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author | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-07-26 23:38:18 +0000 |
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committer | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-07-26 23:38:18 +0000 |
commit | d3b1752408cd650de50b7afb74454f28fe138fdb (patch) | |
tree | b48e55d5c20c84874466122ba6489108e30247d9 | |
parent | e20bd01ae6b43e521af7d20de64554536c60b4b0 (diff) |
Importing files for 1.8.0-beta2 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.0-beta2@279694 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | .lastclean | 3 | ||||
-rw-r--r-- | .version | 1 | ||||
-rw-r--r-- | ChangeLog | 22146 |
3 files changed, 22150 insertions, 0 deletions
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..495ea0f0c --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.8.0-beta2 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..342b3479a --- /dev/null +++ b/ChangeLog @@ -0,0 +1,22146 @@ +2010-07-26 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.8.0-beta2 Released. + +2010-07-26 23:29 +0000 [r279689] Paul Belanger <paul.belanger@polybeacon.com> + + * UPGRADE.txt, CHANGES: Updated documentation for FAX logger level. + +2010-07-26 23:03 +0000 [r279658] Jason Parker <jparker@digium.com> + + * 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 <russell@digium.com> + + * main/channel.c: Ignore a control subclass of -1 in + ast_waitfordigit_full(). + +2010-07-26 21:20 +0000 [r279599-279619] Tilghman Lesher <tlesher@digium.com> + + * /, 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 <ghenry@suretecsystems.com> + + * /: 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 <dvossel@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <sean@malleable.com> + + * 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 <tlesher@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * tests/test_func_file.c: Add trailing backslash to silence warning + message. + +2010-07-25 18:21 +0000 [r279390-279410] Tilghman Lesher <tlesher@digium.com> + + * 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 <brad.latus@gmail.com> + + * doc/asterisk.8: Minor update to man page + +2010-07-24 20:47 +0000 [r279273-279314] Paul Belanger <paul.belanger@polybeacon.com> + + * 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 <rmudgett@digium.com> + + * 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 <lmadsen@digium.com> + + * Asterisk 1.8.0-beta1 Released. + +2010-07-23 18:56 +0000 [r279113] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c: Silly 64-bit compilers (who uses 64-bit anyway?) + +2010-07-23 18:23 +0000 [r279056-279094] Russell Bryant <russell@digium.com> + + * /: 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * doc/asterisk.8: Some left-over hyphen-minus fixes in the man page + +2010-07-23 15:57 +0000 [r278944-278945] Russell Bryant <russell@digium.com> + + * 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 <mmichelson@digium.com> + + * addons/chan_ooh323.c: Well, who knew chan_ooh323 used udptl? I + sure didn't! + +2010-07-23 15:41 +0000 [r278942] Richard Mudgett <rmudgett@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <oej@edvina.net> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <rmudgett@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * include/asterisk/doxyref.h: Add the full current set of CDR + drivers + +2010-07-21 19:16 +0000 [r278539] David Vossel <dvossel@digium.com> + + * 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 <twilson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * channels/chan_dahdi.c: include stat.h for everybody, needed for + device2chan + +2010-07-20 23:23 +0000 [r278275-278307] Tilghman Lesher <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * channels/sig_pri.c: Reference correct struct member for unlikely + event PRI_EVENT_CONFIG_ERR. + +2010-07-20 22:26 +0000 [r278272] Tilghman Lesher <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jgalarneau@digium.com> + + * /, 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 <mmichelson@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <russell@digium.com> + + * 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 <brad.latus@gmail.com> + + * 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 <tim.ringenbach@gmail.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * tests/test_devicestate.c: updated devicestate test for device + state changes + +2010-07-16 19:22 +0000 [r277366] Jeff Peeler <jpeeler@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * /, 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 <mmichelson@digium.com> + + * channels/sip/reqresp_parser.c: Fix up some weird indentation + problems in reqresp_parser.c + +2010-07-16 15:20 +0000 [r277143] Sean Bright <sean@malleable.com> + + * 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 <eliels@gmail.com> + + * CREDITS: Add Despegar.com (my main sponsor) to the CREDITS file. + +2010-07-16 13:32 +0000 [r276950-277102] Olle Johansson <oej@edvina.net> + + * 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <russell@digium.com> + + * contrib/scripts/install_prereq: Add lua5.1 to the handy dandy + list of packages. + +2010-07-14 22:58 +0000 [r276571] Jeff Peeler <jpeeler@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <rmudgett@digium.com> + + * channels/chan_h323.c: Make compile again. + +2010-07-14 21:11 +0000 [r276490-276493] Tilghman Lesher <tlesher@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <rmudgett@digium.com> + + * res/snmp/agent.c: Make compile again. + +2010-07-14 16:36 +0000 [r276389] Jeff Peeler <jpeeler@digium.com> + + * 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 <tim.ringenbach@gmail.com> + + * 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 <rmudgett@digium.com> + + * 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 <lmadsen@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <twilson@digium.com> + + * 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 <russell@digium.com> + + * /: 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <twilson@digium.com> + + * 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 <russell@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <twilson@digium.com> + + * channels/chan_sip.c: Don't try to ref authpeer when it isn't set + +2010-07-12 17:54 +0000 [r275725] Richard Mudgett <rmudgett@digium.com> + + * main/channel.c: Add which ITU spec specifies the numbering plan. + +2010-07-12 17:21 +0000 [r275682] Jeff Peeler <jpeeler@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <support@transnexus.com> + + * configs/osp.conf.sample, apps/app_osplookup.c: Added support for + indirect work mode. + +2010-07-10 20:49 +0000 [r275509] Eliel C. Sardanons <eliels@gmail.com> + + * 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 <russell@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * main/features.c: Add missing ao2_iterator_destroy(). + + * apps/app_voicemail.c: Fix compile error. + +2010-07-09 19:46 +0000 [r275308] Mark Michelson <mmichelson@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <mmichelson@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * /, 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 <tlesher@digium.com> + + * apps/app_meetme.c, channels/chan_iax2.c: Weird, no output and + Bamboo still fails... + +2010-07-09 18:24 +0000 [r275186] Matthew Nicholson <mnicholson@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * channels/chan_sip.c: Return logic of sip_debug_test_addr() to its + original functionality. + +2010-07-09 16:05 +0000 [r275028] Matthew Nicholson <mnicholson@digium.com> + + * 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 <russell@digium.com> + + * 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 <mmichelson@digium.com> + + * 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] <simon.perreault@viagenie.ca> + + * 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 <russell@digium.com> + + * include/asterisk/indications.h: Extend length limit on country + name in indications.conf. + +2010-07-09 11:06 +0000 [r274866] Olle Johansson <oej@edvina.net> + + * 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 <rmudgett@digium.com> + + * 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 <mmichelson@digium.com> + + * /: 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 <rmudgett@digium.com> + + * 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 <eliels@gmail.com> + + * 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 <dvossel@digium.com> + + * channels/chan_sip.c: Fixes some ref count issues introduced by + r274539 + +2010-07-07 18:32 +0000 [r274595-274639] Richard Mudgett <rmudgett@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <support@transnexus.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <twilson@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * main/pbx.c: Uh, yeah. + +2010-07-05 13:53 +0000 [r273886] Paul Belanger <paul.belanger@polybeacon.com> + + * /, 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 <tlesher@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <russell@digium.com> + + * /, 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 <jpeeler@digium.com> + + * /, 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 <mnicholson@digium.com> + + * res/res_fax.c: Properly handle failures of fax->start_session() + FAX-177 + +2010-07-01 16:40 +0000 [r273427] David Vossel <dvossel@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <mnicholson@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <mmichelson@digium.com> + + * /, 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 <russell@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <mnicholson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * /, 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 <russell@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <twilson@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <tim.ringenbach@gmail.com> + + * 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 <tlesher@digium.com> + + * 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 <twilson@digium.com> + + * apps/app_meetme.c: Don't start the sla thread unless we realy + need it + +2010-06-23 18:25 +0000 [r272145] Tilghman Lesher <tlesher@digium.com> + + * 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 <twilson@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <russell@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <russell@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * main/file.c: file.c was truncating audio file formats to the + lower 32bits. + +2010-06-18 18:36 +0000 [r271336] Jeff Peeler <jpeeler@digium.com> + + * /: 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 <dvossel@digium.com> + + * 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 <jpeeler@digium.com> + + * channels/sig_analog.c: Change expected operation from error to + debug message + +2010-06-17 00:30 +0000 [r271089] Paul Belanger <paul.belanger@polybeacon.com> + + * 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 <dvossel@digium.com> + + * channels/sip/reqresp_parser.c: addition of more parse_uri test + cases + +2010-06-16 21:17 +0000 [r270987] Paul Belanger <paul.belanger@polybeacon.com> + + * /, 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 <jparker@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <dvossel@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <russell@digium.com> + + * 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 <twilson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <lmadsen@digium.com> + + * /, 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 <paul.belanger@polybeacon.com> + + * /, 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 <rmudgett@digium.com> + + * 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 <jparker@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <tlesher@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <mmichelson@digium.com> + + * 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 <russell@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <beep> 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 <russell@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <rmudgett@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <russell@digium.com> + + * 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 <brad.latus@gmail.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <russell@digium.com> + + * 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 <sean@malleable.com> + + * /, 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 <lmadsen@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <twilson@digium.com> + + * 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 <rmudgett@digium.com> + + * channels/sip/dialplan_functions.c: Make SIP tests compile again. + +2010-06-07 22:56 +0000 [r268817-268818] Tilghman Lesher <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * main/manager.c: Event well was going dry. (issue #17234) + +2010-06-07 17:34 +0000 [r268690] Paul Belanger <paul.belanger@polybeacon.com> + + * 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 <tlesher@digium.com> + + * res/res_smdi.c: Avoid unloading res_smdi twice. (closes issue + #17237) Reported by: pabelanger + +2010-06-07 15:51 +0000 [r268578] Richard Mudgett <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <kpfleming@digium.com> + + * CHANGES: Typo fix. + + * CHANGES: Grammatical error fix. + +2010-06-05 02:51 +0000 [r268321] Tilghman Lesher <tlesher@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <twilson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <rmudgett@digium.com> + + * codecs/codec_dahdi.c: Make compile again. + +2010-06-03 17:31 +0000 [r267537] Russell Bryant <russell@digium.com> + + * channels/chan_usbradio.c: Don't stop Asterisk if chan_usbradio + isn't configured. + +2010-06-03 17:09 +0000 [r267492] Mark Michelson <mmichelson@digium.com> + + * 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 <russell@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <russell@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <russell@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <russell@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <russell@digium.com> + + * apps/app_voicemail.c: Silence a compiler warning. + +2010-06-02 17:29 +0000 [r267065] Jeff Peeler <jpeeler@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * /, 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 <rmudgett@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <twilson@digium.com> + + * res/res_calendar_exchange.c: Use the correct ical.h file + +2010-06-01 21:28 +0000 [r266828] Tilghman Lesher <tlesher@digium.com> + + * 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 <twilson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <tlesher@digium.com> + + * funcs/func_env.c: Needs to be wrapped in <para> + + * 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 <twilson@digium.com> + + * 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 <libical/ical.h>. + Since we now test for libical/ical.h, we can change the #includes + back to <libical/ical.h> and remove the test which specifically + adds /usr/include/libical as an include directory. + +2010-05-28 22:50 +0000 [r266337-266385] Tilghman Lesher <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <twilson@digium.com> + + * configure, configure.ac, makeopts.in: More build fixes for + ical/neon and res_calendar_ews + +2010-05-27 20:08 +0000 [r266240] Jeff Peeler <jpeeler@digium.com> + + * pbx/pbx_realtime.c: fix compile error + +2010-05-27 19:25 +0000 [r266146-266238] Tilghman Lesher <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <twilson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <rmudgett@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * addons/ooh323c/src/oochannels.c: Typos: 'succesful' (lintian) + +2010-05-24 22:21 +0000 [r265467] Terry Wilson <twilson@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <rmudgett@digium.com> + + * apps/app_rpt.c: Make app_rpt.c able to compile again. + +2010-05-24 19:42 +0000 [r265366] David Vossel <dvossel@digium.com> + + * 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 <twilson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * main/channel.c: fixes segfault when using generic plc + +2010-05-23 18:23 +0000 [r265227] Alexandr Anikin <may@telecom-service.ru> + + * addons/chan_ooh323.c: small changes to avoiding 'freeing unused + memory...' + +2010-05-21 22:46 +0000 [r265174] Richard Mudgett <rmudgett@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <twilson@digium.com> + + * 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 <rmudgett@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <twilson@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <lmadsen@digium.com> + + * main/cli.c: Fix compilation problem with previous commit. (issue + #16009) + +2010-05-19 15:29 +0000 [r264160] Kevin P. Fleming <kpfleming@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <tlesher@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * res/res_timing_kqueue.c: Make happy green color come back + +2010-05-18 20:09 +0000 [r263810] Mark Michelson <mmichelson@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <dvossel@digium.com> + + * CHANGES: Update CHANGES to reflect DAHDI buffer dialstring option + backport to 1.6.2 + +2010-05-16 16:31 +0000 [r263250] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * 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 <directory> 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 <kpfleming@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <rmudgett@digium.com> + + * channels/chan_dahdi.c: Fix inverted logic in cli command: ss7 set + debug on/off + +2010-05-13 20:25 +0000 [r263028] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <lmadsen@digium.com> + + * 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 <dvossel@digium.com> + + * /, 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jparker@digium.com> + + * /, 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 <paul.belanger@polybeacon.com> + + * 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 <tlesher@digium.com> + + * /, 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 <jparker@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <tlesher@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <jparker@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <rmudgett@digium.com> + + * channels/chan_dahdi.c: Avoid a crash on SS7 channels. + +2010-05-05 20:48 +0000 [r261364] Russell Bryant <russell@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * /, 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 <mmichelson@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <jparker@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <paul.belanger@polybeacon.com> + + * 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 <lmadsen@digium.com> + + * 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 <eliels@gmail.com> + + * 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 <jpeeler@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <mmichelson@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * configs/extensions.conf.sample: Pattern match fail. + +2010-04-29 15:33 +0000 [r260050] David Vossel <dvossel@digium.com> + + * /, 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 <rmudgett@digium.com> + + * include/asterisk/extconf.h: Fix comment. + +2010-04-28 22:34 +0000 [r259957] Mark Michelson <mmichelson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <lmadsen@digium.com> + + * /, 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 <jparker@digium.com> + + * /, 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 <jpeeler@digium.com> + + * /, 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 <jparker@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <mmichelson@digium.com> + + * main/ccss.c: Shuffle some casts to make builds on bamboo happier. + +2010-04-27 21:49 +0000 [r259527] Leif Madsen <lmadsen@digium.com> + + * /, 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 <jparker@digium.com> + + * /: 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 <lmadsen@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <jparker@digium.com> + + * /, 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <thedavidfactor@gmail.com> + + * 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 <lmadsen@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <may@telecom-service.ru> + + * 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 <tlesher@digium.com> + + * /, 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 <jparker@digium.com> + + * 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 <mnicholson@digium.com> + + * main/channel.c: Fix previous commit. + +2010-04-22 22:10 +0000 [r258673-258674] Jason Parker <jparker@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <russell@digium.com> + + * 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 <eliels@gmail.com> + + * apps/app_voicemail.c: Pass interactive = 0 and fix a compile + error. + +2010-04-22 19:08 +0000 [r258557] Jason Parker <jparker@digium.com> + + * 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 <eliels@gmail.com> + + * 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 + <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY) + <eliels@gmail.com> 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 <russell@digium.com> + + * doc/tex/channelvariables.tex: Add MEETMEBOOKID from r256019. + +2010-04-21 21:56 +0000 [r258433] Jeff Peeler <jpeeler@digium.com> + + * /, 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 <lmadsen@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <julian@dotr.com> + + * 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 <lmadsen@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <lmadsen@digium.com> + + * 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 <jparker@digium.com> + + * 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 <tlesher@digium.com> + + * apps/app_voicemail.c: Bad merge fix + +2010-04-19 18:42 +0000 [r257851] Mark Michelson <mmichelson@digium.com> + + * 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 <twilson@digium.com> + + * 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 <tlesher@digium.com> + + * configs/cdr_odbc.conf.sample: Removing unused configuration + parameters + +2010-04-16 21:22 +0000 [r257713] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * /, 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 <mmichelson@digium.com> + + * 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 <dwayne.hubbard@gmail.com> + + * 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 <tlesher@digium.com> + + * 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 <lmadsen@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <lmadsen@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <russell@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <mmichelson@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <kpfleming@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <michiel@vanbaak.info> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <lmadsen@digium.com> + + * 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 <russell@digium.com> + + * /, 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 <jaredsmith@jaredsmith.net> + + * 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 <russell@digium.com> + + * main/event.c: Remove a debugging log entry. + +2010-03-27 23:51 +0000 [r255199] Alexandr Anikin <may@telecom-service.ru> + + * 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 <sean@malleable.com> + + * 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 <tlesher@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <sean@malleable.com> + + * 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 <kpfleming@digium.com> + + * 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 <russell@digium.com> + + * 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 <jparker@digium.com> + + * 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 <russell@digium.com> + + * 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 <kpfleming@digium.com> + + * .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 <mmichelson@digium.com> + + * 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 <twilson@digium.com> + + * /, 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 <kpfleming@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <mmichelson@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <russell@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <sean@malleable.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <twilson@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <russell@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <philippe.sultan@gmail.com> + + * 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 <lmadsen@digium.com> + + * /, 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 <tlesher@digium.com> + + * main/stdtime/localtime.c: Just in case of a race, send the signal + on interrupt. + +2010-03-17 19:06 +0000 [r253205] Leif Madsen <lmadsen@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * tests/test_time.c: Re-enable test_time on non-Linux. + +2010-03-16 19:36 +0000 [r252848] Sean Bright <sean@malleable.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <tlesher@digium.com> + + * codecs/gsm/Makefile: OSARCH is not inherited to this directory + +2010-03-16 18:36 +0000 [r252759] Russell Bryant <russell@digium.com> + + * tests/test_time.c: Disable this test on non-Linux for now. + +2010-03-15 22:48 +0000 [r252709] Kevin P. Fleming <kpfleming@digium.com> + + * 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 <russell@digium.com> + + * 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 <sean@malleable.com> + + * 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 <tlesher@digium.com> + + * 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 <lmadsen@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <sean@malleable.com> + + * 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 <may@telecom-service.ru> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <twilson@digium.com> + + * 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 <moises.silva@gmail.com> + + * channels/chan_dahdi.c: add missing mfcr2_skip_category setting + +2010-03-12 19:43 +0000 [r251989] Tilghman Lesher <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * /: 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 <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <michiel@vanbaak.info> + + * 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 <may@telecom-service.ru> + + * addons/ooh323c/src/ooq931.c: small log issue from bug 0016664 + +2010-03-06 14:16 +0000 [r251137] Russell Bryant <russell@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <russell@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <mmichelson@digium.com> + + * 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 <lmadsen@digium.com> + + * /: 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 <rmudgett@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <lmadsen@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <sean@malleable.com> + + * /, 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 <michiel@vanbaak.info> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <tlesher@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <kpfleming@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * main/test.c: adds Time field to "test show results" cli command + +2010-02-26 17:13 +0000 [r249101-249105] Mark Michelson <mmichelson@digium.com> + + * 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 <russell@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <jpeeler@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <dvossel@digium.com> + + * /, 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 <mmichelson@digium.com> + + * 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 <russell@digium.com> + + * 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 <michiel@vanbaak.info> + + * 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 <oej@edvina.net> + + * res/res_rtp_asterisk.c: Improve support for RTCP reports without + report blocks + +2010-02-19 18:38 +0000 [r248003] Moises Silva <moises.silva@gmail.com> + + * 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 <dvossel@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <rmudgett@digium.com> + + * 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 <lmadsen@digium.com> + + * /, 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 <philippe.sultan@gmail.com> + + * 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 <russell@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mmichelson@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <may@telecom-service.ru> + + * 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 <mmichelson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <support@transnexus.com> + + * apps/app_osplookup.c, UPGRADE.txt, CHANGES: Updated doc for OSP + lookup application. + +2010-02-11 21:57 +0000 [r246299-246338] David Vossel <dvossel@digium.com> + + * 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 <russell@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jparker@digium.com> + + * /, 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 <russell@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <twilson@digium.com> + + * 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 <sean@malleable.com> + + * /, 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <rmudgett@digium.com> + + * channels/sip: Add ignore *.i files property to the new + channels/sip directory. + +2010-02-03 20:48 +0000 [r244598] Jeff Peeler <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * channels/iax2-parser.c: Informational message, not an error. + +2010-01-28 18:35 +0000 [r243780-243860] Russell Bryant <russell@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <mmichelson@digium.com> + + * 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 <russell@digium.com> + + * 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 <dvossel@digium.com> + + * /, 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 <thedavidfactor@gmail.com> + + * 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 <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <may@telecom-service.ru> + + * 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 <oej@edvina.net> + + * 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 <tlesher@digium.com> + + * /, 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 <twilson@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <may@telecom-service.ru> + + * 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 <sean@malleable.com> + + * 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 <tlesher@digium.com> + + * /, 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 <tlesher@digium.com> + + * /, 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 <thedavidfactor@gmail.com> + + * 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 <tlesher@digium.com> + + * 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 <oej@edvina.net> + + * /, 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 <russell@digium.com> + + * 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 <mmichelson@digium.com> + + * main/acl.c: Add missing argument to ast_calloc calls. + +2010-01-21 21:05 +0000 [r242043] Olle Johansson <oej@edvina.net> + + * main/acl.c: Make sure we initialize the ast_ha structure with + ast_calloc + +2010-01-21 15:27 +0000 [r241938] Sean Bright <sean@malleable.com> + + * /, 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <kpfleming@digium.com> + + * channels/chan_vpb.cc: Fix up compile breakage from + ast_tvdiff_ms() API change. + +2010-01-20 08:18 +0000 [r241416] Alec L Davis <sivad.a@paradise.net.nz> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <jparker@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <jpeeler@digium.com> + + * 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 <jparker@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <sean@malleable.com> + + * /, 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 <tlesher@digium.com> + + * funcs/func_strings.c, CHANGES: Make HASHes inheritable across + channel creation. + +2010-01-18 18:00 +0000 [r240973-240974] David Ruggles <thedavidfactor@gmail.com> + + * 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 <dvossel@digium.com> + + * 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 <thedavidfactor@gmail.com> + + * 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 <dvossel@digium.com> + + * 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 <sean@malleable.com> + + * 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 <sean@malleable.com> + + * res/res_musiconhold.c: Get MoH building on OpenSolaris. + +2010-01-15 23:50 +0000 [r240629] Tilghman Lesher <tlesher@digium.com> + + * Makefile, main/asterisk.c: Err, oops, it was already the way I + intended. + +2010-01-15 23:09 +0000 [r240548-240552] Russell Bryant <russell@digium.com> + + * 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 <sean@malleable.com> + + * res/res_timing_timerfd.c: Clarify error message in + res_timing_timerfd. + +2010-01-15 21:42 +0000 [r240421-240500] Tilghman Lesher <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <sean@malleable.com> + + * 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 <russell@digium.com> + + * configure: Update configure script for an OSP toolkit related + change. + +2010-01-15 16:28 +0000 [r240328] Kevin P. Fleming <kpfleming@digium.com> + + * configs/sip.conf.sample: Clarify RTP NAT handling a bit. + +2010-01-14 23:13 +0000 [r240226-240271] Sean Bright <sean@malleable.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <lmadsen@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <oej@edvina.net> + + * 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 <support@transnexus.com> + + * 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 <tlesher@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <lmadsen@digium.com> + + * 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 <sean@malleable.com> + + * 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 <dvossel@digium.com> + + * 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 <oej@edvina.net> + + * 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 <tlesher@digium.com> + + * /, 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 <sean@malleable.com> + + * 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 <tlesher@digium.com> + + * 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 <may@telecom-service.ru> + + * 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 <tlesher@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <rmudgett@digium.com> + + * channels/sig_pri.c: Fix using the wrong pointer type in + do_idle_thread(). + +2010-01-07 20:42 +0000 [r238361-238492] David Vossel <dvossel@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * configs/sip.conf.sample: Document the usefulness of explicit + udp:// in the register string + +2010-01-06 21:45 +0000 [r238231] Tilghman Lesher <tlesher@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <sean@malleable.com> + + * 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 <russell@digium.com> + + * /, 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 <tlesher@digium.com> + + * channels/chan_sip.c: Whoa, duplicate setting (dead code). + +2010-01-05 23:08 +0000 [r237920] David Vossel <dvossel@digium.com> + + * 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 <mmichelson@digium.com> + + * apps/app_dial.c: Mismerged a bit. + +2010-01-05 19:29 +0000 [r237839] David Vossel <dvossel@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <russell@digium.com> + + * 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 <michiel@vanbaak.info> + + * apps/app_mixmonitor.c: Make CLI command 'mixmonitor start|stop + <channel> 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 <tlesher@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <oej@edvina.net> + + * 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 <support@transnexus.com> + + * 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 <tlesher@digium.com> + + * channels/chan_sip.c: global_contact_ha was renamed in trunk + +2010-01-02 09:54 +0000 [r237136] Olle Johansson <oej@edvina.net> + + * /, 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 <may@telecom-service.ru> + + * 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 <jparker@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <support@transnexus.com> + + * 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 <jparker@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <sean@malleable.com> + + * /, 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 <tlesher@digium.com> + + * 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 <kpfleming@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <kpfleming@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <sivad.a@paradise.net.nz> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <jcolp@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <jaredsmith@jaredsmith.net> + + * /, 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * include/asterisk/strings.h: Is it Friday yet? + +2009-12-15 23:41 +0000 [r235226] Jeff Peeler <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <kpfleming@digium.com> + + * 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 <sivad.a@paradise.net.nz> + + * 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 <tlesher@digium.com> + + * 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 <jparker@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <lmadsen@digium.com> + + * 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 <sean@malleable.com> + + * 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 <oej@edvina.net> + + * /, 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <jparker@digium.com> + + * 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 <tlesher@digium.com> + + * res/res_musiconhold.c: Missed a case that emits a WARNING where + none is warranted. + +2009-12-10 17:31 +0000 [r234173] Jeff Peeler <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <dvossel@digium.com> + + * main/utils.c: fixes incorrect logic in ast_uri_encode issue + #16299 + +2009-12-07 23:10 +0000 [r233577] Atis Lezdins <atis@iq-labs.net> + + * 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 <thedavidfactor@gmail.com> + + * 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 <dvossel@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <russell@digium.com> + + * 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 <mnick@digium.com> + + * 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 <michiel@vanbaak.info> + + * 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 <tlesher@digium.com> + + * 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 <mnick@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <may@telecom-service.ru> + + * 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 <support@transnexus.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <thedavidfactor@gmail.com> + + * 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 <jpeeler@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <jcolp@digium.com> + + * /, 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 <dvossel@digium.com> + + * /, 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 <jcolp@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <russell@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <kpfleming@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <support@transnexus.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <kpfleming@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <thedavidfactor@gmail.com> + + * 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 <dvossel@digium.com> + + * 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 <thedavidfactor@gmail.com> + + * 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 <kpfleming@digium.com> + + * 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 <support@transnexus.com> + + * 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 <kpfleming@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <jcolp@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <oej@edvina.net> + + * 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 <support@transnexus.com> + + * apps/app_osplookup.c: Added full number portability parameter + support. + +2009-11-12 23:43 +0000 [r229750-229754] Jason Parker <jparker@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <oej@edvina.net> + + * 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 <thedavidfactor@gmail.com> + + * 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 <dbrooks@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <thedavidfactor@gmail.com> + + * 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 <tlesher@digium.com> + + * 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 <jcolp@digium.com> + + * /, 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 <thedavidfactor@gmail.com> + + * /, 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 <dvossel@digium.com> + + * /, 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 <mnicholson@digium.com> + + * channels/chan_sip.c: Reverted revision 201717. (closes issue + 0016175) Reported by: paul-tg + +2009-11-10 15:27 +0000 [r229093] David Vossel <dvossel@digium.com> + + * 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 <ghenry@suretecsystems.com> + + * 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 <twilson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <dvossel@digium.com> + + * 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 <rmudgett@digium.com> + + * 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(<keypad_digits>) R Reverse charging indication (Collect calls) + The earlier Dial(DAHDI/g1[/K<keypad_digits>][/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 <dbrooks@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <rmudgett@digium.com> + + * main/frame.c: Fix compiler warning gcc 4.2.4 found + +2009-11-06 19:47 +0000 [r228620] Matthew Nicholson <mnicholson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jcolp@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <jcolp@digium.com> + + * /, 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 <dvossel@digium.com> + + * /, 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 <mmichelson@digium.com> + + * funcs/func_cdr.c: Fix XML in func_cdr.c + +2009-11-05 22:12 +0000 [r228191-228196] Tilghman Lesher <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <dbrooks@digium.com> + + * 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 <jparker@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <may@telecom-service.ru> + + * 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 <mnicholson@digium.com> + + * 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 <tlesher@digium.com> + + * include/asterisk/unaligned.h: Fixes for gcc 4.4 + +2009-11-04 20:13 +0000 [r227759] Matthew Nicholson <mnicholson@digium.com> + + * 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 <jcolp@digium.com> + + * /, 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 <mmichelson@digium.com> + + * main/frame.c: Add a couple more casts so that code compiles + correctly. + +2009-11-04 16:35 +0000 [r227645] Tilghman Lesher <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * channels/chan_dahdi.c: fix trunk building + +2009-11-04 16:17 +0000 [r227579-227615] Tilghman Lesher <tlesher@digium.com> + + * 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 <oej@edvina.net> + + * 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 <tlesher@digium.com> + + * apps/app_queue.c: Don't crash when state_interface is NULL. + +2009-11-03 22:13 +0000 [r227462-227464] Russell Bryant <russell@digium.com> + + * 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 <dbrooks@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <jcolp@digium.com> + + * configs/queues.conf.sample, apps/app_queue.c: Add support for + using a hint when configuring a state interface using the format + hint:<extension>@<context>. (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 <jparker@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <rmudgett@digium.com> + + * /: 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 <tlesher@digium.com> + + * channels/chan_mgcp.c: Code guidelines fixes only + +2009-11-03 17:12 +0000 [r227238] David Vossel <dvossel@digium.com> + + * 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 <oej@edvina.net> + + * 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 <jcolp@digium.com> + + * /: 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 <lmadsen@digium.com> + + * 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 <oej@edvina.net> + + * 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 <tlesher@digium.com> + + * 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 <dbrooks@digium.com> + + * 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 <oej@edvina.net> + + * main/http.c: Adding external reference for doxygen + +2009-11-02 18:08 +0000 [r226890] Joshua Colp <jcolp@digium.com> + + * 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 <rmudgett@digium.com> + + * 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<span>/<number>[:<subaddress>]-<sequence-number> 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 <tlesher@digium.com> + + * /, 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 <mnicholson@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <russell@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <oej@edvina.net> + + * channels/chan_local.c: Doxygen documentation update + +2009-10-28 20:50 +0000 [r226453] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * 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 <lmadsen@digium.com> + + * /, 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 <tlesher@digium.com> + + * /, 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 <tzafrir.cohen@xorcom.com> + + * 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 <oej@edvina.net> + + * Makefile: Adding compile time flags for Snow Leopard, Leopard and + some other animals + +2009-10-27 20:22 +0000 [r226159] Tilghman Lesher <tlesher@digium.com> + + * 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 <twilson@digium.com> + + * res/res_http_post.c: Don't prepend the URI prefix to the post + directory + +2009-10-27 13:30 +0000 [r226060] Joshua Colp <jcolp@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * /, 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 <kpfleming@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <rmudgett@digium.com> + + * channels/chan_dahdi.c: Make conditionals create previous code + when libpri/ss7 are present. + +2009-10-26 13:29 +0000 [r225767-225836] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * 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 <kpfleming@digium.com> + + * 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 <rmudgett@digium.com> + + * 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<keypad_digits>][/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 <sean@malleable.com> + + * 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 <dvossel@digium.com> + + * channels/chan_sip.c: Fixes an iterator memory leak and + uninitialized memory + +2009-10-23 14:02 +0000 [r225582] Kevin P. Fleming <kpfleming@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <rmudgett@digium.com> + + * channels/sig_pri.c: Search for the subaddress only within the + extension section of the dial string. + Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension]) + +2009-10-22 19:55 +0000 [r225445] David Vossel <dvossel@digium.com> + + * 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 <sean@malleable.com> + + * 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 <tlesher@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <dvossel@digium.com> + + * /, 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 <kpfleming@digium.com> + + * 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 <russell@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <russell@digium.com> + + * 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 <rmudgett@digium.com> + + * channels/sig_pri.c: Make PRI_SUBCMD_xxx handling subaddress + friendly. + +2009-10-20 22:09 +0000 [r224856] Tilghman Lesher <tlesher@digium.com> + + * 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 <jcolp@digium.com> + + * /, 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 <mnicholson@digium.com> + + * CHANGES: Added information to CHANGES about the dynamic range + compression feature added to dahdi. + +2009-10-19 23:47 +0000 [r224671] Kevin P. Fleming <kpfleming@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <kpfleming@digium.com> + + * formats/format_siren14.c: Remove useless debugging message. + +2009-10-19 15:50 +0000 [r224527] Tilghman Lesher <tlesher@digium.com> + + * doc/janitor-projects.txt: Remove a completed project and add + another + +2009-10-19 14:32 +0000 [r224491] Joshua Colp <jcolp@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <tlesher@digium.com> + + * include/asterisk/app.h, main/app.c: Remove unnecessary typedef + +2009-10-17 02:01 +0000 [r224331-224335] Jeff Peeler <jpeeler@digium.com> + + * 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 <rmudgett@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <dbailey@digium.com> + + * 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 <twilson@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <twilson@digium.com> + + * res/res_calendar.c: use Calendar: instead of Calendar/ for + devstate + +2009-10-13 17:11 +0000 [r223911-223912] Richard Mudgett <rmudgett@digium.com> + + * 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 <twilson@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * configs/iax.conf.sample: Clarifies trunkmaxsize, trunkfreq, and + trunkmtu iax2 options SWP-151 + +2009-10-12 15:32 +0000 [r223652-223693] Kevin P. Fleming <kpfleming@digium.com> + + * /: 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 <mmichelson@digium.com> + + * 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 <russell@digium.com> + + * /: 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 <twilson@digium.com> + + * 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 <oej@edvina.net> + + * 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 <twilson@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <mmichelson@digium.com> + + * /: 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 <dvossel@digium.com> + + * /, 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 <mnicholson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <twilson@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <russell@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <rmudgett@digium.com> + + * /, 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 <dvossel@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <oej@edvina.net> + + * 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 <jparker@digium.com> + + * 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 <dvossel@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <kpfleming@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <dvossel@digium.com> + + * /, 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 <tlesher@digium.com> + + * /, 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * UPGRADE.txt, CHANGES: Move DAHDI/ISDN channel naming note from + CHANGES to UPGRADE.txt. + +2009-10-01 20:09 +0000 [r221705] Tilghman Lesher <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * 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-<span>-<sequence-number> + +2009-10-01 19:33 +0000 [r221697] David Vossel <dvossel@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <oej@edvina.net> + + * 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 <mnicholson@digium.com> + + * channels/chan_sip.c: Cleaned up merge from r221432 + +2009-09-30 21:15 +0000 [r221436] Matthias Nick <mnick@digium.com> + + * apps/app_queue.c: Prevents from division by zero + +2009-09-30 20:40 +0000 [r221432] Matthew Nicholson <mnicholson@digium.com> + + * /, 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 <mnick@digium.com> + + * 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 <twilson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <sean@malleable.com> + + * 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 <tlesher@digium.com> + + * 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 <mmichelson@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <mnicholson@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <rmudgett@digium.com> + + * channels/chan_dahdi.c, channels/sig_pri.c: Miscellaneous minor + changes. + +2009-09-28 19:11 +0000 [r220721] Sean Bright <sean@malleable.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <michiel@vanbaak.info> + + * 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 <tlesher@digium.com> + + * include/asterisk/aes.h: Allow AES to compile, when OpenSSL is not + present. + +2009-09-25 19:56 +0000 [r220543] Richard Mudgett <rmudgett@digium.com> + + * channels/sig_pri.c: Reduce indentation in sig_pri_available(). + +2009-09-25 14:50 +0000 [r220494-220496] Kevin P. Fleming <kpfleming@digium.com> + + * 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 <philippe.sultan@gmail.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <oej@edvina.net> + + * 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 <tlesher@digium.com> + + * 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 <sean@malleable.com> + + * 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 <mnicholson@digium.com> + + * 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 <sean@malleable.com> + + * 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 <michiel@vanbaak.info> + + * 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 <tlesher@digium.com> + + * 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 <lmadsen@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <dvossel@digium.com> + + * /, 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 <tlesher@digium.com> + + * /, 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 <russell@digium.com> + + * /, 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 <dvossel@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <mmichelson@digium.com> + + * /, 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 <dvossel@digium.com> + + * /, 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 <jcolp@digium.com> + + * channels/chan_sip.c: Ensure no spaces exist before "refresher=" + when doing the comparison. + +2009-09-17 16:25 +0000 [r219230] Sean Bright <sean@malleable.com> + + * apps/app_chanspy.c: Get this compiling under dev-mode. + +2009-09-17 15:18 +0000 [r219139] Matthew Nicholson <mnicholson@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <sean@malleable.com> + + * 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 <mmichelson@digium.com> + + * 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 <jcolp@digium.com> + + * 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 <dbrooks@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <dvossel@digium.com> + + * channels/chan_sip.c: upward bound checking for port string to int + conversion + +2009-09-15 16:15 +0000 [r218586] Matthew Nicholson <mnicholson@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <mmichelson@digium.com> + + * 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 <kpfleming@digium.com> + + * /: 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 <mmichelson@digium.com> + + * channels/chan_sip.c: Fix off-by-one error when reading SDP sent + over TCP. + +2009-09-15 10:24 +0000 [r218465] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * channels/chan_dahdi.c: Fix false error message on + DAHDI_EVENT_REMOVED (RESULT_SUCCESS == 0) + +2009-09-14 22:38 +0000 [r218430] Jeff Peeler <jpeeler@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * /, 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 <jcolp@digium.com> + + * 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 <mnicholson@digium.com> + + * /, 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 <tzafrir.cohen@xorcom.com> + + * 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 <moises.silva@gmail.com> + + * channels/chan_dahdi.c: get rid of mfcr2 monitor thread condition, + is problematic + +2009-09-12 13:08 +0000 [r218107] Michiel van Baak <michiel@vanbaak.info> + + * res/res_rtp_asterisk.c: use the actual given ip address for 'rtp + set debug ip <foo>' 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <dvossel@digium.com> + + * /, 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 <jpeeler@digium.com> + + * 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 <mnick@digium.com> + + * 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 <oej@edvina.net> + + * channels/chan_sip.c: Don't assign UINT_MAX to an INT. + +2009-09-10 18:17 +0000 [r217638] Tilghman Lesher <tlesher@digium.com> + + * 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 <oej@edvina.net> + + * 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 <rmudgett@digium.com> + + * channels/chan_dahdi.c: Fix available() for SS7, MFC/R2, and + pseudo channels. + +2009-09-09 21:48 +0000 [r217524] Moises Silva <moises.silva@gmail.com> + + * channels/chan_dahdi.c: ast_log replaced for ast_verbose in MFCR2 + event notifications + +2009-09-09 20:09 +0000 [r217482] Olle Johansson <oej@edvina.net> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <sean@malleable.com> + + * 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 <oej@edvina.net> + + * 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 <rmudgett@digium.com> + + * 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 <sean@malleable.com> + + * apps/app_meetme.c: Fix compilation of app_meetme. Reported by + ebroad in #asterisk-bugs + +2009-09-08 21:17 +0000 [r217236] Richard Mudgett <rmudgett@digium.com> + + * channels/sig_pri.c: Remove duplicate entry in the sig_pri_pri + private pointer array. + +2009-09-08 20:28 +0000 [r217199] Tilghman Lesher <tlesher@digium.com> + + * /, 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 <mmichelson@digium.com> + + * 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 <russell@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <tzafrir.cohen@xorcom.com> + + * 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 <dvossel@digium.com> + + * 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 <oej@edvina.net> + + * 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 <tlesher@digium.com> + + * 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 <oej@edvina.net> + + * 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 <dvossel@digium.com> + + * 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 <sean@malleable.com> + + * apps/app_voicemail.c: Use ast_free() instead of free(). + +2009-09-04 17:50 +0000 [r216547-216551] Tilghman Lesher <tlesher@digium.com> + + * 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 <michiel@vanbaak.info> + + * /, 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 <oej@edvina.net> + + * 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 <michiel@vanbaak.info> + + * 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 <russell@digium.com> + + * 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 <oej@edvina.net> + + * 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. </end of long commit message> + +2009-09-04 10:48 +0000 [r216264] Russell Bryant <russell@digium.com> + + * /, 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 <michiel@vanbaak.info> + + * 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 <rmudgett@digium.com> + + * 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 <dbailey@digium.com> + + * 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 <russell@digium.com> + + * /, 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 <kpfleming@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <oej@edvina.net> + + * 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 <michiel@vanbaak.info> + + * 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 <tlesher@digium.com> + + * 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 <twilson@digium.com> + + * /, 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 <rmudgett@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <michiel@vanbaak.info> + + * channels/chan_sip.c, channels/chan_skinny.c: add Parkinglot info + to sip show peer <foo> and skinny show line <foo> 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 <dbailey@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <michiel@vanbaak.info> + + * 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 <dvossel@digium.com> + + * 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 <michiel@vanbaak.info> + + * 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 <oej@edvina.net> + + * 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 <dwayne.hubbard@gmail.com> + + * /, 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <oej@edvina.net> + + * channels/chan_sip.c: Removing whitespace that causes red dots in + reviewboard + +2009-08-31 22:02 +0000 [r215069-215070] Tilghman Lesher <tlesher@digium.com> + + * 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 <oej@edvina.net> + + * 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <rmudgett@digium.com> + + * 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 <mmichelson@digium.com> + + * include/asterisk/sched.h: Fix some incorrect documentation of + sched_thread functions. + +2009-08-28 16:50 +0000 [r214360-214611] Tilghman Lesher <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <rmudgett@digium.com> + + * configs/chan_dahdi.conf.sample: Minor punctuation change. + +2009-08-26 16:53 +0000 [r214199] Tilghman Lesher <tlesher@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <moises.silva@gmail.com> + + * 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 <rmudgett@digium.com> + + * 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 <tlesher@digium.com> + + * channels/chan_sip.c: Clarifying comments in sip_register, and + removing a dead section + +2009-08-21 22:22 +0000 [r213716] David Vossel <dvossel@digium.com> + + * 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 <kpfleming@digium.com> + + * 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 <dvossel@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <jparker@digium.com> + + * /, 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 <moises.silva@gmail.com> + + * channels/chan_dahdi.c: increment the mfcr2 monitor count when + clearing the call request + +2009-08-21 03:48 +0000 [r213450] Terry Wilson <twilson@digium.com> + + * main/loader.c: Make LOAD_ORDER actually work + +2009-08-20 22:13 +0000 [r213414] Tilghman Lesher <tlesher@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <mnicholson@digium.com> + + * 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 <jpeeler@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <moises.silva@gmail.com> + + * 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 <jparker@digium.com> + + * 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 <dvossel@digium.com> + + * /, 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 <tlesher@digium.com> + + * 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 <russell@digium.com> + + * main/features.c: Don't blow up on a NULL cdr. Reported in + #asterisk-dev. + +2009-08-18 23:53 +0000 [r213007] Richard Mudgett <rmudgett@digium.com> + + * 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 <kpfleming@digium.com> + + * /: 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 <tlesher@digium.com> + + * 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 <oej@edvina.net> + + * apps/app_meetme.c: Small doxygen changes + +2009-08-18 16:38 +0000 [r212764] Sean Bright <sean@malleable.com> + + * 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 <rmudgett@digium.com> + + * /, 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 <kpfleming@digium.com> + + * 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 <tlesher@digium.com> + + * 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 <sean@malleable.com> + + * 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 <jpeeler@digium.com> + + * /, 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 <kpfleming@digium.com> + + * 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] Richa |