirota, BlargMaN ........ + +2009-05-19 14:43 +0000 [r195449] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 195448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 + lines Fix a bug where direct RTP setup would partially occur even + when disabled if the calling channel was answered. (issue #13545) + Reported by: davidw (issue #14244) Reported by: mbnwa ........ + +2009-05-18 20:52 +0000 [r195370] Tilghman Lesher + + * res/res_smdi.c, /, include/asterisk/monitor.h, apps/app_queue.c, + include/asterisk/smdi.h, res/res_monitor.c, apps/app_voicemail.c: + Recorded merge of revisions 195366 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) + | 8 lines Add a similar dependency on SMDI for voicemail as + already exists for ADSI. (closes issue #14846) Reported by: pj + Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman + (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by + tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt + uploaded by tilghman (license 14) ........ + +2009-05-18 20:49 +0000 [r195365-195369] Eliel C. Sardanons + + * main/manager.c: Fix the CLI command 'manager show command' + documentation and functionality. The CLI command 'manager show + command' supports passing multiple action names in the same line, + but it was not allowing that because of a incorrect check in the + argumentes counter. Also the documentation was updated to show + that this usage of the command is possible. + + * main/manager.c: Rollback commit 195367. The CLI command 'manager + show command' supports passing multiple AMI actions at a time. + The issue with this command was in another place. + + * main/manager.c: Avoid autocompleting passed the action name + argument in the CLI command. When running the autocomplete of the + CLI command 'manager show command ' it was autocompleting + everything else after the argument, giving an error, + because this command doesn't support multiple AMI action names at + a time. + + * res/res_agi.c: Move AGI documentation from static to the XML + form. Move the AGI commands 'receive text', 'receive char' and + 'record' static documentation to XML docs. + +2009-05-18 19:17 +0000 [r195320] Tilghman Lesher + + * main/asterisk.c: Move the spawn of astcanary down, until after + the call to daemon(3). This avoids possible conflicts with the + internal implementation of daemon(3). (closes issue #15093) + Reported by: tzafrir Patches: 20090513__issue15093__2.diff.txt + uploaded by tilghman (license 14) Tested by: tzafrir + +2009-05-18 18:58 +0000 [r195316] Mark Michelson + + * apps/app_externalivr.c: Fix externalivr's setvariable command so + that it properly sets multiple variables. The command had a for + loop that was guaranteed to only execute once since the + continuation operation of the loop would set the input buffer + NULL. I rewrote the loop so that its operation was more obvious, + and it would set multiple variables correctly. I also reduced + stack space required for the function, constified the input + string, and modified the function so that it would not modify the + input string while I was at it. (closes issue #15114) Reported + by: chris-mac Patches: 15114.patch uploaded by mmichelson + (license 60) Tested by: chris-mac + +2009-05-18 17:08 +0000 [r195279] Sean Bright + + * cdr/cdr_custom.c: Remove some unused code. + +2009-05-18 16:29 +0000 [r195266] Richard Mudgett + + * channels/chan_dahdi.c: The facilityenable parameter does not have + anything to do with pritimer parameters. + +2009-05-18 15:55 +0000 [r195210] Sean Bright + + * cdr/cdr_custom.c: Const-ify a string, fix a log message, and use + the correct signature for the load_module function. + +2009-05-18 15:53 +0000 [r195207] Joshua Colp + + * main/frame.c, /: Merged revisions 195206 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7 + lines Fix a typo which caused loss of audio when using G729 in + some scenarios with a smoother present. (closes issue #15105) + Reported by: bamby Patches: process-vad-correctly.diff uploaded + by bamby (license 430) ........ + +2009-05-18 14:54 +0000 [r195165] Sean Bright + + * configs/cdr_custom.conf.sample, CHANGES, cdr/cdr_custom.c: Allow + cdr_custom to write to multiple files instead of just one. Up to + now, cdr_custom would only accept a single filename/format from + cdr_custom.conf. This change allows you to specify multiple + filename & format directives. + +2009-05-18 14:45 +0000 [r195162] Eliel C. Sardanons + + * apps/app_dial.c, main/pbx.c, apps/app_macro.c: Warn about the use + of the application WaitExten() within a Macro(). Update + applications documentation to warn the user about the use of the + WaitExten() application within a Macro(). Recommend the use of + Read() instead. (closes issue #14444) Reported by: ewieling + +2009-05-18 13:56 +0000 [r195089-195096] Joshua Colp + + * main/rtp_engine.c, /: Merged revisions 195095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5 + lines Fix a bug where the codecs of the called party leg were not + properly sent back to the caller call leg when reinvited. (closes + issue #13569) Reported by: bkw918 ........ + + * channels/chan_sip.c: Fix a bug where specifying an empty + outboundproxy would cause packets to get sent to ourself. (closes + issue #15106) Reported by: timeshell + +2009-05-18 13:30 +0000 [r195075] Eliel C. Sardanons + + * main/xml.c: Do not avoid loading the XML documentation if not + XInclude substitution is done. + +2009-05-18 12:59 +0000 [r195021] Russell Bryant + + * /: Recorded merge of revisions 195020 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009) + | 5 lines Don't try to unlock a bogus channel. (closes issue + #15144) Reported by: cristiandimache ........ + +2009-05-16 20:01 +0000 [r194945-194982] Eliel C. Sardanons + + * Makefile, main/xml.c, doc/appdocsxml.dtd: Allow to include + sections of other parts of the xml documentation. Avoid + duplicating xml documentation by allowing to include other parts + of the xml documentation using XInclude. Example: + (Insert this line to include the synopsis of the CHANNEL function + xml documentation). It is also possible to include documentation + from other files in the 'documentation/' directory using the + href="" attribute inside a xinclude element. (closes issue + #15107) Reported by: lmadsen (issue #14444) Reported by: ewieling + + * main/pbx.c: Fix a missing unlock in case of error, and a missing + free(). Always free the allocated memory for a string field, + because we are always using it (not only when xmldocs are + enabled). Also if there is an error allocating memory for the + string field remember to unlock the list of registered + applications, before returning. + +2009-05-15 22:44 +0000 [r194833-194874] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 194873 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 + May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ + to terminate invalid registrations. Instead it sent another + REGAUTH if the authentication challenge failed. This caused a + loop of REGREQ and REGAUTH frames. (Related to Security fix + AST-2009-001) (closes issue #14867) Reported by: aragon Tested + by: dvossel (closes issue #14717) Reported by: mobeck Patches: + regauth_loop_update_patch.diff uploaded by dvossel (license 671) + Tested by: dvossel ........ + + * channels/iax2-parser.h, /, channels/iax2.h, channels/chan_iax2.c, + channels/iax2-parser.c: Merged revisions 194557,194685 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) + | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue + where people are reporting "Ghost" channels in their 'iax2 show + channels' output. The confusion is caused by channels being + listed as "(NONE)" with format "unknown". These are not channels + of coarse. They are usually just pending registration or poke + requests, but it is confusing output. To help make sense of this + I have added two columns to 'iax2 show channels'. One shows the + first message which started the transaction, and the second shows + the last message sent by either side of the call. This helps + diagnose why the entry exists and why it may not go away. (closes + issue #14207) Reported by: clive18 Review: + https://reviewboard.asterisk.org/r/246/ ........ r194685 | + dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines + Update to previous IAX2 "Ghost" Channels patch. Fixed some + comments made on reviewboard for the previous patch. (issue + #14207) ........ + +2009-05-15 18:43 +0000 [r194714-194765] Russell Bryant + + * /, configs/logger.conf.sample: Merged revisions 194764 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) + | 2 lines Fix some spelling fail. ........ + + * codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Shuttle + some bits around to address some gain issues with G.722. (closes + AST-209) + + * codecs/Makefile, codecs/g722/Makefile (removed): Further simplify + codec_g722 build. + + * codecs/Makefile: Actually force running make for g722. + +2009-05-15 13:43 +0000 [r194649] Michiel van Baak + + * CREDITS: add eliel + +2009-05-15 13:23 +0000 [r194635] Eliel C. Sardanons + + * doc/appdocsxml.dtd, main/xmldoc.c: Allow to specify an enumlist + inside an enum. It was not possible to use an enumlist inside an + enum: ... + Now we will be able to insert as many levels + as we want. (closes issue #15112) Reported by: lmadsen + +2009-05-15 13:13 +0000 [r194520-194610] Kevin P. Fleming + + * include/asterisk/logger.h, tests/test_logger.c (added), + main/logger.c: Add ability for modules to dynamically register + logger levels This patch adds the ability for modules to + dynamically create logger levels for their own use; these are + named levels just like the built-in levels, and can be directed + to any destination that the logger can send any level to, by + including their names in logger.conf. Review: + https://reviewboard.asterisk.org/r/244/ + + * /: Merged revisions 194509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May + 2009) | 1 line Update URL to Reviewboard ........ + +2009-05-14 22:20 +0000 [r194496] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 194484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May + 2009) | 24 lines Fix a race condition where a reinvite could + trigger a 482 response. The loop detection/spiral detection code + in chan_sip used the owner channel's state as a criterion for + determining if the incoming INVITE is a looped request. The + problem with this is that the INVITE-handling code happens in a + different thread than the thread that marks the owner channel as + being up. As a result, if a reinvite were to come in very + quickly, say from another Asterisk on the same LAN, it was + possible for the reinvite to arrive before the owner channel had + been set to the up state. This patch corrects the problem by + using the invitestate of the sip_pvt instead, since that can be + guaranteed to be set correctly by the time the reinvite arrives. + Since there is a switch statement further in the INVITE-handling + code, the AST_STATE_RINGING state also checks the invitestate of + the sip_pvt in case we should actually be treating the channel as + if it were up already. (closes issue #12215) Reported by: jpyle + Patches: 12215_confirmed.patch uploaded by mmichelson (license + 60) Tested by: lmadsen ........ + +2009-05-14 22:03 +0000 [r194479] Richard Mudgett + + * channels/misdn/isdn_lib.h, channels/chan_misdn.c, + channels/misdn/chan_misdn_config.h, + channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample, + CHANGES, channels/misdn/isdn_lib.c, channels/misdn_config.c: Add + outgoing_colp misdn.conf port parameter. Select what to do with + outgoing COLP information on this port. 0 - Send out COLP + information unaltered. (default) 1 - Force COLP to restricted on + all outgoing COLP information. 2 - Do not send COLP information. + outgoing_colp=0 Also fixed sending the EctInform message so it + always has the required redirectionNumber parameter when the + status is active. JIRA ABE-1853 + +2009-05-14 21:24 +0000 [r194477] Russell Bryant + + * main/features.c: Fix a typo where an equality check should be an + assignment. (closes issue #15103) Reported by: lmsteffan Patches: + transfer_crash.patch uploaded by lmsteffan (license 779) + +2009-05-14 17:05 +0000 [r194434] Joshua Colp + + * apps/app_meetme.c: Fix a bug where the 'T' option to Meetme did + not work. (closes issue #15031) Reported by: Stochastic (closes + issue #13801) Reported by: justdave + +2009-05-14 16:22 +0000 [r194430] Tilghman Lesher + + * main/pbx.c: If the timing ended on a zero, then we would loop + forever. (closes issue #14983) Reported by: teox Patches: + 20090513__issue14983.diff.txt uploaded by tilghman (license 14) + Tested by: teox + +2009-05-13 15:02 +0000 [r194283] Eliel C. Sardanons + + * main/manager.c: Do not lock the 'sessions' container, lock the + allocated 'session'. There was a typo in the structure being + locked, and we were locking the 'sessions' container instead of + the 'session' structure thar we are modifying. Reported by + seanbright on #asterisk-dev, thanks! + +2009-05-13 13:39 +0000 [r194209] Joshua Colp + + * res/res_rtp_asterisk.c, /: Merged revisions 194208 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May + 2009) | 11 lines Fix RFC2833 issues with DTMF getting duplicated + and with duration wrapping over. (closes issue #14815) Reported + by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license + 88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes + issue #14460) Reported by: moliveras Tested by: moliveras + ........ + +2009-05-13 00:52 +0000 [r194101-194138] Tilghman Lesher + + * main/pbx.c, /: Merged revisions 194137 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009) + | 7 lines Fix logic for how to proceed with a single digit + extension. (closes issue #15091) Reported by: andrew Patches: + 20090512__issue15091.diff.txt uploaded by tilghman (license 14) + Tested by: andrew ........ + + * main/pbx.c, main/logger.c: Two fixes found while debugging with + ast_backtrace(): 1) If MALLOC_DEBUG is used when concurrently + using ast_backtrace, the free() used in that routine will trigger + an error, because the memory was allocated internally to libc, + where we could not intercept that call to wrap it. Therefore, + it's not memory we knew about, and the free is reported as an + error. 2) Now that channels are objects, the old hack of + initializing a channel to all zeroes no longer works, since we + may try to call something like ast_channel_lock() within a + function on that reference. In that case, it's reported as an + error, because the pointer isn't an object reference. + +2009-05-12 22:49 +0000 [r194060] Eliel C. Sardanons + + * main/manager.c: Fix a crash when logging out from the AMI and + avoid astobj2 warning messages. When the user logout the session + was being destroyed twice and the file descriptor was being + closed twice. The sessions reference counter wasn't used in a + proper way. The 'mansession' structure was being treated as an + astobj2 and we were calling ao2_lock/ao2_unlock causing astobj2 + report a warning message and not locking the structure. Also we + were using an ugly naming convention 'destroy_session', + 'session_destroy', 'free_session', ... all this "duplicated" code + was merged. (closes issue #14974) Reported by: pj Patches: + manager.diff2 uploaded by eliel (license 64) Tested by: dhubbard, + eliel, mnicholson (closes issue #15088) Reported by: eliel + Review: http://reviewboard.asterisk.org/r/248/ + +2009-05-12 22:32 +0000 [r194057] Matthew Nicholson + + * /, apps/app_queue.c: Merged revisions 194028 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May + 2009) | 16 lines This change modifies app_queue to properly + generate CDR records in failure situations. This involves setting + a proper cdr disposition coresponding to the given failure + condition and ensuring the proper information is stored in the + cdr record. (closes issue #13691) Reported by: dferrer Tested by: + mnicholson (closes issue #13637) Reported by: atis Tested by: + atis ........ + +2009-05-12 20:40 +0000 [r193956] Tilghman Lesher + + * /, apps/app_voicemail.c: Merged revisions 193955 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 + May 2009) | 6 lines Avoid initializing routines if the + authentication fails. Fixes a crash (RR) issue. (closes issue + #14508) Reported by: tiziano Patches: + 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license + 377) ........ + +2009-05-12 20:28 +0000 [r193954] Mark Michelson + + * channels/chan_sip.c: Update spiral support in trunk and 1.6.X to + match what is in 1.4. In 1.4, a SIP spiral is treated the same + way as a call forward. This works much better than what is + currently in trunk and 1.6.X. The code in trunk and 1.6.X did not + create a new call to the recipient of the spiral, instead trying + to continue the same call. In addition to just being plain wrong, + this also had the side effect of only being able to spiral calls + to other SIP channels. With this in place, as long as call + forwards are honored, SIP spirals will work properly. This means + that it will work for outbound calls made by the Queue, Dial, and + Page applications. For originated calls and spool calls, however, + the spiral will not work properly until a generic call forward + mechanism is introduced into Asterisk. (relates to issue #13630) + +2009-05-12 17:29 +0000 [r193870] Tilghman Lesher + + * apps/app_voicemail.c: Convert a THREADSTORAGE object into a + simple malloc'd object (as suggested by Russell on -dev) + +2009-05-12 13:59 +0000 [r193832] Kevin P. Fleming + + * apps/app_dial.c, main/pbx.c, apps/app_meetme.c, apps/app_page.c, + main/devicestate.c, apps/app_queue.c, apps/app_transfer.c, + apps/app_playback.c, apps/app_controlplayback.c, main/term.c, + channels/chan_dahdi.c, channels/chan_misdn.c, funcs/func_curl.c, + apps/app_sendtext.c, apps/app_directed_pickup.c, + channels/console_gui.c, main/features.c, apps/app_confbridge.c, + apps/app_externalivr.c, apps/app_chanspy.c, + apps/app_mixmonitor.c, apps/app_stack.c, res/res_odbc.c, + apps/app_voicemail.c: add 'const' qualifiers in various places + where they should have been + +2009-05-11 23:04 +0000 [r193756-193757] Tilghman Lesher + + * apps/app_voicemail.c: Found and fixed a memory leak + + * /: Recorded merge of revisions 193755 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009) + | 18 lines Move 300 bytes around on the stack, to make more room + for an extension buffer. This allows more concurrent extensions + to be copied for a single voicemail, without creating a + possibility of upsetting existing users, where a dialplan could + run out of stack space where it had run fine before. + Alternatively, we could have allocated off the heap, but that is + a larger change and would have increased the chance for + instability introduced by this change. This is really solved + starting in 1.6.0.11, as the use of an ast_str buffer allows an + unlimited number of extensions (up to available memory). We + additionally create a new warning message when the buffer length + is exceeded, permitting administrators to see an issue after the + fact, whereas previously the list was silently truncated. (closes + issue #14739) Reported by: p_lindheimer Patches: + 20090417__bug14739.diff.txt uploaded by tilghman (license 14) + Tested by: p_lindheimer ........ + +2009-05-11 22:04 +0000 [r193718] Russell Bryant + + * res/res_timing_timerfd.c: Fix some timer state corruption. In + res_timer_timerfd, handle the case that set_rate gets called + while a timer is still in continuous mode. In this case, we want + to remember the configured rate, but not actually set it until + continuous mode has been disabled. Thanks to dvossel for finding + and helping to debug the problem. (closes issue #15080) Reported + by: dvossel Tested by: dvossel + +2009-05-11 19:32 +0000 [r193678] Tilghman Lesher + + * apps/app_voicemail.c: Don't nullify an ast_str pointer. (closes + issue #15061) Reported by: alecdavis + +2009-05-11 19:11 +0000 [r193614] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 193613 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 + May 2009) | 12 lines Sent wrong message to clear a call we + started if the other end has not responed yet. In the state + MISDN_CALLING (i.e. SETUP was sent but no answer has arrived + yet), it is not allowed to clear the call with RELEASE_COMPLETE. + It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only + allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, + 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. + JIRA ABE-1862 ........ + +2009-05-11 18:01 +0000 [r193545] Leif Madsen + + * /, funcs/func_channel.c: Recorded merge of revisions 193544 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009) + | 7 lines Document CHANNEL(transfercapability) in CLI + documentation. (issue #15073) Reported by: pkempgen Patches: + 20090511__issue15073.diff.txt uploaded by tilghman (license 14) + ........ + +2009-05-10 17:07 +0000 [r193502] Joshua Colp + + * main/bridging.c: Fix a bug where receiving a control frame of + subclass -1 would cause certain channels to get hung up. + +2009-05-09 11:33 +0000 [r193459-193461] Russell Bryant + + * include/asterisk/event.h: Minor documentation update for + ast_event_queue(). + + * main/channel.c: Declare private data as static. + +2009-05-08 20:32 +0000 [r193387] David Vossel + + * channels/chan_sip.c: TCP not matching valid peer. find_peer() + does not find a valid peer when using pvt->recv as the + sockaddr_in argument. Because of the way TCP works, the port + number in pvt->recv is not what we're looking for at all. There + is currently only one place that find_peer searches for a peer + using the sockaddr_in argument. If the peer is not found after + using pvt->recv (works for UDP since the port number will be + correct), a temp sockaddr_in struct is made using the Contact + header in the sip_request. This has the correct port number in + it. Review: http://reviewboard.digium.com/r/236/ + +2009-05-08 19:50 +0000 [r193349] Mark Michelson + + * apps/app_queue.c: Reset the members' call counts when resetting + queue statistics. This helps to prevent odd scenarios where a + queue will claim to have taken 0 calls, but the members appear to + have taken a non-zero amount. (closes issue #15068) Reported by: + sum Patches: patchreset.patch uploaded by sum (license 766) + Tested by: sum + +2009-05-08 15:18 +0000 [r193274] Sean Bright + + * funcs/func_devstate.c: Fix the spelling of UNAVAILABLE in + func_devstate CLI completion. + +2009-05-08 14:52 +0000 [r193263] David Vossel + + * /, channels/misdn_config.c: Merged revisions 193262 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 + May 2009) | 9 lines "misdn show config" segfaults asterisk, if no + MSN lists (closes issue #14976) Reported by: alecdavis Patches: + misdn_config.diff.txt uploaded by alecdavis (license 585) Tested + by: alecdavis, FabienToune ........ + +2009-05-08 14:06 +0000 [r193194] Kevin P. Fleming + + * /, main/logger.c, configs/logger.conf.sample: Merged revisions + 193193 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May + 2009) | 7 lines Make absolute paths for logger channels work + properly (Note: This is not a new feature, it was previously + undocumented and broken.) The Asterisk logger has a feature to + support absolute pathnames for logger channels, but the code + implementing the feature was broken. This has been fixed, and the + absolute path feature is now documented in the sample + logger.conf. ........ + +2009-05-07 23:42 +0000 [r193120] Tilghman Lesher + + * main/pbx.c, /: Merged revisions 193119 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009) + | 19 lines Fix Background within a Macro for FreePBX. If the + single digit DTMF is an extension in the specified context, then + go there and signal no DTMF. Otherwise, we should exit with that + DTMF. If we're in Macro, we'll exit and seek that DTMF as the + beginning of an extension in the Macro's calling context. If + we're not in Macro, then we'll simply seek that extension in the + calling context. Previously, someone complained about the + behavior as it related to the interior of a Gosub routine, and + the fix (#14011) inadvertently broke FreePBX (#14940). This + change should fix both of these situations, but with the possible + incompatibility that if a single digit extension does not exist + (but a longer extension COULD have matched), it would have + previously gone immediately to the "i" extension, but will now + need to wait for a timeout. (closes issue #14940) Reported by: + p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by + tilghman (license 14) Tested by: p_lindheimer ........ + +2009-05-07 22:24 +0000 [r193077] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 193050 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 + May 2009) | 5 lines Give a more helpful message when an incoming + call's dialed extension does not match. Added the dialed + extension and context to the chan_misdn messages warning that the + dialed number cannot be matched in the dialplan. ........ + +2009-05-07 17:51 +0000 [r192933-193006] Tilghman Lesher + + * funcs/func_odbc.c: Second result should not contain data from the + first result. (closes issue #15039) Reported by: jims Patches: + 20090506__issue15039.diff.txt uploaded by tilghman (license 14) + Tested by: jims + + * channels/chan_unistim.c: Send DTMF frame before playing back + audio. (closes issue #14858) Reported by: barryf Patches: + 20090507__bug14858.diff.txt uploaded by tilghman (license 14) + + * /, channels/chan_sip.c: Merged revisions 192932 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) + | 10 lines Eliminate repetition of fullcontact during + reconstruction. If the fullcontact field appears in both the + sippeers and the sipregs table, then during reconstruction of the + field, it will otherwise be doubled. (closes issue #14754) + Reported by: Alexei Gradinari Patches: + 20090506__bug14754.diff.txt uploaded by tilghman (license 14) + Tested by: lmadsen ........ + +2009-05-06 22:17 +0000 [r192853-192861] Jeff Peeler + + * /, main/features.c: Merged revisions 192858 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) + | 10 lines Make ParkedCall application stop execution of the + dialplan after hang up Just changed park_exec to always return + non-zero. I really wasn't entirely sure at first if this was a + bug. Decided it was since it would be surprising when not using + ParkedCall in the dialplan to hang up and have dialplan execution + continue. (closes issue #14555) Reported by: francesco_r ........ + + * main/pbx.c: If no extension was found in the pattern tree, don't + crash. + +2009-05-06 17:38 +0000 [r192808] Joshua Colp + + * channels/chan_iax2.c: Fix a bug where a timer would be created + but not acknowledged. This scenario crept up if chan_iax2 was + loaded with no configuration file present. It would create a + timer and tell it to go at an interval but the thread that + normally acknowledges it would not be created because no + configuration file was present. The timer will now be closed if + no configuration file is present. (closes issue #15014) Reported + by: madkins + +2009-05-06 16:28 +0000 [r192772] Tilghman Lesher + + * main/say.c, doc/lang/urdu.ods (added): Add numbers in Urdu, the + national language of Pakistan (closes issue #15034) Reported by: + nasirq Patches: ast_say_number_full_ur-patch.c uploaded by nasirq + (license 772) urdu.ods uploaded by nasirq (license 772) + +2009-05-06 16:09 +0000 [r192634-192736] Joshua Colp + + * res/res_clialiases.c: Make the code that prevents an infinite + loop from happening into a case insensitive check. (thanks eliel) + + * res/res_clialiases.c: Fix an infinite loop with tab completion of + CLI aliases that reference themselves. (closes issue #15020) + Reported by: junky + + * /, channels/chan_sip.c: Merged revisions 192633 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 + lines Update some old logic to stop both begin and end DTMF + frames from reaching the core if rfc2833 is not enabled. (closes + issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded + by dimas (license 88) ........ + +2009-05-05 20:54 +0000 [r192590] Richard Mudgett + + * apps/app_dial.c, channels/chan_sip.c, apps/app_directed_pickup.c, + main/features.c, apps/app_queue.c: Fixed crashes from issue8824 + review board channel locking changes. The local struct + ast_party_connected_line connected_caller variable was + uninitialized when the copy function was called. + +2009-05-05 19:57 +0000 [r192525] Sean Bright + + * /, static-http/astman.js: Merged revisions 192524 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, + 05 May 2009) | 11 lines Fix Javascript error when using astman.js + in Internet Explorer. Internet Explorer (tested with 7.0) does + not like trailing commas on constructs like object initializers, + so get rid of them to avoid some errors. (closes issue #15026) + Reported by: rajnishgiri Patches: bug15026.patch uploaded by + seanbright (license 71) Tested by: seanbright ........ + +2009-05-05 18:23 +0000 [r192430-192462] Joshua Colp + + * /, main/features.c: Merged revisions 192454 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 + lines Fix an incorrect assumption that certain values on the + channel will always exist when they may not. The CDR code + involved with bridges wrongly assumed that the currently + executing application and data values will always exist. It is + possible for this to be false when call forwarding is involved. + (closes issue #14984) Reported by: gincantalupo ........ + + * /, apps/app_followme.c: Merged revisions 192429 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5 + lines Fix a bug where the followme application would continue + trying numbers after the caller hung up. (closes issue #13624) + Reported by: sgenyuk ........ + +2009-05-05 17:33 +0000 [r192427] Matthew Fredrickson + + * channels/chan_dahdi.c: Revert CPC patch for now, until I decide + whether or not it all should be merged into libss7/1.0 (It's + still in the bug13495 branch and in libss7/trunk) + +2009-05-05 14:22 +0000 [r192387] Joshua Colp + + * channels/chan_sip.c: Fix a bug with setting t38pt_udptl at the + user or peer level. If an incoming call authenticated as a user + or peer and t38pt_udptl was not set to yes in general then no + UDPTL session would be present and any T38 related things would + fail. This commit changes it so that if after authenticating T38 + is enabled but no UDPTL session is present one will be created. + (issue AST-215) + +2009-05-05 14:17 +0000 [r192279-192362] Kevin P. Fleming + + * main/utils.c, include/asterisk/stringfields.h: Add a more + efficient way of allocating structures that use stringfields This + commit adds an API call that can be used to allocate a structure + along with this stringfield storage in a single allocation. + + * main/utils.c, main/astobj2.c, include/asterisk/stringfields.h: + Correct some flaws in the memory accounting code for stringfields + and ao2 objects Under some conditions, the memory allocation for + stringfields and ao2 objects would not have supplied valid + file/function names for MALLOC_DEBUG tracking, so this commit + corrects that. + + * main/channel.c, include/asterisk/astobj2.h, + include/asterisk/datastore.h, include/asterisk/channel.h, + main/astobj2.c, main/datastore.c: Properly account for memory + allocated for channels and datastores As in previous commits, + when channels are allocated (with ast_channel_alloc) or + datastores are allocated (with ast_datastore_alloc) properly + account for the memory being owned by the caller, instead of the + allocator function itself. + + * main/utils.c, include/asterisk/stringfields.h: Ensure that string + pools allocated to hold stringfields are properly accounted in + MALLOC_DEBUG mode This commit modifies the stringfield pool + allocator to remember the 'owner' of the stringfield manager the + pool is being allocated for, and ensures that pools allocated in + the future when fields are populated are owned by that + file/function. + +2009-05-04 22:44 +0000 [r192214] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 192213 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 + May 2009) | 11 lines global mohinterpret setting is ignored + mohinterpret and mohsuggest global variables were not copied over + during build_users and build_peers. (closes issue #14728) + Reported by: dimas Patches: v1-14728.patch uploaded by dimas + (license 88) Tested by: dimas, dvossel ........ + +2009-05-04 19:29 +0000 [r192132-192171] Tilghman Lesher + + * include/asterisk/autoconfig.h.in, res/res_agi.c: Restore + 'asyncagi break' command to 1.6.1 and higher. (closes issue + #14985) Reported by: nikkk Patches: 20090428__bug14985.diff.txt + uploaded by tilghman (license 14) + 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license + 14) Tested by: nikkk + + * autoconf/ast_ext_tool_check.m4: Pass libraries in LIBS, not + LDFLAGS. (closes issue #14671) Reported by: Chainsaw Patches: + asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by + Chainsaw (license 723) + +2009-05-04 17:42 +0000 [r192096] Leif Madsen + + * apps/app_forkcdr.c: Commit documentation changes related to issue + #14801. (issue #14801) + +2009-05-04 16:24 +0000 [r192059] Kevin P. Fleming + + * include/asterisk/astobj2.h, main/astobj2.c: Ensure that astobj2 + memory allocations are properly accounted for when MALLOC_DEBUG + is used This commit ensures that all astobj2 allocated objects + are properly accounted for in MALLOC_DEBUG mode by passing down + the file/function/line information from the module/function that + actually called the astobj2 allocation function. + +2009-05-04 15:35 +0000 [r192032] Eliel C. Sardanons + + * main/xml.c: Do not re-define _POSIX_C_SOURCE if it was already + defined. + +2009-05-04 12:52 +0000 [r191919-191997] Kevin P. Fleming + + * tests/test_skel.c, tests/test_sched.c: Minor changes in test + modules Correct command description in test_sched.c and include + asterisk/cli.h in test_skel.c, since it's highly unlikely that a + test module will *not* want to provide CLI commands to execute + the tests + + * configs/modules.conf.sample: Ensure that by default only one + console channel driver is loaded This configuration file was + changed to ensure that only one console channel driver (chan_oss) + is loaded by default, but the change would only work if + chan_console was not built. Now it will work as expected; if + chan_alsa or chan_console are built and installed, they will not + be loaded unless explicity requested. + + * include/asterisk/event.h, include/asterisk/event_defs.h, + main/event.c: Add 'bitflags'-style information elements to event + framework This patch add a new payload type for information + elements, a set of bit flags. The payload is transported as a + 32-bit unsigned integer but when matching is performed between + events and subscribers, the matching is done by using a bitwise + AND instead of numeric value comparison. Review: + http://reviewboard.asterisk.org/r/242/ + +2009-05-03 14:05 +0000 [r191848-191884] Russell Bryant + + * Makefile: Remove unnecessary compiler flag + + * main/event.c: Do a bit of code cleanup. - convert handling of IE + PLTYPEs to switch statements - add braces to various small blocks + - remove a bit of trailing whitespace - remove a couple of + unnecessary ast_strdupa() uses + +2009-05-02 19:02 +0000 [r191775-191785] Kevin P. Fleming + + * include/asterisk/logger.h, main/manager.c, pbx/pbx_spool.c, + main/logger.c, apps/app_sms.c, CHANGES, apps/app_verbose.c, + configs/logger.conf.sample: Remove rarely-used + event_log/LOG_EVENT support In discussions today at the Europe + Asterisk Developer Meet-Up, we determined that the event_log was + used in only 9 places in the entire tree, and really was not + needed at all. The users have been converted to use LOG_NOTICE, + or the messages have been removed since other messages were + already in place that provided the same information. + + * main/logger.c: Fix an error in queue_log file rotation + optimization code This code was copy-and-pasted without properly + changing references to event_rotate into queue_rotate, so under + some conditions the log rotation would rotate queue_log even + though it was not necessary. + +2009-05-02 16:43 +0000 [r191700-191739] Sean Bright + + * channels/chan_dahdi.c: Conditional include ioctl's to change EC + policy based on DAHDI caps. This feels like a sane change + (wouldn't compile without this addition), but I'm not intimately + familiar with this code. + + * main/asterisk.c: Update copyright year to 2009 + +2009-05-01 20:01 +0000 [r191494-191560] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 191559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) + | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1. + (closes issue #14993) Reported by: BigJimmy Patches: causepatch + uploaded by BigJimmy (license 371) ........ + + * channels/chan_iax2.c: Set debug message back to DEBUG level. + (closes issue #15007) Reported by: hulber + +2009-05-01 18:09 +0000 [r191489] Jeff Peeler + + * main/channel.c, /: Merged revisions 191488 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) + | 9 lines Fix DTMF not being sent to other side after a partial + feature match This fixes a regression from commit 176701. The + issue was that ast_generic_bridge never exited after the feature + digit timeout had elapsed, which prevented the queued DTMF from + being sent to the other side. This issue was reported to me + directly. ........ + +2009-05-01 14:58 +0000 [r191419] Joshua Colp + + * main/audiohook.c: Drop my IRC nickname. + +2009-05-01 09:50 +0000 [r191418] TransNexus OSP Development + + * configs/osp.conf.sample, apps/app_osplookup.c: Made security + features optional. + +2009-04-30 21:42 +0000 [r191411] Kevin P. Fleming + + * channels/chan_dahdi.c, configure, + include/asterisk/autoconfig.h.in, configure.ac, CHANGES: Add + buffer and echo canceller control to CHANNEL() dialplan function + for DAHDI channels Adds ability for CHANNEL() dialplan function, + when used on DAHDI channels, to temporarily change the number of + buffers and/or the buffer policy, and also to enable, disable, or + switch the echo canceller between FAX/data and voice modes. + +2009-04-30 17:40 +0000 [r191367] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + main/asterisk.c: Detect eaccess (or euidaccess) before using it. + Reported by Andrew Lindh via the -dev list. + +2009-04-30 09:11 +0000 [r191300-191332] TransNexus OSP Development + + * apps/app_osplookup.c: Added routing number support. + + * apps/app_osplookup.c: Fixed not report source network ID and not + export destination network ID issues. + +2009-04-30 06:47 +0000 [r191219-191283] Tilghman Lesher + + * main/asterisk.c: Change working directory to / under certain + conditions. If backgrounding and no core will be produced, then + changing the directory won't break anything; likewise, if the CWD + isn't accessible by the current user, then a core wasn't possible + anyway. (closes issue #14831) Reported by: chris-mac Patches: + 20090428__bug14831.diff.txt uploaded by tilghman (license 14) + 20090430__bug14831.diff.txt uploaded by tilghman (license 14) + Tested by: chris-mac + + * /: Recorded merge of revisions 191220 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r191220 | tilghman | 2009-04-29 18:10:54 -0500 (Wed, 29 Apr 2009) + | 2 lines Allow H.323 to compile with FDLEAK checking enabled. + ........ + + * channels/h323/ast_h323.cxx, channels/chan_h323.c: Make H.323 + compile with FDLEAK detection code enabled + +2009-04-29 22:56 +0000 [r191213] Jeff Peeler + + * res/res_phoneprov.c: fix typos + +2009-04-29 22:23 +0000 [r191211] Tilghman Lesher + + * main/pbx.c: Part of the merge did not happen correctly, which + resulted in a compile error + +2009-04-29 21:13 +0000 [r191177] David Vossel + + * main/tcptls.c, configs/sip.conf.sample, + include/asterisk/tcptls.h, CHANGES: SIP option to specify + outbound TLS/SSL client protocol. chan_sip allows for outbound + TLS connections, but does not allow the user to specify what + protocol to use (default was SSLv2, and still is if this new + option is not specified). This patch lets the user pick the + SSL/TLS client method for outbound connections in sip. (closes + issue #14770) Reported by: TheOldSaint (closes issue #14768) + Reported by: TheOldSaint Review: + http://reviewboard.digium.com/r/240/ + +2009-04-29 21:07 +0000 [r191175] Richard Mudgett + + * channels/chan_misdn.c, CHANGES: Outgoing PTP redirected calls did + not wait for the COLR from the redirected-to party. For outgoing + PTP redirected calls, you now need to use the inhibit(i) option + on all of the REDIRECTING statements before dialing the + redirected-to party. You still have to set the + REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The + PTP call will update the redirecting-to presentation when it + becomes available and queue the redirecting update to the calling + channel. + +2009-04-29 18:53 +0000 [r191140] Tilghman Lesher + + * tests/test_substitution.c (added), funcs/func_base64.c, + funcs/func_rand.c, funcs/func_speex.c, funcs/func_md5.c, + funcs/func_module.c, include/asterisk/autoconfig.h.in, + funcs/func_env.c, funcs/func_strings.c, res/res_phoneprov.c, + funcs/func_sysinfo.c, funcs/func_vmcount.c, funcs/func_sha1.c, + funcs/func_logic.c, apps/app_exec.c, funcs/func_groupcount.c, + configure, funcs/func_aes.c, main/ast_expr2f.c, res/res_agi.c, + apps/app_minivm.c, include/asterisk/ast_expr.h, cdr/cdr_custom.c, + main/strings.c, main/pbx.c, funcs/func_dialplan.c, + funcs/func_db.c, funcs/func_timeout.c, funcs/func_lock.c, + funcs/func_cut.c, funcs/func_extstate.c, res/res_config_curl.c, + funcs/func_curl.c, funcs/func_blacklist.c, apps/app_macro.c, + include/asterisk/pbx.h, funcs/func_callerid.c, + apps/app_voicemail.c: Merge str_substitution branch. This branch + adds additional methods to dialplan functions, whereby the result + buffers are now dynamic buffers, which can be expanded to the + size of any result. No longer are variable substitutions limited + to 4095 bytes of data. In addition, the common case of needing + buffers much smaller than that will enable substitution to only + take up the amount of memory actually needed. The existing + variable substitution routines are still available, but users of + those API calls should transition to using the dynamic-buffer + APIs. Reviewboard: http://reviewboard.digium.com/r/174/ + +2009-04-29 18:32 +0000 [r191136] David Brooks + + * pbx/pbx_config.c: Removing crufty code that is no longer + necessary. Code cleanup. + +2009-04-29 14:39 +0000 [r191028] David Vossel + + * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c, + configs/manager.conf.sample, include/asterisk/tcptls.h, CHANGES, + configs/http.conf.sample: Consistent SSL/TLS options across conf + files ast_tls_read_conf() is a new api call for handling SSL/TLS + options across all conf files. Before this change, SSL/TLS + options were not consistent. http.conf and manager.conf required + the 'ssl' prefix while sip.conf used options with the 'tls' + prefix. While the options had different names in different conf + files, they all did the exact same thing. Now, instead of mixing + 'ssl' or 'tls' prefixes to do the same thing depending on what + conf file you're in, all SSL/TLS options use the 'tls' prefix. + For example. 'sslenable' in http.conf and manager.conf is now + 'tlsenable' which matches what already existed in sip.conf. Since + this has the potential to break backwards compatibility, previous + options containing the 'ssl' prefix still work, but they are no + longer documented in the sample.conf files. The change is noted + in the CHANGES file though. Review: + http://reviewboard.digium.com/r/237/ + +2009-04-29 08:58 +0000 [r190989-190993] Russell Bryant + + * main/indications.c: Log an error message if indications.conf is + not found. (closes issue #14990) Reported by: tzafrir Patches: + indications_err.diff uploaded by tzafrir (license 46) + + * apps/app_queue.c: Fix app_queue XML documentation. I think it + would behoove us to force "make validate-docs" to be run after + the XML documentation has been generated if dev-mode is enabled. + (closes issue #14989) Reported by: tzafrir Patches: + app_queue_xml.diff uploaded by tzafrir (license 46) + + * main/rtp_engine.c, include/asterisk/channel.h: Resolve Solaris + build issues and add some API documentation. (issue #14981) + Reported by: snuffy + +2009-04-28 22:07 +0000 [r190946-190947] Matthew Fredrickson + + * channels/chan_dahdi.c: Add support setting CPC from channel + variable + + * channels/chan_dahdi.c: Make sure that we do not clear the down + flag on the BRI during PTMP link transients + +2009-04-28 17:31 +0000 [r190904] Tilghman Lesher + + * doc/tex/cdrdriver.tex: UniqueID column has a maximum size of 150 + +2009-04-28 14:15 +0000 [r190861-190865] Kevin P. Fleming + + * Makefile: Build XML documention from *only* the source files that + have docs in them Change the build process so that + doc/core-en_US.xml is dependent solely on the source files that + have documentation in them, not on all source files. + + * Makefile.rules: Remove Makefile rules for bison and flex sources + We never, ever want these files to processed automatically, + because we store the output files in Subversion and users should + never need to rebuild them. + +2009-04-28 09:10 +0000 [r190830] TransNexus OSP Development + + * apps/app_osplookup.c: Updated for OSP Toolkit 3.5. + +2009-04-27 21:22 +0000 [r190735-190797] Richard Mudgett + + * main/channel.c: Fix a small memory leak on error in + ast_channel_alloc(). + + * channels/misdn/isdn_lib.h, channels/chan_misdn.c, CHANGES, + channels/misdn/isdn_lib.c, funcs/func_redirecting.c: Make PTP + DivertingLegInformation3 message behavior closer to the + specifications. * Wait for a DivertingLegInformation3 message + after receiving a DivertingLegInformation1 message to complete + the redirecting-to information before queuing a redirecting + update to the other channel. * A DivertingLegInformation2 message + should be responded to with a DivertingLegInformation3 when the + COLR is determined. If the call could or does experience another + redirection, you should manually determine the COLR to send to + the switch by setting REDIRECTING(to-pres) to the COLR and + setting REDIRECTING(to-num) = ${EXTEN}. * A + DivertingLegInformation2 message must have an original called + number if the redirection count is greater than one. Since + Asterisk does not keep track of this information, we can only + indicate that the number is not available due to interworking. + +2009-04-27 19:34 +0000 [r190726] Tilghman Lesher + + * main/pbx.c: Don't warn on pipe in the System call. (closes issue + #14979) Reported by: pj + +2009-04-27 19:30 +0000 [r190725] Kevin P. Fleming + + * /, configure, include/asterisk/autoconfig.h.in: Merged revisions + 190721 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr + 2009) | 7 lines Fix 'inconsistent line endings' when autoconf + 2.63 is used Attempt to make configure script regeneration 'safe' + using autoconf 2.63, which embeds a bare CR into the script, thus + making Subversion complain about inconsistent line endings This + commit changes the MIME type of the configure script to be + 'binary' thus making Subversion no longer inspect line endings, + and as a bonus 'svn diff' will no longer try to generate diff + output for it, which is not generally useful anyway. ........ + +2009-04-27 19:08 +0000 [r190663] Russell Bryant + + * res/res_smdi.c, /: Merged revisions 190661-190662 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 + Apr 2009) | 9 lines Resolve a crash in res_smdi when used with + chan_dahdi. When chan_dahdi goes to get an SMDI message, it + provides no search criteria. It just grabs the next message that + arrives. This code was written with the SMDI dialplan functions + in mind, since that is now the preferred method of using SMDI. + However, this broke support of it being used from chan_dahdi. + (closes AST-212) ........ r190662 | russell | 2009-04-27 14:03:59 + -0500 (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661. + ........ + +2009-04-27 16:37 +0000 [r190622-190626] Mark Michelson + + * doc/tex/channelvariables.tex, apps/app_queue.c: Allow for a + position to be specified when entering a queue. This would allow + for one to add a caller to a specific place in the queue instead + of just placing the caller in the back every time. To help + facilitate some interesting manipulations, a new channel variable + called QUEUEPOSITION has been added. When a caller is removed + from a queue, his position in that queue is stored in the + QUEUEPOSITION variable. One such strategy an administrator can + employ is to allow for the removal of a caller from one queue + followed by the insertion of the same caller into a separate + queue in the same position. Review: + http://reviewboard.digium.com/r/189 + + * apps/app_queue.c: Update warning message to not have pipes and + contain all options. + +2009-04-27 15:18 +0000 [r190586] Joshua Colp + + * main/manager.c: Fix a bug where we tried to send events out when + no sessions container was present. This commit stops a warning + message (user_data is NULL) from getting output when manager + events get sent before manager is initialized. This happens + because manager is initialized *after* modules are loaded and the + act of loading modules triggers manager events. (issue #14974) + Reported by: pj + +2009-04-27 14:46 +0000 [r190577] Mark Michelson + + * configs/sip.conf.sample: Remove nonexistent option from + sip.conf.sample. The option to choose which connected line header + to use is not 'rpid_header' but 'sendrpid' + +2009-04-24 21:22 +0000 [r190545] David Vossel + + * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c, + configs/manager.conf.sample, configs/sip.conf.sample, + include/asterisk/tcptls.h, CHANGES, configs/http.conf.sample: + TLS/SSL private key option Adds option to specify a private key + .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. + Before this, the certificate file was used for both the public + and private key. It is possible for this file to hold both, but + most configurations allow for a separate private key file to be + specified. Clarified in .conf files how these options are to be + used. The current conf files do not explain how the private key + is handled at all, so without knowledge of Asterisk's TLS + implementation, it would be hard to know for sure what was going + on or how to set it up. Review: + http://reviewboard.digium.com/r/234/ + +2009-04-24 17:59 +0000 [r190516-190517] Richard Mudgett + + * channels/chan_misdn.c, funcs/func_connectedline.c: There is no + need to use the struct ast_party_connected_line.source update + values. The messages sent by a technology when a connected line + update is received are best determined by the current call state + of the channel. The struct ast_party_connected_line.source value + is really only useful as a possible tracing aid. + + * include/asterisk/channel.h: Update comment. + +2009-04-24 15:26 +0000 [r190423-190484] Russell Bryant + + * include/asterisk/channel.h: Add \since tag for new API calls. + + * channels/chan_misdn.c: Fix a build error. + + * channels/chan_unistim.c, channels/chan_local.c, + apps/app_dahdiscan.c (removed), main/devicestate.c, + main/autochan.c (added), funcs/func_logic.c, + channels/chan_gtalk.c, channels/chan_iax2.c, main/cli.c, + main/channel.c, build_tools/cflags.xml, channels/chan_dahdi.c, + main/manager.c, funcs/func_odbc.c, apps/app_minivm.c, + main/features.c, res/res_agi.c, main/logger.c, + channels/chan_mgcp.c, res/res_clioriginate.c, main/pbx.c, + channels/chan_sip.c, include/asterisk/autochan.h (added), + channels/chan_bridge.c, main/Makefile, apps/app_softhangup.c, + channels/chan_agent.c, UPGRADE.txt, include/asterisk/channel.h, + CHANGES, funcs/func_global.c, res/res_monitor.c, + apps/app_channelredirect.c, channels/chan_misdn.c, + apps/app_directed_pickup.c, funcs/func_channel.c, + res/snmp/agent.c, include/asterisk/lock.h, apps/app_senddtmf.c, + apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c: + Convert the ast_channel data structure over to the astobj2 + framework. There is a lot that could be said about this, but the + patch is a big improvement for performance, stability, code + maintainability, and ease of future code development. The channel + list is no longer an unsorted linked list. The main container for + channels is an astobj2 hash table. All of the code related to + searching for channels or iterating active channels has been + rewritten. Let n be the number of active channels. Iterating the + channel list has gone from O(n^2) to O(n). Searching for a + channel by name went from O(n) to O(1). Searching for a channel + by extension is still O(n), but uses a new method for doing so, + which is more efficient. The ast_channel object is now a + reference counted object. The benefits here are plentiful. Some + benefits directly related to issues in the previous code include: + 1) When threads other than the channel thread owning a channel + wanted access to a channel, it had to hold the lock on it to + ensure that it didn't go away. This is no longer a requirement. + Holding a reference is sufficient. 2) There are places that now + require less dealing with channel locks. 3) There are places + where channel locks are held for much shorter periods of time. 4) + There are places where dealing with more than one channel at a + time becomes _MUCH_ easier. ChanSpy is a great example of this. + Writing code in the future that deals with multiple channels will + be much easier. Some additional information regarding channel + locking and reference count handling can be found in channel.h, + where a new section has been added that discusses some of the + rules associated with it. Mark Michelson also assisted with the + development of this patch. He did the conversion of ChanSpy and + introduced a new API, ast_autochan, which makes it much easier to + deal with holding on to a channel pointer for an extended period + of time and having it get automatically updated if the channel + gets masqueraded. Mark was also a huge help in the code review + process. Thanks to David Vossel for his assistance with this + branch, as well. David did the conversion of the DAHDIScan + application by making it become a wrapper for ChanSpy internally. + The changes come from the + svn/asterisk/team/russell/ast_channel_ao2 branch. Review: + http://reviewboard.digium.com/r/203/ + +2009-04-24 13:49 +0000 [r190421] Joshua Colp + + * channels/chan_sip.c: Fix nat setting on RTP instances. (closes + issue #14827) Reported by: pj + +2009-04-23 21:13 +0000 [r190357] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 190356 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r190356 | russell | 2009-04-23 16:07:07 -0500 (Thu, 23 Apr 2009) + | 2 lines Remove a bogus ast_channel_unlock(). ........ + +2009-04-23 20:42 +0000 [r190349-190352] Tilghman Lesher + + * main/pbx.c: Labels are sometimes (most of the time?) NULL for + extensions. (closes issue #14895) Reported by: chris-mac Patches: + 20090423__bug14895__2.diff.txt uploaded by tilghman (license 14) + Tested by: lmadsen + + * include/asterisk/http.h, include/asterisk/utils.h, + main/manager.c, res/res_phoneprov.c, main/http.c, main/utils.c, + res/res_http_post.c, main/astobj2.c: Support HTTP digest + authentication for the http manager interface. (closes issue + #10961) Reported by: ys Patches: digest_auth_r148468_v5.diff + uploaded by ys (license 281) SVN branch + http://svn.digium.com/svn/asterisk/team/group/manager_http_auth + Tested by: ys, twilson, tilghman Review: + http://reviewboard.digium.com/r/223/ Reviewed by: + tilghman,russellb,mmichelson + +2009-04-23 19:15 +0000 [r190287] Joshua Colp + + * channels/chan_local.c, /: Merged revisions 190286 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr + 2009) | 6 lines Fix a bug in chan_local glare hangup detection. + If both sides of a Local channel were hung up at around the same + time it was possible for one thread to destroy the local private + structure and have the other thread immediately try to remove the + already freed structure from the local channel list. ........ + +2009-04-23 17:45 +0000 [r190250] Mark Michelson + + * apps/app_queue.c: Fix reversed behavior of leavewhenempty option + in queues.conf. (closes issue #14650) Reported by: alecdavis + Patches: 14650.patch uploaded by mmichelson (license 60) Tested + by: mmichelson, lmadsen + +2009-04-23 16:55 +0000 [r190217] Joshua Colp + + * apps/app_directed_pickup.c: Fix a double free issue with the + Pickup dialplan application. As part of the pickup process the + connected line information is updated. Part of this process does + a shallow copy of the target channel's connected line information + to a local structure. Once complete the structure contents are + freed. As a result any information in the target channel's + connected line information structure is no longer valid. This + change will now set the contents back to a clean state so that + the freeing of the target channel's connected line information + structure when the channel is destroyed will no longer try to + double free things. (closes issue #14839) Reported by: lmsteffan + +2009-04-23 00:44 +0000 [r190154] Terry Wilson + + * funcs/func_strings.c: Fix example that could fail in certain + circumstances + +2009-04-22 21:38 +0000 [r190093] Tilghman Lesher + + * configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/lock.h: Merged revisions 190092 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 + Apr 2009) | 7 lines Detect availability of + pthread_rwlock_timedwrlock() before using it. (closes issue + #14930) Reported by: tilghman Patches: + 20090420__bug14930.diff.txt uploaded by tilghman (license 14) + Tested by: mvanbaak, tilghman ........ + +2009-04-22 21:15 +0000 [r190057] Jeff Peeler + + * funcs/func_groupcount.c, main/app.c, include/asterisk/channel.h, + main/cli.c: Fix building of chan_h323 with gcc-3.3 There seems to + be a bug with old versions of g++ that doesn't allow a structure + member to use the name list. Rename list member to group_list in + ast_group_info and change the few places it is used. (closes + issue #14790) Reported by: stuarth + +2009-04-22 20:07 +0000 [r190000] Terry Wilson + + * funcs/func_strings.c: Add funcs for manipulating delimited lists + in the dialplan Adds PUSH and POP for appending to and + retrieving/removing from the end of a list and UNSHIFT and SHIFT + for insert to and retrieiving/ removing from the beginning of a + list. Review: http://reviewboard.digium.com/r/230 + +2009-04-22 19:23 +0000 [r189993] Jeff Peeler + + * channels/h323/ast_h323.cxx, channels/chan_h323.c, + channels/h323/chan_h323.h: Make chan_h323 respect packetization + settings and fix small reload issue. Previously, packetization + settings were ignored and now they are not. A new config option + 'autoframing' has been added to mirror the way chan_sip handles + it. Turning on the autoframing option (available both as a global + option or per peer) overrides the local settings with the remote + packetization settings. Testing was performed with varying + packetization levels with the following codecs: ulaw, alaw, gsm, + and g729. Also, an unrelated config reload issue has been fixed + in the case of the config file not changing. (closes issue + #12415) Reported by: pj Patches: + 2009012200_h323packetization.diff.txt uploaded by mvanbaak + (license 7), modified by me + +2009-04-22 16:56 +0000 [r189951] Russell Bryant + + * main/features.c: Fix call parking callback. Pipes -> Commas. + +2009-04-22 16:01 +0000 [r189911] Tilghman Lesher + + * channels/chan_unistim.c: Do not continue to receive DTMF, when + the channel is hungup and about to be destroyed. (closes issue + #14858) Reported by: barryf Patches: 20090421__bug14858.diff.txt + uploaded by tilghman (license 14) Tested by: barryf + +2009-04-22 14:30 +0000 [r189850] Michiel van Baak + + * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 189849 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189849 | mvanbaak | 2009-04-22 16:29:28 +0200 (Wed, 22 Apr 2009) + | 12 lines replace sed with tr to remove \r from downloaded file + On some systems, sed does not recognize \r in the pattern the way + it was used here. Use tr instead because this works the same + across systems. (closes issue #14936) Reported by: leobrown + Patches: 2009042201_14936.diff.txt uploaded by mvanbaak (license + 7) Tested by: leobrown, mvanbaak ........ + +2009-04-22 06:33 +0000 [r189813] Tilghman Lesher + + * configure, configure.ac: Detect liblua on SuSE, and add libm for + linking for Fedora. (Reported via the -dev list, Subject: + Compiling Asterisk with LUA) + +2009-04-21 20:28 +0000 [r189771] David Vossel + + * channels/chan_sip.c: Fixes segfault when switching UDP to TCP in + sip.conf after reload. If transport in sip.conf is switched from + UDP to TCP, Asterisk segfaults right after issuing a sip reload. + The problem is the socket type is changed to TCP but the fd may + still be present for UDP. Later, when the TCP session should be + created or set using an existing one, it isn't because the old + file descriptor is still present. Now every time transport is + changed during a sip.conf reload, the file descriptor is set to + -1, signifying it must be created or found. (closes issue #14727) + Reported by: pj Tested by: dvossel Review: + http://reviewboard.digium.com/r/229/ + +2009-04-21 17:44 +0000 [r189735] Richard Mudgett + + * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, + channels/chan_misdn.c, channels/misdn/chan_misdn_config.h, + channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, + configs/misdn.conf.sample, CHANGES, channels/misdn/isdn_lib.c, + channels/misdn_config.c: Added CCBS/CCNR Party A support and + enhanced COLP support. This change adds the following features to + chan_misdn: * CCBS/CCNR Party A support for PTMP and PTP modes. * + Enhances COLP support for call diversion and explicit call + transfer. These enhanced features require a modified version of + mISDN. The latest modified mISDN v1.1.x based version is + available at: http://svn.digium.com/svn/thirdparty/mISDN/trunk + http://svn.digium.com/svn/thirdparty/mISDNuser/trunk Taged + versions of the modified mISDN code are available under: + http://svn.digium.com/svn/thirdparty/mISDN/tags + http://svn.digium.com/svn/thirdparty/mISDNuser/tags Review: + http://reviewboard.digium.com/r/218/ Merged from + team/rmudgett/misdn_facility branch. + +2009-04-21 15:54 +0000 [r189629-189665] Doug Bailey + + * utils/muted.c, /: Merged revisions 189664 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189664 | dbailey | 2009-04-21 10:52:13 -0500 (Tue, 21 Apr 2009) + | 2 lines Remove daemon call on systems that do not support + forking. ........ + + * /, configure, include/asterisk/autoconfig.h.in, + include/asterisk/compat.h, configure.ac: Merged revisions 189601 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009) + | 3 lines Add check in configure script to check for GLOB_NOMAGIC + and GLOB_BRACE in glob.h This allows config.c to compile when + linked against uclibc that does not support these parameters + ........ + +2009-04-20 22:10 +0000 [r189539] Tilghman Lesher + + * main/stdtime/localtime.c: Use nanosleep instead of poll. This is + not just because mmichelson suggested it, but also because Mac OS + X puked on my poll(). + +2009-04-20 21:29 +0000 [r189495-189516] Terry Wilson + + * apps/app_dial.c, /: Merged revisions 189465 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009) + | 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is + set ........ + + * apps/app_dial.c, /: Merged revisions 189463 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20 Apr 2009) + | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL ........ + +2009-04-20 21:09 +0000 [r189464] Sean Bright + + * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189462 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr + 2009) | 13 lines Properly handle @s within hints in AEL. AEL was + not handling the case of a device hint containing an @ symbol, + which caused parking hints (e.g. hint(park:exten@context)) to + error out the parser. This patch makes AEL treat the @ the same + way it treats colon and ampersand now, meaning the characters are + included in verbatim. (closes issue #14941) Reported by: bpgoldsb + Patches: bug14941.patch uploaded by seanbright (license 71) + Tested by: bpgoldsb ........ + +2009-04-20 19:28 +0000 [r189419] Doug Bailey + + * main/manager.c, /, main/db1-ast/recno/rec_open.c, + channels/chan_iax2.c: Merged revisions 189391 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189391 | dbailey | 2009-04-20 14:10:56 -0500 (Mon, 20 Apr 2009) + | 4 lines Clean up problem with manager implementation of mmap + where it was not testing against MAP_FAILED response. Got rid of + shadowed variable used in processign the mmap results. Change + test of mmap results to compare against MAP_FAILED ........ + +2009-04-20 17:05 +0000 [r189350] Joshua Colp + + * channels/chan_sip.c: Fix a bug with non-UDP connections that + caused dialogs to not get freed. This issue crept up because of a + reference count issue on non-UDP based dialogs. The dialog + reference count was increased when transmitting a packet reliably + but never decreased. This caused the dialog structure to hang + around despite being unlinked from the dialogs container. (closes + issue #14919) Reported by: vrban + +2009-04-20 14:05 +0000 [r189278] Mark Michelson + + * main/channel.c, /: Merged revisions 189277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr + 2009) | 12 lines Move the check for chan->fdno == -1 to after the + zombie/hangup check. Many users were finding that their hung up + channels were staying up and causing 100% CPU usage. (issue + #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch + uploaded by mmichelson (license 60) Tested by: falves11, bamby + ........ + +2009-04-18 01:28 +0000 [r189204] David Vossel + + * /, channels/chan_agent.c: Merged revisions 189203 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 + Apr 2009) | 12 lines Fixed autologoff in agents.conf not working + when agent logs in via AgentLogin app An agent logs in by calling + an extension that calls the AgentLogin app. In agents.conf + ackcall=always is set, so when they get a call they have the + choice to either acknowledge it or ignore it. autologoff=10 is + set as well, so if the agent ignores the call over 10sec one may + assume that the agent should be logged out (and in this case + hungup on as well), but this was not happening. (closes issue + #14091) Reported by: evandro Patches: autologoff.diff uploaded by + dvossel (license 671) Review: + http://reviewboard.digium.com/r/225/ ........ + +2009-04-17 21:48 +0000 [r189137] Richard Mudgett + + * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged + revisions 188833,189134 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) + | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled. + Leave jitter setting alone. JIRA ABE-1835 ........ r189134 | + rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines + Modifed/added some debug messages. JIRA ABE-1835 ........ + +2009-04-17 20:20 +0000 [r189097] Mark Michelson + + * channels/chan_sip.c: Prevent a crash when SIP blonde transferring + an unbridged call. If one attempts to use the attended transfer + button on a SIP phone to transfer an unbridged call (such as a + call to an IVR) but hangs up while the target of the transfer is + still ringing, we need to not crash. The problem was that + ast_hangup was called from outside the channel thread. AST-211 + +2009-04-17 19:36 +0000 [r189077] Sean Bright + + * main/asterisk.c: Fix copy/paste error with 'transmit silence' + flag. + +2009-04-17 15:44 +0000 [r189010] Matthew Nicholson + + * main/pbx.c, /: Merged revisions 189009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr + 2009) | 5 lines Make Busy() application set the CDR disposition + to BUSY. (closes issue #14306) Reported by: cristiandimache + ........ + +2009-04-17 14:44 +0000 [r188947] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 188946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | + 15 lines Fix a bug where a value used to create the channel name + was bogus. This commit fixes the scenario where an incoming call + is authenticated using a peer entry. Previously the channel name + was created using either the username setting from the sip.conf + entry or the IP address that the call came from. Now the channel + name will be created using the peer name itself. This commit will + not change the way the channel name is generated for users or + friends. (closes issue #14256) Reported by: Nick_Lewis Patches: + chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by: + Nick_Lewis, file ........ + +2009-04-17 14:33 +0000 [r188942] Mark Michelson + + * main/pbx.c: Fix a spacing issue that I claimed I would when I + committed this code. Nothing major though. + +2009-04-17 14:26 +0000 [r188938] Joshua Colp + + * channels/chan_dahdi.c, /: Merged revisions 188937 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr + 2009) | 4 lines Fix a situation where the DAHDI channel private + structure lock was not unlocked when it should have been. (issue + AST-210) ........ + +2009-04-17 13:29 +0000 [r188901] Mark Michelson + + * main/pbx.c: Several fixes to the extenpatternmatchnew logic. 1. + Differentiate between literal characters in an extension and + characters that should be treated as a pattern match. Prior to + these fixes, an extension such as NNN would be treated as a + pattern, rather than a literal string of N's. 2. Fixed the logic + used when matching an extension with a bracketed expression, such + as 2[5-7]6. 3. Removed all areas of code that were executed when + NOT_NOW was #defined. The code in these areas had the potential + to crash, for one thing, and the actual intent of these blocks + seemed counterproductive. 4. Fixed many many coding guidelines + problems I encountered while looking through the corresponding + code. 5. Added failure cases and warning messages for when + duplicate extensions are encountered. 6. Miscellaneous fixes to + incorrect or redundant statements. (closes issue #14615) Reported + by: steinwej Tested by: mmichelson Review: + http://reviewboard.digium.com/r/194/ + +2009-04-16 21:57 +0000 [r188774-188836] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 188835 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) + | 7 lines Only update realtime, if global option rtupdate != + false (closes issue #14885) Reported by: deepesh Patches: + 20090413__bug14885.diff.txt uploaded by tilghman (license 14) + Tested by: deepesh ........ + + * /, apps/app_voicemail.c: Merged revisions 188773 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 + Apr 2009) | 4 lines Umask should not be exported into global + namespace. (closes issue #14912) Reported by: jcapp ........ + +2009-04-16 19:30 +0000 [r188742] David Vossel + + * channels/chan_sip.c: SIP state notify reorganization What I've + done here is simply break up how a state NOTIFY is built. + Originally both the XML and sip header information were built + within the same function. While this does work, it does not allow + for the creation of multipart/related message bodies that can + contain multiple XML entries with only one sip header. Now a + separate function builds the XML for each notify. This patch also + makes maintaining and modifying state notifications in the future + much less of a pain. Review: http://reviewboard.digium.com/r/224/ + +2009-04-16 13:42 +0000 [r188705] Joshua Colp + + * channels/chan_dahdi.c: Fix a bug with the dahdi_setoption + callback in chan_dahdi. This function incorrectly reported + success even if the option was unsupported. This was exposed by + the options to change the underlying channel format. The function + now returns a failure if the option is unsupported. + +2009-04-15 22:10 +0000 [r188647] David Vossel + + * channels/chan_dahdi.c, /: Merged revisions 188646 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 + Apr 2009) | 12 lines National prefix inserted even when caller ID + not available When the caller ID is restricted, the expected + behavior is for the caller id to be blank. In chan_dahdi, the + national prefix is placed onto the callers number even if its + restricted (empty) causing the caller id to be the national + prefix rather than blank. (closes issue #13207) Reported by: + shawkris Patches: national_prefix.diff uploaded by dvossel + (license 671) Review: http://reviewboard.digium.com/r/220/ + ........ + +2009-04-15 20:17 +0000 [r188544-188585] Mark Michelson + + * /, main/file.c: Merged revisions 188582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr + 2009) | 7 lines Update ast_readvideo_callback to match + ast_readaudio_callback. This fixes potential refcount errors that + may occur on ast_filestreams. AST-208 ........ + + * apps/app_dial.c: Make the cancellation of the dial timeout on a + call forward optional. This introduces the 'z' option to + app_dial. With it set, a call forward will cancel any timeout + originally set for this instance of the Dial application. AST-207 + +2009-04-15 14:57 +0000 [r188515] Jeff Peeler + + * channels/chan_dahdi.c: Don't try to do anything in + pri_check_restart with service messages unless libpri supports + it. + +2009-04-14 23:28 +0000 [r188470] Mark Michelson + + * apps/app_queue.c: Fix a couple of queue member reference leaks. + +2009-04-14 17:40 +0000 [r188413] Joshua Colp + + * res/res_rtp_asterisk.c: Fix an incorrect clock rate when sending + T140 text. (closes issue #14029) Reported by: epicac + +2009-04-14 16:49 +0000 [r188342-188378] Jeff Peeler + + * channels/chan_dahdi.c, CHANGES: change some capitalization + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure, + include/asterisk/autoconfig.h.in, configure.ac, CHANGES: Add + service maintenance message support This is the companion commit + to libpri r732. Service messages are now supported for switch + types 4ess/5ess. A new option service_message_support has been + added to chan_dahdi.conf and is noted in the sample config file. + The service message support is turned off by default. The current + implementation relies on AstDB to keep track of channel state, + which allows the statuses to be preserved across Asterisk + restarts. Below is a description of the storage format. The state + and reason for the service state are in the form + :, where: ::= { 'O' } // 'O' – Out Of + Service ::= { '0' | '1' | '2' | '3' }, where: '0' – No + reason (backwards compatibility) '1' – NEAR END '2' – FAR END '3' + – both NEAR and FAR END The new CLI commands to handle channel + service state are: pri service disable channel pri service + enable channel Many people contributed to the development + of this functionality. Because I entered at the very end I do not + know the exact history. Special thanks to all who moved the bug + forward one way or another: cmaj, PCadach, markster, mattf, + drmac, MikeJ, serge-v, murf, kanelbullar, Seb7, tilghman, + lmadsen, and especially dhubbard (he answered lots of my + questions and did a large portion of the work) (closes issue + #3450) Reported by: cmaj + +2009-04-14 14:22 +0000 [r188283-188284] Olle Johansson + + * doc/manager_1_1.txt: New actions should go under "New Actions", + not "new events" + + * main/xmldoc.c, apps/app_jack.c: Making sure we have references to + external libraries. Note: Update h.323 with the recent changes + too + +2009-04-14 13:14 +0000 [r188247] Joshua Colp + + * channels/chan_sip.c: Fix a bug with the change I made yesterday + to outbound proxy support. Per discussion with oej on IRC we need + the actual IP address, not the outbound proxy IP address, in the + sa field. This change matches the already existing code for all + other uses of the outbound proxy setting. + +2009-04-14 05:45 +0000 [r188206-188210] Tilghman Lesher + + * main/pbx.c: As suggested by Russell, warn users when their + dialplan arguments contain pipes, but not commas. + + * utils/smsq.c: Application delimiter is ',', not '|'. (closes + issue #14881) Reported by: stegro Patches: smsq.patch uploaded by + stegro (license 752) + +2009-04-13 19:31 +0000 [r188102] Mark Michelson + + * res/res_musiconhold.c: Fix another crash related to cached + realtime music on hold. This was another off-by-one problem + caused by moh_register. + +2009-04-13 16:28 +0000 [r188067] Joshua Colp + + * channels/chan_sip.c: Fix a bug where using an outbound proxy + would cause the local address to be 127.0.0.1. Copy the outbound + proxy IP address into the SIP dialog structure as the IP address + we will be sending to. This has to be done because the logic that + determines what local IP address to use in the SIP messages is + not aware of an outbound proxy being in place. It only knows what + IP address we are sending to. (closes issue #12006) Reported by: + mnicholson + +2009-04-13 14:17 +0000 [r188032] Mark Michelson + + * apps/app_queue.c: Set all queue variables on both the caller and + member channels. This allows for the variables to be accessed if + a member macro is run. Thanks to Grigoriy Puzankin for bringing + this up on the -dev list. + +2009-04-10 20:26 +0000 [r187906] Jeff Peeler + + * channels/Makefile: Fix module embedding for chan_h323. Include + libchanh323.a in the modules.link file so that all the symbols + can be resolved at link time. (closes issue #11966) Reported by: + dome Patches: issue_11966.patch uploaded by kpfleming (license + 421) Tested by: jpeeler + +2009-04-10 18:56 +0000 [r187830] Mark Michelson + + * channels/chan_local.c: Indicating connected line or redirecting + updates were missing a call to lock the local_pvt. + +2009-04-10 18:14 +0000 [r187772-187773] Joshua Colp + + * res/res_rtp_asterisk.c, main/rtp_engine.c: Change how we set the + local and remote address. The code will now only change the + address and port. It will not overwrite any other values. + + * channels/chan_jingle.c, channels/chan_unistim.c, + res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c, + channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_gtalk.c, channels/chan_mgcp.c: Fix some + uninitialized memory notices that appeared under valgrind. + +2009-04-10 17:32 +0000 [r187770] Mark Michelson + + * apps/app_dial.c: Make sure tc is unlocked before calling ast_call + since calling a Local channel could result in a deadlock. + +2009-04-10 17:29 +0000 [r187764] Tilghman Lesher + + * contrib/scripts/realtime_pgsql.sql, /, + contrib/scripts/sip-friends.sql: Merged revisions 187763 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009) + | 2 lines Add lastms column to the contributed table designs + ........ + +2009-04-10 16:51 +0000 [r187721] Kevin P. Fleming + + * build_tools/embed_modules.xml: clean up some patterns for files + to remove add embedding support for bridge and test modules + +2009-04-10 16:26 +0000 [r187680-187714] Mark Michelson + + * channels/chan_local.c: ast_strdup failures aren't really failures + if the original value was NULL. + + * main/channel.c: Don't let ast_channel_alloc fail if explicitly + passed NULL cid_name or cid_number. This also fixes a small + memory leak. + +2009-04-10 16:00 +0000 [r187675] Russell Bryant + + * tests/test_heap.c, tests/test_sched.c: Disable test modules by + default. + +2009-04-10 15:59 +0000 [r187674] Tilghman Lesher + + * channels/chan_sip.c: Ensure pvt is not NULL before dereferencing + it. (closes issue #14784) Reported by: pj + +2009-04-10 15:49 +0000 [r187673] David Vossel + + * apps/app_dial.c: Even more changes concerning r187426. Revised + where locks are placed yet once again. ast_call() should not be + called with a channel locked. could cause deadlock issues with + local channels. + +2009-04-10 15:11 +0000 [r187636] Kevin P. Fleming + + * include/asterisk/logger.h, main/logger.c, apps/app_verbose.c, + configs/logger.conf.sample: revert addition of LOG_SECURITY log + channel; after further discussion, a much better solution will be + used + +2009-04-10 14:53 +0000 [r187634-187635] Richard Mudgett + + * channels/misdn/isdn_lib.h, channels/chan_misdn.c, + channels/misdn/isdn_lib.c: Miscellaneous minor changes to + chan_misdn. * Miscellaneous spacing and comment changes. * Minor + code rearangements. * Miscellaneous doxygen comments. + + * channels/chan_misdn.c: Make chan_misdn_log() avoid generating the + log message if logging is disabled. + +2009-04-10 03:55 +0000 [r187599] Tilghman Lesher + + * main/channel.c, main/pbx.c, main/manager.c, + include/asterisk/linkedlists.h, main/features.c, main/http.c, + main/app.c, include/asterisk/lock.h, main/audiohook.c, + main/bridging.c: Modify headers and macros, according to + Russell's suggestions on the -dev list + +2009-04-09 21:06 +0000 [r187560] Mark Michelson + + * channels/chan_sip.c, configs/sip.conf.sample: Add a new option, + mwi_from, to sip.conf. This allows for you to change the From + header for outgoing MWI NOTIFY requests. Prior to this, the best + you could do was to set a callerid in the general section of + sip.conf. The problem was that this was used for all outbound + requests, not just MWI NOTIFY requests. AST-201 + +2009-04-09 20:40 +0000 [r187556] David Vossel + + * apps/app_dial.c: More changes concerning r187426. Revised where + locks are placed. + +2009-04-09 19:10 +0000 [r187491] Jeff Peeler + + * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: Add + ability for dialplan execution to continue when caller hangs up. + The F option to app_dial has been modified to accept no + parameters and perform the above functionality. I don't see + anywhere else that is doing function overloading, but this really + is the best place for this operation because: - It makes it close + to the 'g' option in the argument list which provides similar + functionality. - The existing code to support the current F + option provides a very convienient location to add this new + feature. (closes issue #12381) Reported by: michael-fig + +2009-04-09 18:58 +0000 [r187488] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 187484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr + 2009) | 18 lines Handle a SIP race condition (reinvite before an + ACK) properly. RFC 5047 explains the proper course of action to + take if a reINVITE is received before the ACK from a previous + invite transaction. What we are to do is to treat the reINVITE as + if it were both an ACK and a reINVITE and process it normally. + Later, when we receive the ACK we had been expecting, we will + ignore it since its CSeq is less than the current iseqno of the + sip_pvt representing this dialog. (closes issue #13849) Reported + by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson + (license 60) Tested by: mmichelson, klaus3000 ........ + +2009-04-09 18:40 +0000 [r187483] Tilghman Lesher + + * main/manager.c, /, include/asterisk/linkedlists.h, + include/asterisk/lock.h: Merged revisions 187428 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 + Apr 2009) | 8 lines Race condition between ast_cli_command() and + 'module unload' could cause a deadlock. Add lock timeouts to + avoid this potential deadlock. (closes issue #14705) Reported by: + jamessan Patches: 20090320__bug14705.diff.txt uploaded by + tilghman (license 14) Tested by: jamessan ........ + +2009-04-09 17:39 +0000 [r187426] David Vossel + + * apps/app_dial.c: Fixes deadlock caused by calling get_cid_name + with chan locked. get_cid_name should not be called with a + channel lock. get_cid_name calls ast_get_hint which eventually + calls pbx_find_extension. pbx_find_extension starts and stops + autoservice which should not be done with a channel lock, so + get_cid_name should not be called with one. + +2009-04-09 17:34 +0000 [r187421-187424] Mark Michelson + + * res/res_musiconhold.c: Use safe macro practices even though they + really aren't necessary. + + * res/res_musiconhold.c: Fix a crash in res_musiconhold when using + cached realtime moh. The moh_register function links an mohclass + and then immediately unrefs the class since the container now has + a reference. The problem with using realtime music on hold is + that the class is allocated, registered, and started in one fell + swoop. The refcounting logic resulted in the count being off by + one. The same problem did not happen when using a static config + because the allocation and registration of an mohclass is a + separate operation from starting moh. This also did not affect + non-cached realtime moh because the classes are not registered at + all. I also have modified res_musiconhold to use the _t_ variants + of the ao2_ functions so that more info can be gleaned when + attempting to trace the refcounts. I found this to be incredibly + helpful for debugging this issue and there's no good reason to + remove it. (closes issue #14661) Reported by: sum + +2009-04-09 17:20 +0000 [r187363-187381] Tilghman Lesher + + * channels/chan_sip.c: Allow '/' in username portion of register; + this is a regression. (closes issue #14668) Reported by: Netview + + * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions + 187362 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) + | 3 lines Permit zero-length text messages in SIP. (Related to an + issue posted to the -users list, subject "AEL2, BASE64_DECODE and + hexadecimal") ........ + +2009-04-09 16:27 +0000 [r187360-187361] Joshua Colp + + * channels/chan_iax2.c: Do not try to send the format read/format + write/make compatible options over IAX2. + + * main/channel.c, channels/chan_sip.c, include/asterisk/frame.h: + Add support for allowing the channel driver to handle + transcoding. This was accomplished using a set of options and the + setoption channel callback. The core calls into the channel + driver using these options and the channel driver either returns + success or failure. + +2009-04-09 04:59 +0000 [r187302] Tilghman Lesher + + * agi/Makefile, build_tools/cflags.xml, utils/Makefile, + include/asterisk.h, /, main/Makefile, main/file.c, main/astfd.c + (added), main/asterisk.c: Merged revisions 187300-187301 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) + | 3 lines Add debugging mode for diagnosing file descriptor + leaks. (Related to issue #14625) ........ r187301 | tilghman | + 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops, + missed this file in the last commit. ........ + +2009-04-09 02:44 +0000 [r187269] Kevin P. Fleming + + * include/asterisk/logger.h, main/logger.c, apps/app_verbose.c, + configs/logger.conf.sample: add a dedicated log channel for + modules to be able report security-related events, so that they + can be fed into external processes for analysis and possible + mitigation efforts (inspired by this evening's Toronto Asterisk + Users Group meeting and previous dicussions amongst various + community members) + +2009-04-08 21:00 +0000 [r187211] Jeff Peeler + + * main/channel.c, main/features.c, include/asterisk/channel.h: Add + timer for features so that backup bridge config can go away The + biggest change done here was elimination of the backup_config for + use with features. Previously, the bridging code upon detecting a + feature would set the start time of the bridge to the start time + of the feature. Then after the feature had either expired or + timed out the start time would be reset to the true bridge start + time from the backup_config. Now, the time differences are + calculated with respect to the newly added feature_start_time + timeval instead. There should be no behavior changes from the + previous functionality aside from the bridge timing being + unaffected by either valid or partial feature matches. Previously + the timing would be increased by the length of time configured + for featuredigittimeout, which was probably never noticed. + (closes issue #14503) Reported by: KNK Tested by: jpeeler Review: + http://reviewboard.digium.com/r/179/ + +2009-04-08 20:39 +0000 [r187210] Tilghman Lesher + + * /: Recorded merge of revisions 187209 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r187209 | tilghman | 2009-04-08 15:39:13 -0500 (Wed, 08 Apr 2009) + | 4 lines Backport resolution for file descriptor leak in 1.6.0 + to 1.4. This fixes short reads in http manager sessions, such as + those done by the ast-gui branch. (Fixes AST-198) ........ + +2009-04-08 19:59 +0000 [r187179] Russell Bryant + + * include/asterisk/doxyref.h, + include/asterisk/doxygen/reviewboard.h (added): Add documentation + for reviewboard usage and guidelines. + +2009-04-08 18:12 +0000 [r187108] Joshua Colp + + * main/rtp_engine.c: Fix a bug where we would native bridge when we + did not want to. + +2009-04-08 17:51 +0000 [r187105] Russell Bryant + + * channels/chan_sip.c: Remove duplicate prototype for temp_peer(). + +2009-04-08 17:08 +0000 [r187050] Tilghman Lesher + + * funcs/func_odbc.c: If the first column is empty, output a + delimiter anyway. (closes issue #14848) Reported by: john8675309 + Patches: 20090408__bug14848.diff.txt uploaded by tilghman + (license 14) Tested by: john8675309 + +2009-04-08 16:52 +0000 [r187046] Mark Michelson + + * /, res/res_musiconhold.c: Merged revisions 187045 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, + 08 Apr 2009) | 10 lines Fix a small logical error when loading + moh classes. We were unconditionally incrementing the number of + mohclasses registered. However, we should actually only increment + if the call to moh_register was successful. While this probably + has never caused problems, I noticed it and decided to fix it + anyway. ........ + +2009-04-08 16:27 +0000 [r187036] Joshua Colp + + * res/res_rtp_asterisk.c, main/rtp_engine.c: Turn a warning message + into a debug message and do not treat two situations as errors + when they are not. + +2009-04-08 15:27 +0000 [r186985] Mark Michelson + + * main/channel.c, /: Merged revisions 186984 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr + 2009) | 24 lines Make a couple of changes with regards to a new + message printed in ast_read(). "ast_read() called with no + recorded file descriptor" is a new message added after a bug was + discovered. Unfortunately, it seems there are a bunch of places + that potentially make such calls to ast_read() and trigger this + error message to be displayed. This commit does two things to + help to make this message appear less. First, the message has + been downgraded to a debug level message if dev mode is not + enabled. The message means a lot more to developers than it does + to end users, and so developers should take an effort to be sure + to call ast_read only when a channel is ready to be read from. + However, since this doesn't actually cause an error in operation + and is not something a user can easily fix, we should not spam + their console with these messages. Second, the message has been + moved to after the check for any pending masquerades. ast_read() + being called with no recorded file descriptor should not + interfere with a masquerade taking place. This could be seen as a + simple way of resolving issue #14723. However, I still want to + try to clear out the existing ways of triggering this message, + since I feel that would be a better resolution for the issue. + ........ + +2009-04-08 13:38 +0000 [r186928-186957] Russell Bryant + + * include/asterisk/doxygen/releases.h: Add some additional notes on + release numbering. + + * Makefile, include/asterisk/doxygen/releases.h (added), + include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen, + include/asterisk/doxygen (added), + include/asterisk/doxygen/commits.h (added), + include/asterisk/doxygen/licensing.h (added), main/asterisk.c: + Start splitting up miscellaneous doxygen documentation into + separate files. doxyref.h was created to hold miscellaneous + documentation that was not specific to a part of the code. This + file has grown quite a bit so I decided to start splitting parts + of it out into new files. Now, you can drop a new file into + include/asterisk/doxygen/ and it will be processed by doxygen. + + * channels/chan_sip.c: Update some comments and resolve potential + memory corruption in chan_sip. While browsing chan_sip the other + day, I noticed this dangerous code in dialog_needdestroy(). This + function is an ao2_callback. It is absolutely _not_ okay to + unlock the container from within this function. It's also not + clear why it was useful. Given that it could cause memory + corruption, I have removed it. There was also a TODO comment left + describing a potential implementation of an improvement to the + needdestroy handling. I'm not convinced that what was described + is the best choice here, so I have briefly described the way that + this function is used today that could be improved. + +2009-04-08 05:06 +0000 [r186899] Tilghman Lesher + + * channels/chan_sip.c: Add lastms to the require API call. + +2009-04-08 00:09 +0000 [r186833-186842] Mark Michelson + + * /, formats/format_wav.c, formats/format_wav_gsm.c: Merged + revisions 186841 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr + 2009) | 8 lines Fix a few typos of the word "frequency." (closes + issue #14842) Reported by: jvandal Patches: frequency-typo.diff + uploaded by jvandal (license 413) ........ + + * channels/chan_sip.c: Fix bad merge from fix for issue 13867. + (closes issue #14686) Reported by: davidw + + * main/channel.c, /: Merged revisions 186832 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr + 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a + p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, + warning sounds will not be properly played to either party of the + bridge. (closes issue #14845) Reported by: adomjan ........ + +2009-04-07 22:23 +0000 [r186799] Tilghman Lesher + + * /, apps/app_macro.c: Merged revisions 186775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) + | 3 lines Fix Macro documentation to match current (and intended) + behavior. (See -dev mailing list) ........ + +2009-04-07 20:46 +0000 [r186720] Mark Michelson + + * main/manager.c, /: Merged revisions 186719 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr + 2009) | 6 lines Ensure that \r\n is printed after the ActionID in + an OriginateResponse. (closes issue #14847) Reported by: kobaz + ........ + +2009-04-06 23:11 +0000 [r186624-186687] Joshua Colp + + * res/res_rtp_asterisk.c: Fix a log message getting output when it + should not have been. + + * channels/chan_sip.c: Fix problem when authenticating a non-RTP + dialog. + + * channels/chan_sip.c, doc/tex/channelvariables.tex, CHANGES: Add + support for changing the outbound codec on a SIP call using a + dialplan variable. This adds a dialplan variable + (SIP_CODEC_OUTBOUND) which controls the codec offered for an + outgoing SIP call. This is much like the SIP_CODEC dialplan + variable and has the same restrictions. The codec set must be one + that is configured for the call. (closes issue #13243) Reported + by: samdell3 Patches: 13243.diff uploaded by file (license 11) + +2009-04-06 16:06 +0000 [r186620] Mark Michelson + + * funcs/func_connectedline.c (added), funcs/func_redirecting.c + (added): Silly svn. These files didn't get merged over in the + merge of the issue8824 branch. + +2009-04-06 13:23 +0000 [r186563] Joshua Colp + + * main/rtp_engine.c: Pass the correct value to sizeof when copying + address information. (issue #14827) Reported by: pj Patches: + 14827.diff uploaded by file (license 11) Tested by: pj + +2009-04-04 00:13 +0000 [r186537] Richard Mudgett + + * /: Remove merged branch properties accidentally merged to trunk. + +2009-04-03 22:41 +0000 [r186525] Mark Michelson + + * channels/chan_unistim.c, channels/misdn/isdn_lib_intern.h, + channels/chan_local.c, main/rtp_engine.c, /, + channels/misdn/isdn_msg_parser.c, channels/chan_iax2.c, + channels/misdn/isdn_lib.c, channels/misdn_config.c, + include/asterisk/callerid.h, main/channel.c, main/dial.c, + channels/misdn/isdn_lib.h, channels/chan_dahdi.c, + channels/chan_phone.c, channels/chan_skinny.c, main/features.c, + configs/sip.conf.sample, include/asterisk/frame.h, + include/asterisk/rtp_engine.h, channels/chan_mgcp.c, + apps/app_dial.c, res/res_rtp_asterisk.c, main/stun.c, + channels/chan_sip.c, channels/chan_agent.c, + configs/misdn.conf.sample, include/asterisk/channel.h, CHANGES, + apps/app_queue.c, channels/chan_misdn.c, + apps/app_directed_pickup.c, channels/misdn/chan_misdn_config.h, + channels/chan_h323.c, main/callerid.c, include/asterisk/stun.h: + This commit introduces COLP/CONP and Redirecting party + information into Asterisk. The channel drivers which have been + most heavily tested with these enhancements are chan_sip and + chan_misdn. Further work is being done to add Q.SIG support and + will be introduced in a later commit. chan_skinny has code added + to it here, but according to user pj, the support on chan_skinny + is not working as of now. This will be fixed in a later commit. A + special thanks goes out to bugtracker user gareth for getting the + ball rolling and providing the initial support for this work. + Without his initial work on this, this would not have been nearly + as painless as it was. This functionality has been tested by + Digium's product quality department, as well as a customer site + running thousands of calls every day. In addition, many many many + many bugtracker users have tested this, too. (closes issue #8824) + Reported by: gareth Review: http://reviewboard.digium.com/r/201 + +2009-04-03 20:20 +0000 [r186461] Kevin P. Fleming + + * channels/chan_dahdi.c, /: Merged revisions 186458 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 + Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would + not properly switch formats when requested Don't offer + AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could + provide a slight performance benefit, the translation core in + Asterisk has some flaws when a channel driver offers multiple raw + formats. this fix is much simpler than fixing the translation + core to solve that issue (although that will be done later). + ........ + +2009-04-03 19:59 +0000 [r186444-186447] Tilghman Lesher + + * /, apps/app_voicemail.c: Merged revisions 186445 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 + Apr 2009) | 2 lines Found a conflict in the last commit, due to + multiple targets ........ + + * /, configs/voicemail.conf.sample, apps/app_voicemail.c: Merged + revisions 186415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) + | 7 lines Distinguish in a sent email between simple sends and + forwards. (closes issue #11678) Reported by: jamessan Patches: + 20090330__bug11678.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman, lmadsen ........ + +2009-04-03 16:47 +0000 [r186382] Joshua Colp + + * main/channel.c, channels/chan_sip.c, channels/chan_iax2.c, + include/asterisk/frame.h: Add better support for relaying success + or failure of the ast_transfer() API call. This API call now + waits for a special frame from the underlying channel driver to + indicate success or failure. This allows the return value to + truly convey whether the transfer worked or not. In the case of + the Transfer() dialplan application this means the value of the + TRANSFERSTATUS dialplan variable is actually true. (closes issue + #12713) Reported by: davidw Tested by: file + +2009-04-03 16:29 +0000 [r186379] David Vossel + + * main/audiohook.c: audio_audiohook_write_list() did not correctly + update sample size after ast_translate. + audio_audiohook_write_list() did not take into account that the + sample size may change after translation depending on if the + original frame is is 8khz or 16khz. the sample size is now + updated after translating to reflect this possibility. This + caused the audio on the receiving end to sound terrible. Thanks + to jcolp and mmichelson for helping me work this out. (issue + AST-197) + +2009-04-03 15:52 +0000 [r186321] Joshua Colp + + * include/asterisk/crypto.h, /: Merged revisions 186320 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 + lines Fix a problem with the crypto variable definitions not + actually being defined properly. (closes issue #14804) Reported + by: jvandal ........ + +2009-04-03 15:18 +0000 [r186297] Tilghman Lesher + + * main/stdtime/localtime.c: Compatibility fix for glibc 2.4 (Closes + issue #14820) Reported by: phsultan + +2009-04-03 14:32 +0000 [r186286] Mark Michelson + + * apps/app_voicemail.c: Fix the ability to retrieve voicemail + messages from IMAP. A recent change made interactive vm_states no + longer get added to the list of vm_states and instead get stored + in thread-local storage. In trunk and all the 1.6.X branches, the + problem is that when we search for messages in a voicemail box, + we would attempt to update the appropriate vm_state struct by + directly searching in the list of vm_states instead of using the + get_vm_state_by_imap_user function. This meant we could not find + the interactive vm_state that we wanted. (closes issue #14685) + Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson + (license 60) Tested by: BlargMaN, qualleyiv, mmichelson + +2009-04-03 02:03 +0000 [r186230] Russell Bryant + + * /, cdr/cdr_radius.c: Merged revisions 186229 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) + | 21 lines Fix a memory leak in cdr_radius. I came across this + while doing some testing of my ast_channel_ao2 branch. After + running a test overnight that generated over 5 million calls, + Asterisk had taken up about 1 GB of my system memory. So, I + re-ran the test with MALLOC_DEBUG turned on. However, it showed + no leaks in Asterisk during the test, even though Asterisk was + still consuming it somehow. Instead, I turned to valgrind, which + when run with --leak-check=full, told me exactly where the leak + came from, which was from allocations inside the radiusclient-ng + library. This explains why MALLOC_DEBUG did not report it. After + a bit of analysis, I found that we were leaking a little bit of + memory every time a CDR record was passed to cdr_radius. I don't + actually have a radius server set up to receive CDR records. + However, I always have my development systems compile and install + all modules. In addition to making sure there are not build + errors across modules, always loading modules helps find bugs + like this, too, so it is strongly recommend for all developers. + ........ + +2009-04-02 21:56 +0000 [r186175] Mark Michelson + + * /, configs/features.conf.sample: Merged revisions 186174 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr + 2009) | 5 lines Fix instructions in one-step parking comment to + make more sense. Changed a capital K to a lowercase k. ........ + +2009-04-02 17:26 +0000 [r186101] Kevin P. Fleming + + * channels/chan_dahdi.c, /: Merged revisions 186081 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 + Apr 2009) | 3 lines ensure that the buffer passed to + DAHDI_SET_BUFINFO is fully initialized ........ + +2009-04-02 17:20 +0000 [r186078] Joshua Colp + + * res/res_rtp_asterisk.c (added), channels/chan_unistim.c, + apps/app_dial.c, main/stun.c (added), main/rtp_engine.c (added), + channels/chan_local.c, channels/chan_sip.c, + channels/chan_bridge.c, main/Makefile, channels/chan_agent.c, + include/asterisk/rtp.h (removed), UPGRADE.txt, + channels/chan_gtalk.c, include/asterisk/_private.h, main/rtp.c + (removed), main/loader.c, channels/chan_jingle.c, + channels/chan_skinny.c, channels/chan_h323.c, + configs/sip.conf.sample, include/asterisk/stun.h (added), + include/asterisk/rtp_engine.h (added), main/asterisk.c, + channels/chan_mgcp.c: Merge in the RTP engine API. This API + provides a generic way for multiple RTP stacks to be integrated + into Asterisk. Right now there is only one present, + res_rtp_asterisk, which is the existing Asterisk RTP stack. + Functionality wise this commit performs the same as previously. + API documentation can be viewed in the rtp_engine.h header file. + Review: http://reviewboard.digium.com/r/209/ + +2009-04-02 17:10 +0000 [r186021-186060] Tilghman Lesher + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 186059 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 + (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 + Apr 2009) | 2 lines Fix for AST-2009-003 ........ + ................ + + * main/strings.c: Missed a common case for needing to extend the + buffer. (closes issue #14716) Reported by: sum Patches: + 20090402__bug14716.diff.txt uploaded by tilghman (license 14) + Tested by: sum + +2009-04-02 13:51 +0000 [r185953] Kevin P. Fleming + + * channels/chan_dahdi.c, /: Merged revisions 185952 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 + Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and + DAHDI_GET_PARAMS ioctls were recently corrected to show that they + do, in fact, read data from userspace as part of their work. due + to this fix, valgrind now reports a number of cases where + chan_dahdi passed an uninitialized (or partially) buffer to these + ioctls, which could lead to unexpected behavior. this patch + corrects chan_dahdi to ensure that buffers passed to these ioctls + are always fully initialized. ........ + +2009-04-01 20:13 +0000 [r185912] Tilghman Lesher + + * include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c, + main/manager.c, main/tdd.c, include/asterisk/astobj2.h, + main/ast_expr2f.c, include/asterisk/pbx.h, + include/asterisk/strings.h, main/taskprocessor.c, res/res_odbc.c: + Merge changes from str_substitution that are unrelated to that + branch. Included is a small bugfix to an ast_str helper, but most + of these changes are simply doxygen fixes. + +2009-04-01 19:03 +0000 [r185846] David Vossel + + * /, channels/chan_sip.c: Merged revisions 185845 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) + | 10 lines Fixes issue with dropped calles due to re-Invite glare + and re-Invites never executing after a 491 Acknowledgement for + 491 responses were never being processed because it didn't match + our pending invite's seqno. Since the ACK was never processed, + the 491 frame would continue to be retransmitted until eventually + the call was dropped due to max retries. Now during a pending + invite, if we receive another invite, we send an 491 and hold on + to that glare invite's seqno in the "glareinvite" variable for + that sip_pvt struct. When ACK's are received, we first check to + see if it is in response to our pending invite, if not we check + to see if it is in response to a glare invite. In this case, it + is in response to the glare invite and must be dealt with or the + call is dropped. I've changed the wait time for resending the + re-Invite after receving a 491 response to comply with RFC 3261. + Before this patch the scheduled re-Invite would only change a + flag indicating that the re-Invite should be sent out, now it + actually sends it out as well. (closes issue #12013) Reported by: + alx Review: http://reviewboard.digium.com/r/213/ ........ + +2009-04-01 13:59 +0000 [r185777] Mark Michelson + + * main/manager.c: Address Russell's comments regarding rev 185704. + Use ast_debug and ast_softhangup_nolock. + +2009-04-01 13:48 +0000 [r185741-185772] Russell Bryant + + * main/channel.c, /: Merged revisions 185771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) + | 6 lines Fix a case where DTMF could bypass audiohooks. This + change fixes a situation where an audiohook that wants DTMF would + not actually get it. This is in the code path where we end DTMF + digit length emulation while handling a NULL frame. ........ + + * include/asterisk/stringfields.h: Fix dev-mode build on my box. + +2009-04-01 00:39 +0000 [r185704] Mark Michelson + + * main/manager.c, CHANGES: Allow the AMI Hangup command to accept a + Cause header. (closes issue #14695) Reported by: mneuhauser + Patches: cause-for-hangup-manager-action.patch uploaded by + mneuhauser (license 425) + +2009-03-31 22:35 +0000 [r185664] Kevin P. Fleming + + * utils: ignore copied (generated) file + +2009-03-31 22:12 +0000 [r185600-185604] Mark Michelson + + * apps/app_queue.c: Fix trunk's compilation. + + * /, apps/app_queue.c: Merged revisions 185599 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar + 2009) | 6 lines Fix crash that would occur if an empty member was + specified in queues.conf. (closes issue #14796) Reported by: pida + ........ + +2009-03-31 21:29 +0000 [r185581] Kevin P. Fleming + + * main/utils.c, include/asterisk/stringfields.h: Optimizations to + the stringfields API This patch provides a number of + optimizations to the stringfields API, focused around saving (not + wasting) memory whenever possible. Thanks to Mark Michelson for + inspiring this work and coming up with the first two + optimizations that are represented here: Changes: - Cleanup of + some code, fix incorrect doxygen comments - When a field is + emptied or replaced with a new allocation, decrease the amount of + 'active' space in the pool it was held in; if that pool reaches + zero active space, and is not the current pool, then free it as + it is no longer in use - When allocating a pool, try to allocate + a size that will fit in a 'standard' malloc() allocation without + wasting space - When allocating space for a field, store the + amount of space in the two bytes immediately preceding the field; + this eliminates the need to call strlen() on the field when + overwriting it, and more importantly it 'remembers' the amount of + space the field has available, even if a shorter string has been + stored in it since it was allocated - Don't automatically double + the size of each successive pool allocated; it's wasteful + http://reviewboard.digium.com/r/165/ + +2009-03-31 19:46 +0000 [r185469] Mark Michelson + + * /, apps/app_voicemail.c: Merged revisions 185468 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, + 31 Mar 2009) | 8 lines Fix Russian voicemail intro to say the + word "messages" properly. (closes issue #14736) Reported by: + chappell Patches: voicemail_no_messages.diff uploaded by chappell + (license 8) ........ + +2009-03-31 19:07 +0000 [r185432] Russell Bryant + + * channels/chan_iax2.c: Improve performance of the code handling + the frame queue in chan_iax2. In my tests that exercised full + frame handling in chan_iax2, the version with these changes took + 30% to 40% of the CPU time compared to the same test of Asterisk + trunk before these modifications. While doing some profiling for + , one function that caught + my eye was network_thread() in chan_iax2.c. After the things that + I was working on there, it was the next target for analysis and + optimization. I used oprofile's source annotation functionality + and found that the loop traversing the frame queue in + network_thread() was to blame for the excessive CPU cycle + consumption. The frame_queue in chan_iax2 previously held all + frames that either were pending transmission or had been + transmitted and are still pending acknowledgment. In + network_thread(), the previous code would go back through the + main for loop after reading a single incoming frame or after + being signaled because a frame had been queued up for initial + transmission. In each iteration of the loop, it traverses the + entire frame queue looking for frames that need to be + transmitted. On a busy server, this could easily be quite a few + entries. This patch is actually quite simple. The frame_queue has + become only a list of frames pending acknowledgment. Frames that + need to be transmitted are queued up to a dedicated transmit + thread via the taskprocessor API. As a result, the code in + network_thread() becomes much simpler, as its only job is to read + incoming frames. In addition to the previously described changes, + this patch includes some additional changes to the frame_queue. + Instead of one big frame_queue, now there is a list per call + number to further reduce wasted list traversals. The biggest + impact of this change is in socket_process(). For additional + details on testing and test results, see the review request. + Review: http://reviewboard.digium.com/r/212/ + +2009-03-31 16:46 +0000 [r185363] David Brooks + + * /, channels/chan_gtalk.c: Merged revisions 185362 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 + Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when + xmpp contains extra whitespaces To drill into the xmpp to find + the capabilities between channels, chan_gtalk calls iks_child() + and iks_next(). iks_child() and iks_next() are functions in the + iksemel xml parsing library that traverse xml nodes. The bug here + is that both iks_child() and iks_next() will return the next + iks_struct node *regardless* of type. chan_gtalk expects the next + node to be of type IKS_TAG, which in most cases, it is, but in + this case (a call being made from the Empathy IM client), there + exists iks_struct nodes which are not IKS_TAG data (they are + extraneous whitespaces), and chan_gtalk doesn't handle that case, + so capabilities don't match, and a call cannot be made. + iks_first_tag() and iks_next_tag(), on the other hand, will not + return the very next iks_struct, but will check to see if the + next iks_struct is of type IKS_TAG. If it isn't, it will be + skipped, and the next struct of type IKS_TAG it finds will be + returned. This assures that chan_gtalk will find the iks_struct + it is looking for. This fix simply changes all calls to + iks_child() and iks_next() to become calls to iks_first_tag() and + iks_next_tag(), which resolves the capability matching. The + following is a payload listing from Empathy, which, due to the + extraneous whitespace, will not be parsed correctly by iksemel: + + + Review: http://reviewboard.digium.com/r/181/ + ........ + +2009-03-31 14:53 +0000 [r185261] Russell Bryant + + * apps/app_queue.c: Don't free() an astobj2 object. (closes issue + #14672) Reported by: makoto + +2009-03-31 14:07 +0000 [r185197] Joshua Colp + + * /, main/audiohook.c: Merged revisions 185196 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 + lines Fix crash when moving audiohooks between channels. Handle + the scenario where we are called to move audiohooks between + channels and the source channel does not actually have any on it. + (closes issue #14734) Reported by: corruptor ........ + +2009-03-30 20:42 +0000 [r185122-185123] Richard Mudgett + + * /, configs/misdn.conf.sample, channels/misdn_config.c: Merged + revisions 185121 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) + | 1 line Update the channel allocation method documentation. + ........ + + * /, channels/misdn/isdn_lib.c: Merged revisions 185120 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) + | 19 lines Make chan_misdn BRI TE side normally defer channel + selection to the NT side. Channel allocation collisions are not + handled by chan_misdn very well. This patch simply avoids the + problem for BRI only. For PRI, allocation collisions are still + possible but less likely since there are simply more channels + available and each end could use a different allocation strategy. + misdn.conf options available: te_choose_channel - Use to force + the TE side to allocate channels. method - Specify the channel + allocation strategy. (closes issue #13488) Reported by: + Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich + Tested by: crich, siepkes, festr ........ + +2009-03-30 16:26 +0000 [r185072] Mark Michelson + + * /, apps/app_queue.c: Merged revisions 185031 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar + 2009) | 39 lines Fix queue weight behavior so that calls in + low-weight queues are not inappropriately blocked. (This is + copied and pasted from the review request I made for this patch) + Asterisk has some odd behavior when queue weights are used. The + current logic used when potentially calling a queue member is: If + the member we are going to call is part of another queue and + _that other queue has any callers in it_ and has a higher weight + than the queue we are calling from, then don't try to contact + that member. The issue here is what I have marked with + underscores. If the higher-weighted queue has any callers in it + at all, then the queue member will be unreachable from the + lower-weighted queue. This has the potential to be really really + bad if using a queue strategy, such as leastrecent or + fewestcalls, with the potential to call the same member + repeatedly. The fix proposed by garychen on issue 13220 is very + simple and, as far as I can see, works well for this situation. + With this set of changes, the logic used becomes: If the member + we are going to call is part of another queue, the other queue + has a higher weight than the queue we are calling from, and the + higher weight queue has at least as many callers as available + members, then do not try to contact the queue member. If the + higher weighted queue has fewer callers than available members, + then there is no reason to deny the call to this member since the + other queue can afford to spare a member. Since the fix involved + writing a generic function for determining the number of + available members in the queue, I also modified the is_our_turn + function to make use of the new num_available_members function to + determine if it is our turn to try calling a member. There is one + small behavior change. Before writing this patch, if you had + autofill disabled, then if you were the head caller in a queue, + you would automatically be told that it was your turn to try + calling a member. This did not take into account whether there + were actually any queue members available to take the call. Now + we actually make sure there is at least one member available to + take the call if autofill is disabled. (closes issue #13220) + Reported by: garychen Review: + http://reviewboard.digium.com/r/202/ ........ + +2009-03-30 14:37 +0000 [r184948] Joshua Colp + + * /, channels/chan_sip.c: Merged revisions 184947 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | + 14 lines Improve our handling of T38 in the initial INVITE from a + device. We now answer with matching media streams to what is + requested. If an INVITE is received with both a T38 and RTP media + stream this means we answer with both. For any outgoing calls + created as a result of this inbound one no T38 is requested in + the initial INVITE. Instead if we start receiving udptl packets + we trigger a reinvite on the outbound side. (closes issue #12437) + Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu + Review: http://reviewboard.digium.com/r/208/ ........ + +2009-03-30 13:55 +0000 [r184910] Russell Bryant + + * channels/h323/Makefile.in: Fix build error when chan_h323 is not + being built. (reported by cai1982 in #asterisk-dev) + +2009-03-29 05:52 +0000 [r184838-184843] Russell Bryant + + * /, apps/app_followme.c: Merged revisions 184842 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) + | 5 lines Ensure targs variable is fully initialized. (closes + issue #14758) Reported by: tim_ringenbach ........ + + * channels/Makefile: Simplify chan_h323 build to not require a + second run of "make". (closes issue #14715) Reported by: jthurman + Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman + (license 614) Tested by: tzafrir, russell + +2009-03-27 20:08 +0000 [r184798-184801] Leif Madsen + + * apps/app_ices.c: Fix a typo in app_ices. (closes issue #14765) + Reported by: timeshell Patches: app_ices.svn-1.6.0.diff uploaded + by timeshell (license 399) + + * include/asterisk/doxyref.h: Update commit message guidelines in + re: to punctuation. The doxygen documentation has now been + updated to state explicitly that I want punctuation atthe end of + the first sentence in a commit message. :). + +2009-03-27 19:10 +0000 [r184762] Kevin P. Fleming + + * main/channel.c, bridges/bridge_softmix.c, + include/asterisk/timing.h, include/asterisk/channel.h, + channels/chan_iax2.c, main/timing.c: Improve timing interface to + remember which provider provided a timer The ability to + load/unload timing interfaces is nice, but it means that when a + timer is allocated, it may come from provider A, but later + provider B becomes the 'preferred' provider. If this happens, all + timer API calls on the timer that was provided by provider A will + actually be handed to provider B, which will say WTF and return + an error. This patch changes the timer API to include a pointer + to the provider of the timer handle so that future operations on + the timer will be forwarded to the proper provider. (closes issue + #14697) Reported by: moy Review: + http://reviewboard.digium.com/r/211/ + +2009-03-27 18:04 +0000 [r184693-184726] Russell Bryant + + * main/manager.c, apps/app_minivm.c: Use ast_random() instead of + rand() to ensure we use the best RNG available. + + * include/asterisk/app.h, apps/app_dumpchan.c, main/app.c, + apps/app_queue.c, apps/app_voicemail.c, main/cli.c: Change + global_app_buf to ast_str_thread_global_buf. + +2009-03-27 15:57 +0000 [r184639-184677] Joshua Colp + + * bridges/bridge_softmix.c: Fix a potential timer leak in + bridge_softmix. It is possible for a bridge to be created without + actually being used. In that scenario a timing file descriptor + would be opened and not closed. To fix this the timing file + descriptor is now closed in the destroy callback, not the thread + function. + + * res/res_agi.c: Fix speech structure leak in the AGI speech + recognition integration. The AGI dialplan applications did not + destroy the speech structure automatically if it was not + destroyed by the running AGI script. They will now do this. + (issue LUMENVOX-15) + + * bridges/bridge_softmix.c: Remove a cast that is not needed. + +2009-03-27 14:00 +0000 [r184630] Russell Bryant + + * include/asterisk/utils.h, main/pbx.c, res/ais/evt.c, + main/event.c, pbx/pbx_dundi.c, main/asterisk.c: Change g_eid to + ast_eid_default. + +2009-03-27 13:57 +0000 [r184566-184628] Joshua Colp + + * bridges/bridge_softmix.c: Fix a potential race condition when + creating a software based mixing bridge. It was possible for no + timer to become available between creating the bridge and + starting it. We now open a timer when creating it and keep it + open until the bridge is destroyed. + + * /, channels/chan_sip.c: Merged revisions 184565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 + lines Fix an issue where nat=yes would not always take effect for + the RTP session on outgoing calls. If calls were placed using an + IP address or hostname the global nat setting was copied over but + was not set on the RTP session itself. This caused the RTP stack + to not perform symmetric RTP actions. (closes issue #14546) + Reported by: acunningham ........ + +2009-03-27 02:20 +0000 [r184512-184531] Russell Bryant + + * include/asterisk/lock.h: Fix some issues with rwlock corruption + that caused deadlock like symptoms. When dvossel and I were doing + some load testing last week, we noticed that we could make + Asterisk trunk lock up instantly when we started generating a + bunch of calls. The backtraces of locked threads were bizarre, + and many were stuck on an _unlock_ of an rwlock. The changes are: + 1) Fix a number of places where a backtrace would be loaded into + an invalid index of the backtrace array. It's an off by one + error, which ends up writing over the rwlock itself. 2) Ensure + that in the array of held locks, we NULL out an index once it is + not being used so that it's not confusing when analyzing its + contents. 3) Remove a bunch of logging referring to an rwlock + operating being done with "deep reentrancy". It is normal for + _many_ threads to hold a read lock on an rwlock. + + * main/file.c: Don't act surprised if we get a -1 indication. + + * main/heap.c, include/asterisk/heap.h: Pass more useful + information through to lock tracking when DEBUG_THREADS is on. + +2009-03-26 22:18 +0000 [r184448] Kevin P. Fleming + + * /, sounds/Makefile: Merged revisions 184447 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar + 2009) | 3 lines use new, improved 8kHz prompts ........ + +2009-03-26 21:09 +0000 [r184389] David Vossel + + * /, apps/app_test.c: Merged revisions 184388 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) + | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF + 8 app_test was failing when sending the last DTMF digit, 8, + because of the 100ms pause issued after DTMF is sent. During this + pause the other side would hang up causing the test to look like + it failed. Now the other side waits a second before hanging up. + (closes issue #12442) Reported by: tzafrir ........ + +2009-03-25 22:11 +0000 [r184339-184344] Russell Bryant + + * main/event.c: Remove unneeded AST_LIST_ENTRY() and comment on the + purpose of ast_event_ref. + + * channels/chan_unistim.c, channels/chan_dahdi.c, + include/asterisk/devicestate.h, include/asterisk/event.h, + channels/chan_sip.c, apps/app_minivm.c, res/ais/evt.c, + main/devicestate.c, main/event.c, include/asterisk/_private.h, + include/asterisk/strings.h, channels/chan_iax2.c, + main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c: + Improve performance of the ast_event cache functionality. This + code comes from svn/asterisk/team/russell/event_performance/. + Here is a summary of the changes that have been made, in order of + both invasiveness and performance impact, from smallest to + largest. 1) Asterisk 1.6.1 introduces some additional logic to be + able to handle distributed device state. This functionality comes + at a cost. One relatively minor change in this patch is that the + extra processing required for distributed device state is now + completely bypassed if it's not needed. 2) One of the things that + I noticed when profiling this code was that a _lot_ of time was + spent doing string comparisons. I changed the way strings are + represented in an event to include a hash value at the front. So, + before doing a string comparison, we do an integer comparison on + the hash. 3) Finally, the code that handles the event cache has + been re-written. I tried to do this in a such a way that it had + minimal impact on the API. I did have to change one API call, + though - ast_event_queue_and_cache(). However, the way it works + now is nicer, IMO. Each type of event that can be cached (MWI, + device state) has its own hash table and rules for hashing and + comparing objects. This by far made the biggest impact on + performance. For additional details regarding this code and how + it was tested, please see the review request. (closes issue + #14738) Reported by: russell Review: + http://reviewboard.digium.com/r/205/ + +2009-03-25 19:22 +0000 [r184280] Joshua Colp + + * channels/chan_sip.c: Fix issue with a T38 reinvite being sent + even if not configured to do so. If we receive a T38 request + negotiate control frame we should only attempt to do so if the + option is enabled on the dialog. + +2009-03-25 14:38 +0000 [r184220] Eliel C. Sardanons + + * /, main/asterisk.c: Merged revisions 184188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | + 13 lines Avoid destroying the CLI line when moving the cursor + backward and trying to autocomplete. When moving the cursor + backward and pressing TAB to autocomplete, a NULL is put in the + line and we are loosing what we have already wrote after the + actual cursor position. (closes issue #14373) Reported by: eliel + Patches: asterisk.c.patch uploaded by eliel (license 64) Tested + by: lmadsen ........ + +2009-03-25 14:33 +0000 [r184147-184219] Russell Bryant + + * main/timing.c: Include poll-compat.h + + * main/timing.c: Change poll() to ast_poll(). + + * utils/Makefile, include/asterisk/compat.h: Fix build issues on + Mac OSX. (closes issue #14714) Reported by: ygor + +2009-03-24 22:40 +0000 [r184079] Mark Michelson + + * /, apps/app_senddtmf.c: Merged revisions 184078 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar + 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. + The 'digit' variable is guaranteed to be non-NULL, so the if + statement could never evaluate true. Changing to ast_strlen_zero + makes the logic correct. This was found while reviewing + ast_channel_ao2 code review. ........ + +2009-03-24 22:00 +0000 [r184037-184043] Russell Bryant + + * main/channel.c: Put siren7 and siren14 in ast_best_codec() just + so they're in there somewhere. + + * channels/chan_iax2.c: Exclude slin16, siren7, and siren14 from + bandwidth=low and =medium The default codec configuration for + chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as + the codec in some test calls, but that no longer happens after + this change. + +2009-03-24 20:01 +0000 [r183995] David Vossel + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: SIP + preferred codec only feature Added an option to respond to a SIP + invite with only the single most preferred joint codec. This + limits the options of what codecs the other side can use. (closes + issue #12485) Reported by: bamby Review: + http://reviewboard.digium.com/r/206/ + +2009-03-24 15:26 +0000 [r183865-183914] Tilghman Lesher + + * /, configs/voicemail.conf.sample: Merged revisions 183913 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) + | 3 lines Additionally note that the operator option needs an 'o' + extension. (Related to issue #14731) ........ + + * main/http.c: Allow browsers to cache images and other static + content. + +2009-03-23 22:35 +0000 [r183831] Richard Mudgett + + * channels/chan_misdn.c, channels/misdn/Makefile, + channels/misdn/chan_misdn_config.h, channels/misdn/ie.c, + channels/misdn/isdn_msg_parser.c, channels/misdn/portinfo.c, + channels/misdn/isdn_lib.c, channels/misdn_config.c: Removed + trailing whitespace in chan_misdn files. + +2009-03-23 18:58 +0000 [r183766] Mark Michelson + + * /, res/res_monitor.c: Merged revisions 183700 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar + 2009) | 7 lines Fix a memory leak in res_monitor.c The only way + that this leak would occur is if Monitor were started using the + Manager interface and no File: header were given. Discovered + while reviewing the ast_channel_ao2 review request. ........ + +2009-03-23 18:06 +0000 [r183701] Leif Madsen + + * channels/chan_dahdi.c: Fixes a documentation error introduced + during the CLI cleanup at AstriDevCon 2008. (closes issue #14655) + Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic + (license 728) Tested by: lmadsen + +2009-03-22 21:00 +0000 [r183652] Joshua Colp + + * main/bridging.c: Fix a minor logic flaw with the bridge generic + thread. We only want to move the channel pointers that are + actually present. + +2009-03-20 17:00 +0000 [r183560] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 183559 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 + Mar 2009) | 2 lines Fix a crash in IAX2 registration handling + found during load testing with dvossel. ........ + +2009-03-20 16:25 +0000 [r183553-183555] Mark Michelson + + * channels/chan_sip.c: Fix chan_sip so it builds. + + * include/asterisk/rtp.h, main/rtp.c, main/asterisk.exports: Remove + symbols I just added to main/asterisk.exports and instead rename + the functions. + + * main/asterisk.exports: Add some missing symbols to + main/asterisk.exports Hey! chan_sip.so loads now! + +2009-03-20 12:12 +0000 [r183511] Eliel C. Sardanons + + * channels/chan_dahdi.c: Remove duplicate inside the + xml documentation. + +2009-03-19 20:30 +0000 [r183436] David Vossel + + * apps/app_dial.c, /, main/features.c, include/asterisk/features.h: + Merged revisions 183386 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) + | 6 lines Cleaning up a few things in detect disconnect patch + Initialized ast_call_feature in detect_disconnect to avoid + accessing uninitialized memory. Cleaned up /param tags in + features.h. No longer send dynamic features in + ast_feature_detect. issue #11583 ........ + +2009-03-19 19:22 +0000 [r183321-183345] Tilghman Lesher + + * /: Recorded merge of revisions 183342 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183342 | tilghman | 2009-03-19 14:21:30 -0500 (Thu, 19 Mar 2009) + | 2 lines Reordering, to change prior to unlocking ........ + + * channels/chan_dahdi.c, /: Merged revisions 183319 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 + Mar 2009) | 8 lines Delay signalling progress until a PRI channel + really signals progress. (closes issue #13034) Reported by: + klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by + tilghman (license 14) patch_trunk_183progress_klaus3000.txt + uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ + +2009-03-19 18:34 +0000 [r183312] Jason Parker + + * /, main/asterisk.exports: Merged revisions 183291 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r183291 | qwell | 2009-03-19 13:28:16 -0500 (Thu, 19 Mar + 2009) | 1 line Export some more required symbols. ........ + +2009-03-19 18:10 +0000 [r183244] Mark Michelson + + * apps/app_queue.c: Fix a memory leak associated with queues. For + every attempt that app_queue made to place an outbound call to a + queue member, we would allocate a queue_end_bridge structure. + When the bridge for the call had completed, we would free the + structure. Unfortunately not all call attempts actually end up + bridged to a member, so we need to be more selective of when to + allocate the structure. With this change, the allocation occurs + in an area where we can guarantee that the call will be bridged. + (closes issue #14680) Reported by: caspy Patches: 14680.patch + uploaded by mmichelson (license 60) Tested by: caspy + +2009-03-19 18:00 +0000 [r183239-183242] Russell Bryant + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + main/loader.c: Merged revisions 183241 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) + | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving + like expected. ........ + + * /, main/asterisk.exports: Merged revisions 183238 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r183238 | russell | 2009-03-19 12:41:39 -0500 (Thu, 19 + Mar 2009) | 1 line Allow the AES API to work. ........ + +2009-03-19 17:00 +0000 [r183196] Tilghman Lesher + + * res/res_odbc.exports: 2 symbols defined when DEBUG_THREADS + +2009-03-19 16:28 +0000 [r183172] David Vossel + + * apps/app_dial.c, /, main/features.c, include/asterisk/features.h: + Merged revisions 183126 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) + | 17 lines Allow disconnect feature before a call is bridged + feature.conf has a disconnect option. By default this option is + set to '*', but it could be anything. If a user wishes to + disconnect a call before the other side answers, only '*' will + work, regardless if the disconnect option is set to something + else. This is because features are unavailable until bridging + takes place. The default disconnect option, '*', was hardcoded in + app_dial, which doesn't make any sense from a user perspective + since they may expect it to be something different. This patch + allows features to be detected from outside of the bridge, but + not operated on. In this case, the disconnect feature can be + detected before briding and handled outside of features.c. + (closes issue #11583) Reported by: sobomax Patches: + patch-apps__app_dial.c uploaded by sobomax (license 359) + 11583.latest-patch uploaded by murf (license 17) + detect_disconnect.diff uploaded by dvossel (license 671) Tested + by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ + ........ + +2009-03-19 16:22 +0000 [r183124-183148] Russell Bryant + + * /, main/asterisk.exports: Merged revisions 183145 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r183145 | russell | 2009-03-19 11:21:56 -0500 (Thu, 19 + Mar 2009) | 1 line Add missing semicolon in exports script. + ........ + + * /, main/asterisk.exports: Merged revisions 183123 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r183123 | russell | 2009-03-19 11:13:18 -0500 (Thu, 19 + Mar 2009) | 2 lines Allow the CallerID API to work again. + ........ + +2009-03-19 16:07 +0000 [r183117] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 183115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar + 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls + would erroneously report the device as "in use." A user was + having an issue where if an outgoing SIP call was canceled, the + SIP device would remain in use if we had not received any + response to the initial INVITE we sent out. The SIP device would + remain in use until the autocongestion timer was exhausted. I + tracked down the cause of this to be the section of code I am + removing here. I asked several people what the purpose of this + code was meant to be, but no one could give me any sort of answer + as to why this was here. The person who was having this issue has + been using this patch for several months and it has stopped the + problems they have had. AST-196 ........ + +2009-03-19 15:37 +0000 [r183057-183108] Joshua Colp + + * channels/chan_sip.c: Improve our triggering of a T38 switchover + internally when triggered by a received reinvite. Previously we + reached across the channel bridge to get the other party's SIP + dialog structure in order to trigger an outgoing reinvite. This + is extremely dangerous to do and only works if bridged to another + SIP channel. This patch changes this to use the T38 control frame + method of requesting a switchover. This change also causes the + SIP channel driver to propogate back whether the switchover + worked or not instead of blindly accepting the incoming T38 + reinvite. Review: http://reviewboard.digium.com/r/200/ + + * main/channel.c: Fix an issue where a T38 control frame would get + dropped. If two channels were bridged together using a generic + bridge the T38 control frame would get passed up instead of being + indicated on the other channel. + +2009-03-18 21:28 +0000 [r183032] Kevin P. Fleming + + * res/res_ael_share.exports (added): allow this module to export + everything for now + +2009-03-18 21:18 +0000 [r183028] Jeff Peeler + + * channels/h323/ast_h323.cxx: Add some code removed by mistake from + commit 182722 that works around a file descriptor leak in + versions of PWLib prior to 1.12.0. + +2009-03-18 19:41 +0000 [r182960] Tilghman Lesher + + * main/asterisk.exports: Fixing a lost symbol in manager.c + +2009-03-18 11:40 +0000 [r182848-182883] Kevin P. Fleming + + * include/asterisk/callerid.h, channels/chan_dahdi.c, /, + main/callerid.c: Merged revisions 182882 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r182882 | kpfleming | 2009-03-18 06:31:41 -0500 (Wed, 18 Mar + 2009) | 3 lines fix another symbol namespace issue (reported by + Andrew on asterisk-dev) ........ + + * res/res_phoneprov.c, res/res_config_ldap.c, res/res_curl.c, + res/res_config_sqlite.c, res/res_jabber.exports, res/res_odbc.c, + res/res_odbc.exports: a few more namespace updates... + res_ael_share still needs some work before this can be merged to + other release branches + +2009-03-18 02:28 +0000 [r182847] Russell Bryant + + * apps/app_nbscat.c, /, main/Makefile, + include/asterisk/autoconfig.h.in, configure.ac, main/utils.c, + include/asterisk/io.h, include/asterisk/channel.h, main/poll.c, + main/io.c, main/channel.c, channels/chan_skinny.c, configure, + apps/app_mp3.c, res/res_agi.c, channels/chan_alsa.c, + include/asterisk/poll-compat.h, main/asterisk.c: Merged revisions + 182810 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) + | 44 lines Fix cases where the internal poll() was not being used + when it needed to be. We have seen a number of problems caused by + poll() not working properly on Mac OSX. If you search around, + you'll find a number of references to using select() instead of + poll() to work around these issues. In Asterisk, we've had poll.c + which implements poll() using select() internally. However, we + were still getting reports of problems. vadim investigated a bit + and realized that at least on his system, even though we were + compiling in poll.o, the system poll() was still being used. So, + the primary purpose of this patch is to ensure that we're using + the internal poll() when we want it to be used. The changes are: + 1) Remove logic for when internal poll should be used from the + Makefile. Instead, put it in the configure script. The logic in + the configure script is the same as it was in the Makefile. + Ideally, we would have a functionality test for the problem, but + that's not actually possible, since we would have to be able to + run an application on the _target_ system to test poll() + behavior. 2) Always include poll.o in the build, but it will be + empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() + throughout the source tree to ast_poll(). I feel that it is good + practice to give the API call a new name when we are changing its + behavior and not using the system version directly in all cases. + So, normally, ast_poll() is just redefined to poll(). On systems + where AST_POLL_COMPAT is defined, ast_poll() is redefined to + ast_internal_poll(). 4) Change poll() in main/poll.c to be + ast_internal_poll(). It's worth noting that any code that still + uses poll() directly will work fine (if they worked fine before). + So, for example, out of tree modules that are using poll() will + not stop working or anything. However, for modules to work + properly on Mac OSX, ast_poll() needs to be used. (closes issue + #13404) Reported by: agalbraith Tested by: russell, vadim + http://reviewboard.digium.com/r/198/ ........ + +2009-03-18 02:21 +0000 [r182826] Kevin P. Fleming + + * res/res_config_pgsql.c, /, res/res_snmp.c, res/res_smdi.exports + (added), main/Makefile, include/asterisk/astobj2.h, + res/res_agi.exports (added), Makefile.rules, main/astobj2.c, + main/asterisk.exports (added), res/res_odbc.exports (added), + res/res_speech.exports (added), res/res_config_odbc.c, + res/res_features.exports (added), build_tools/strip_nonapi + (removed), res/res_adsi.exports (added), default.exports (added), + makeopts.in, res/res_jabber.exports (added), + res/res_monitor.exports (added): Merged revisions 182808 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r182808 | kpfleming | 2009-03-17 20:55:22 -0500 (Tue, 17 Mar + 2009) | 5 lines Improve the build system to *properly* remove + unnecessary symbols from the runtime global namespace. Along the + way, change the prefixes on some internal-only API calls to use a + common prefix. With these changes, for a module to export symbols + into the global namespace, it must have *both* the + AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows + the linker to leave the symbols exposed in the module's .so file + (see res_odbc.exports for an example). ........ + +2009-03-17 21:28 +0000 [r182762] Russell Bryant + + * funcs/func_channel.c, CHANGES: Add support for the "name" option + in the CHANNEL() function. Review: + http://reviewboard.digium.com/r/199/ + +2009-03-17 20:47 +0000 [r182722] Jeff Peeler + + * channels/h323/compat_h323.cxx, channels/h323/ast_h323.cxx, + configure, autoconf/ast_check_openh323.m4, + channels/h323/compat_h323.h, channels/chan_h323.c, + channels/h323/ast_h323.h, channels/h323/chan_h323.h: Allow H.323 + Plus library to be used in addition to the OpenH323 library + Chan_h323 can now be compiled against both the previously + supported versions of OpenH323 as well as the current H.323 Plus + (version 1.20.2). The configure script has been modified to look + in the default install location of h323 to hopefully help avoid + using the environment variables OPENH323DIR and PWLIBDIR. Also, + the CLI command "h323 show version" has been added which + indicates which version of h323 is in use. (closes issue #11261) + Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch + uploaded by jthurman (license 614) + +2009-03-17 18:06 +0000 [r182596-182607] David Vossel + + * CHANGES: Fixing CHANGES in rev 182596. Progress DTMF was added + into app_dial's D() option. In CHANGES it should have been + updated under 1.6.3 rather than 1.6.2. + + * apps/app_dial.c, CHANGES: Option to send DTMF when receiving + PROGRESS status The D() option in app_dial is only able to send + DTMF after the call has been answered. A progress option has been + added to D() to allow DTMF to be sent upon receiving PROGRESS. + This allows DTMF to be sent before the call is answered. (closes + issue #12123) Reported by: VoipForces Patches: + app_dial.c_patch_trunk_valid uploaded by VoipForces (license 419) + dtmf_progress.patch uploaded by dvossel (license 671) Tested by: + VoipForces, dvossel + +2009-03-17 15:22 +0000 [r182553] Russell Bryant + + * main/channel.c: Tweak the handling of the frame list inside of + ast_answer(). This does not change any behavior, but moves the + frames from the local frame list back to the channel read queue + using an O(n) algorithm instead of O(n^2). + +2009-03-17 14:59 +0000 [r182525-182530] Kevin P. Fleming + + * main/channel.c: correct logic flaw in ast_answer() changes in + r182525 + + * main/channel.c, main/features.c, include/asterisk/channel.h: + Improve behavior of ast_answer() to not lose incoming frames + ast_answer(), when supplied a delay before returning to the + caller, use ast_safe_sleep() to implement the delay. + Unfortunately during this time any incoming frames are discarded, + which is problematic for T.38 re-INVITES and other sorts of + channel operations. When a delay is not passed to ast_answer(), + it still delays for up to 500 milliseconds, waiting for media to + arrive. Again, though, it discards any control frames, or + non-voice media frames. This patch rectifies this situation, by + storing all incoming frames during the delay period on a list, + and then requeuing them onto the channel before returning to the + caller. http://reviewboard.digium.com/r/196/ + +2009-03-17 14:24 +0000 [r182521] Sean Bright + + * autoconf/ast_ext_lib.m4: Don't include a space before the + optional extra text that may follow a help string. + +2009-03-17 05:51 +0000 [r182450] Tilghman Lesher + + * /, main/db.c: Merged revisions 182449 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) + | 7 lines Fix race in astdb The underlying db1 implementation + does not fully isolate the pages retrieved from astdb, so the + lock protecting accesses needs to be extended until the copy from + the shared memory structure is done. (closes issue #14682) + Reported by: makoto ........ + +2009-03-17 01:54 +0000 [r182408] Richard Mudgett + + * channels/chan_dahdi.c: OPENR2 uses an incorrect string value if + the extension delimiter is not present. * Fixed OPENR2 using an + incorrect string value if the extension delimiter is not present + in the Dial() function. This was fixed for SS7 and PRI in trunk + -r172400. * Made OPENR2 stripmsd behavior the same as the SS7, + PRI, and others. * Removed trailing whitespace that appeared with + OPENR2. + +2009-03-16 20:53 +0000 [r182362] Russell Bryant + + * UPGRADE.txt, CHANGES: Update UPGRADE.txt and CHANGES for 1.6.3 + -- cgit v1.2.3