[svn-commits] tilghman: tag 1.6.1.5-rc1 r211724 - /tags/1.6.1.5-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Aug 11 16:32:31 CDT 2009
Author: tilghman
Date: Tue Aug 11 16:32:27 2009
New Revision: 211724
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=211724
Log:
Importing files for 1.6.1.5-rc1 release.
Added:
tags/1.6.1.5-rc1/.lastclean (with props)
tags/1.6.1.5-rc1/.version (with props)
tags/1.6.1.5-rc1/ChangeLog (with props)
Added: tags/1.6.1.5-rc1/.lastclean
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.1.5-rc1/.lastclean?view=auto&rev=211724
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--- tags/1.6.1.5-rc1/ChangeLog (added)
+++ tags/1.6.1.5-rc1/ChangeLog Tue Aug 11 16:32:27 2009
@@ -1,0 +1,56863 @@
+2009-08-11 Tilghman Lesher <tlesher at digium.com>
+
+ * Asterisk 1.6.1.5-rc1 released
+
+2009-08-10 19:51 +0000 [r211569-211586] Tilghman Lesher <tlesher at digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500
+ (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
+ Aug 2009) | 1 line Conversion specifiers, not format specifiers
+ ........ ................
+
+ * channels/chan_iax2.c, main/channel.c, main/cdr.c, res/ael/pval.c,
+ apps/app_setcallerid.c, main/manager.c, apps/app_rpt.c,
+ main/asterisk.c, funcs/func_rand.c, apps/app_dahdibarge.c,
+ res/res_config_pgsql.c, funcs/func_timeout.c,
+ codecs/codec_speex.c, apps/app_record.c, apps/app_morsecode.c,
+ main/acl.c, funcs/func_cut.c, cdr/cdr_pgsql.c,
+ apps/app_followme.c, main/enum.c, res/res_config_sqlite.c,
+ agi/eagi-sphinx-test.c, main/config.c, channels/misdn_config.c,
+ channels/chan_dahdi.c, funcs/func_channel.c, apps/app_macro.c,
+ apps/app_sms.c, pbx/pbx_config.c, apps/app_verbose.c, main/dsp.c,
+ apps/app_voicemail.c, apps/app_adsiprog.c, funcs/func_speex.c,
+ channels/chan_sip.c, res/res_limit.c, agi/eagi-test.c,
+ funcs/func_math.c, channels/chan_agent.c, main/utils.c,
+ channels/iax2-provision.c, apps/app_talkdetect.c,
+ main/indications.c, channels/chan_oss.c, main/cli.c,
+ pbx/pbx_loopback.c, res/res_config_curl.c, channels/chan_misdn.c,
+ res/res_smdi.c, apps/app_osplookup.c, channels/chan_skinny.c,
+ pbx/pbx_dundi.c, utils/extconf.c, apps/app_mixmonitor.c,
+ channels/chan_mgcp.c, main/timing.c, doc/CODING-GUIDELINES,
+ main/pbx.c, utils/muted.c, apps/app_readfile.c,
+ apps/app_meetme.c, /, apps/app_privacy.c, apps/app_waituntil.c,
+ cdr/cdr_adaptive_odbc.c, res/res_http_post.c, pbx/dundi-parser.c,
+ res/res_musiconhold.c, apps/app_queue.c, main/netsock.c,
+ utils/frame.c, channels/chan_usbradio.c, funcs/func_enum.c,
+ channels/chan_phone.c, pbx/pbx_spool.c, apps/app_waitforring.c,
+ funcs/func_odbc.c, main/features.c, res/res_agi.c,
+ apps/app_minivm.c, main/http.c, res/snmp/agent.c,
+ res/res_config_ldap.c, apps/app_chanspy.c, apps/app_stack.c,
+ res/res_odbc.c, funcs/func_dialplan.c, main/dnsmgr.c,
+ main/frame.c, apps/app_waitforsilence.c, funcs/func_strings.c,
+ apps/app_disa.c, apps/app_alarmreceiver.c: AST-2009-005
+
+2009-08-10 14:12 +0000 [r211349] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 |
+ file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix
+ retrieval of the port used for the video stream when adding SDP
+ to a SIP message. (closes issue #15121) Reported by: jsmith
+ ........
+
+2009-08-09 15:43 +0000 [r211234-211277] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/astfd.c: Merged revisions 211275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009)
+ | 9 lines Merged revisions 211274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
+ | 2 lines Small oops. Clear the flags which have been checked.
+ ........ ................
+
+ * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 |
+ tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
+ Check for NULL frame, before dereferencing pointer. (closes issue
+ #15617) Reported by: rain ........
+
+2009-08-07 20:17 +0000 [r211115] Russell Bryant <russell at digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009)
+ | 11 lines Recorded merge of revisions 211112 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
+ | 4 lines Resolve a deadlock involving app_chanspy and
+ masquerades. (ABE-1936) ........ ................
+
+2009-08-07 18:19 +0000 [r211047] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 211040 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009)
+ | 21 lines Merged revisions 211038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
+ | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
+ not the membername. This is a partial revert of revision 82590,
+ which was an attempted cleanup, but in reality, it broke
+ QUEUE_MEMBER_LIST, which has always been intended as a method by
+ which component interfaces could be queried from the queue.
+ Membername isn't useful here, because that field cannot be used
+ to obtain further information about the member. See the
+ documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+ QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+ member argument for further justification. (closes issue #15664)
+ Reported by: rain Patches: app_queue-queue_member_list.diff
+ uploaded by rain (license 327) ........ ................
+
+2009-08-07 13:09 +0000 [r210994] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/udptl.c, /: Merged revisions 210992 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 |
+ kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13
+ lines Workaround broken T.38 endpoints that offer tiny
+ MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
+ the maximum IFP size that should be sent to them, rather than the
+ maximum packet payload size. If such an endpoint also requests
+ UDPRedundancy as the error correction mode, we'll end up
+ calculating a tiny maximum IFP size, so small as to be unusable.
+ This patch sets a lower bound on what we'll consider the remote's
+ maximum IFP size to be, assuming that endpoints that do this
+ really can accept larger packets than they've offered to accept.
+ (closes issue #15649) Reported by: dazza76 ........
+
+2009-08-06 21:47 +0000 [r210910-210916] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c, /: Merged revisions 210914 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009)
+ | 14 lines Merged revisions 210913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
+ | 7 lines Because channel information can be accessed outside of
+ the channel thread, we must lock the channel prior to modifying
+ it. (closes issue #15397) Reported by: caspy Patches:
+ 20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy ........ ................
+
+ * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged
+ revisions 210908 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 |
+ tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
+ Allow Gosub to recognize quote delimiters without consuming them.
+ (closes issue #15557) Reported by: rain Patches:
+ 20090723__issue15557.diff.txt uploaded by tilghman (license 14)
+ Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
+ ........
+
+2009-08-06 17:48 +0000 [r210819] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 |
+ file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
+ Accept additional T.38 reinvites after an initial one has been
+ handled. Discussion of this subject has yielded that it is not
+ actually acceptable to change T.38 parameters after the initial
+ reinvite but declining is harsh and can cause the fax to fail
+ when it may be possible to allow it to continue. This patch
+ changes things so that additional T.38 reinvites are accepted but
+ parameter changes ignored. This gives the fax a fighting chance.
+ (closes issue #15610) Reported by: huangtx2009 ........
+
+2009-08-05 20:28 +0000 [r210681] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500
+ (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
+ | 14 lines Dialplan starts execution before the channel setup is
+ complete. * Issue 15655: For the case where dialing is complete
+ for an incoming call, dahdi_new() was asked to start the PBX and
+ then the code set more channel variables. If the dialplan hungup
+ before these channel variables got set, asterisk would likely
+ crash. * Fixed potential for overlap incoming call to erroneously
+ set channel variables as global dialplan variables if the
+ ast_channel structure failed to get allocated. * Added missing
+ set of CALLINGSUBADDR in the dialing is complete case. (closes
+ issue #15655) Reported by: alecdavis ........ ................
+
+2009-08-05 18:57 +0000 [r210567] Leif Madsen <lmadsen at digium.com>
+
+ * doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500
+ (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
+ | 11 lines Update imapstorage.txt documentation. Updated the
+ imapstorage.txt documentation to reflect that issues with
+ c-client versions older than 2007 seem to cause crashing issues
+ that are not seen with more recent versions. Documentation has
+ been updated to reflect this. (closes issue #14496) Reported by:
+ vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks ........ ................
+
+2009-08-04 14:54 +0000 [r210240] Kevin P. Fleming <kpfleming at digium.com>
+
+ * Makefile, /: Merged revisions 210238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug
+ 2009) | 16 lines Merged revisions 210237 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
+ 2009) | 10 lines Eliminate spurious compiler warnings from system
+ headers on *BSD platforms. Ensure that system headers located in
+ /usr/local/include are actually treated as system headers by the
+ compiler, and not as local headers which are subject to warnings
+ from the -Wundef compiler option and others. (closes issue
+ #15606) Reported by: mvanbaak ........ ................
+
+2009-08-01 11:32 +0000 [r209836-209900] Russell Bryant <russell at digium.com>
+
+ * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209887 | russell | 2009-08-01 06:29:25 -0500
+ (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
+ | 5 lines Resolve a valgrind warning about a read from
+ uninitialized memory. (issue #15396) Reported by: aragon ........
+ ................
+
+ * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209839 | russell | 2009-08-01 06:02:07 -0500
+ (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009)
+ | 13 lines Modify how Playtones() is used in Milliwatt() to
+ resolve gain issue. When Milliwatt() was changed internally to
+ use Playtones() so that the proper tone was used, it introduced a
+ drop in gain in the output signal. So, use the playtones API
+ directly and specify a volume argument such that the output
+ matches the gain of the original Milliwatt() code. (closes issue
+ #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff
+ uploaded by russell (license 2) Tested by: rue_mohr ........
+ ................
+
+ * /, main/event.c: Merged revisions 209835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 |
+ russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines
+ Fix ast_event_queue_and_cache() to actually do the cache() part.
+ (closes issue #15624) Reported by: ffossard Tested by: russell
+ ........
+
+2009-08-01 01:25 +0000 [r209781] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/misdn/isdn_lib.c, utils/frame.c, /, main/Makefile,
+ channels/misdn/ie.c: Merged revisions 209760-209761 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500
+ (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
+ 2009) | 7 lines Minor changes inspired by testing with latest
+ GCC. The latest GCC (what will become 4.5.x) has a few new
+ warnings, that in these cases found some either downright buggy
+ code, or at least seriously poorly designed code that could be
+ improved. ........ ................ r209761 | kpfleming |
+ 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert
+ accidental Makefile change. ................
+
+2009-07-31 21:58 +0000 [r209714] Russell Bryant <russell at digium.com>
+
+ * /, main/event.c: Merged revisions 209711 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 |
+ russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines
+ Fix some places where ast_event_type was used instead of
+ ast_event_ie_type. ........
+
+2009-07-30 18:46 +0000 [r209593] David Brooks <dbrooks at digium.com>
+
+ * include/asterisk/abstract_jb.h, channels/chan_dahdi.c,
+ contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c,
+ codecs/lpc10/pitsyn.c, channels/chan_console.c: Merged revisions
+ 209554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
+ dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
+ Fixes numerous spelling errors. Patch submitted by alecdavis.
+ (closes issue #15595) Reported by: alecdavis ........
+
+2009-07-30 14:40 +0000 [r209517] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 |
+ mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8
+ lines Fix a crash that can result if text codecs are allowed but
+ textsupport is disabled. (closes issue #15596) Reported by:
+ fabled Patches: sip-red.patch uploaded by fabled (license 448)
+ ........
+
+2009-07-28 00:19 +0000 [r209327] Tilghman Lesher <tlesher at digium.com>
+
+ * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009)
+ | 9 lines Merged revisions 209315 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
+ | 2 lines Publish French extra sounds ........ ................
+
+2009-07-27 21:44 +0000 [r209262-209281] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 |
+ kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7
+ lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE
+ messages about T.38 negotiation in debug level 1 messages, clean
+ up some looping logic, and correct an improper use of ast_free()
+ for freeing an ast_frame. ........
+
+ * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 |
+ kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10
+ lines Make T.38 switchover in ReceiveFAX synchronous. In receive
+ mode, if the channel that ReceiveFAX is running on supports T.38,
+ we should *always* attempt to switch T.38, rather than listening
+ for an incoming CNG tone and only triggering on that. The channel
+ may be using a low-bitrate codec that distorts the CNG tone, the
+ sending FAX endpoint may not send CNG at all, or there could be a
+ variety of other reasons that we don't detect it, but in all
+ those cases if T.38 is available we certainly want to use it.
+ ........
+
+2009-07-27 20:57 +0000 [r209237] Mark Michelson <mmichelson at digium.com>
+
+ * main/rtp.c, /: Merged revisions 209235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 |
+ mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5
+ lines Gracefully handle malformed RTP text packets. AST-2009-004
+ ........
+
+2009-07-27 20:28 +0000 [r209233] David Brooks <dbrooks at digium.com>
+
+ * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c,
+ channels/chan_vpb.cc, res/res_smdi.c, /,
+ include/asterisk/module.h, main/features.c, res/res_agi.c: Merged
+ revisions 209098 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 |
+ dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
+ Fixing typos. Replaces "recieved" with "received" and "initilize"
+ with "initialize" (closes issue #15571) Reported by: alecdavis
+ ........
+
+2009-07-27 20:17 +0000 [r209134-209199] Mark Michelson <mmichelson at digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
+ 2009) | 9 lines Honor channel's music class when using realtime
+ music on hold. (closes issue #15051) Reported by: alexh Patches:
+ 15051.patch uploaded by mmichelson (license 60) Tested by: alexh
+ ........
+
+ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
+ 209132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
+ 2009) | 24 lines Merged revisions 209131 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
+ 2009) | 18 lines Allow for UDPTL to use only even-numbered ports
+ if desired. There are some VoIP providers out there that will not
+ accept SDP offers with odd numbered UDPTL ports. While it is my
+ personal opinion that these VoIP providers are misinterpreting
+ RFC 2327, it really is not a big deal to play along with their
+ silly little games. Of course, since restricting UDPTL ports to
+ only even numbers reduces the range of available ports by half,
+ so the option to use only even port numbers is off by default. A
+ user can enable the behavior by setting use_even_ports=yes in
+ udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
+ 15182.patch uploaded by mmichelson (license 60) Tested by:
+ CGMChris ........ ................
+
+2009-07-27 15:40 +0000 [r209058] Kevin P. Fleming <kpfleming at digium.com>
+
+ * Makefile, /: Merged revisions 209056 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 |
+ kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10
+ lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
+ underscore-variants to sub-makes. During the recent Makefile
+ improvements I made, it seemed the 'make' was automatically
+ carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
+ I removed the explict export of them. However, there are some
+ circumstances where make does this, and some where it does not,
+ so I've brought them back to ensure they are always exported. I
+ also removed an extraneous double setting of _ASTLDFLAGS on *BSD
+ platforms. ........
+
+2009-07-27 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.1.2 released
+
+2009-06-05 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.1.1 released
+
+2009-06-04 David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Additional updates for AST-2009-001
+
+2009-06-04 David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001
+
+2009-04-27 Leif Madsen <lmadsen at digium.com>
+
+ * Create Asterisk 1.6.1.0
+
+2009-04-20 Leif Madsen <lmadsen at digium.com>
+
+ * Create Asterisk 1.6.1.0-rc5
+
+2009-04-20 17:08 +0000 [r189352] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 |
+ file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
+ Fix a bug with non-UDP connections that caused dialogs to not get
+ freed. This issue crept up because of a reference count issue on
+ non-UDP based dialogs. The dialog reference count was increased
+ when transmitting a packet reliably but never decreased. This
+ caused the dialog structure to hang around despite being unlinked
+ from the dialogs container. (closes issue #14919) Reported by:
+ vrban ........
+
+2009-04-20 14:06 +0000 [r189280] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c, /: Merged revisions 189278 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr
+ 2009) | 18 lines Merged revisions 189277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
+ 2009) | 12 lines Move the check for chan->fdno == -1 to after the
+ zombie/hangup check. Many users were finding that their hung up
+ channels were staying up and causing 100% CPU usage. (issue
+ #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
+ uploaded by mmichelson (license 60) Tested by: falves11, bamby
+ ........ ................
+
+2009-04-18 01:38 +0000 [r189206] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 189204 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500
+ (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009)
+ | 12 lines Fixed autologoff in agents.conf not working when agent
+ logs in via AgentLogin app An agent logs in by calling an
+ extension that calls the AgentLogin app. In agents.conf
+ ackcall=always is set, so when they get a call they have the
+ choice to either acknowledge it or ignore it. autologoff=10 is
+ set as well, so if the agent ignores the call over 10sec one may
+ assume that the agent should be logged out (and in this case
+ hungup on as well), but this was not happening. (closes issue
+ #14091) Reported by: evandro Patches: autologoff.diff uploaded by
+ dvossel (license 671) Review:
+ http://reviewboard.digium.com/r/225/ ........ ................
+
+2009-04-17 21:55 +0000 [r189139] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
+ revisions 189137 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009)
+ | 17 lines Merged revisions 188833,189134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
+ | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
+ Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
+ rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
+ Modifed/added some debug messages. JIRA ABE-1835 ........
+ ................
+
+2009-04-17 20:21 +0000 [r189103] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 |
+ mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13
+ lines Prevent a crash when SIP blonde transferring an unbridged
+ call. If one attempts to use the attended transfer button on a
+ SIP phone to transfer an unbridged call (such as a call to an
+ IVR) but hangs up while the target of the transfer is still
+ ringing, we need to not crash. The problem was that ast_hangup
+ was called from outside the channel thread. AST-211 ........
+
+2009-04-17 19:46 +0000 [r189080] Sean Bright <sean.bright at gmail.com>
+
+ * main/asterisk.c, /: Merged revisions 189077 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 |
+ seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1
+ line Fix copy/paste error with 'transmit silence' flag. ........
+
+2009-04-17 17:33 +0000 [r189069] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, /: Merged revisions 189010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr
+ 2009) | 12 lines Merged revisions 189009 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
+ 2009) | 5 lines Make Busy() application set the CDR disposition
+ to BUSY. (closes issue #14306) Reported by: cristiandimache
+ ........ ................
+
+2009-04-17 14:48 +0000 [r188940-188949] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) |
+ 22 lines Merged revisions 188946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
+ 15 lines Fix a bug where a value used to create the channel name
+ was bogus. This commit fixes the scenario where an incoming call
+ is authenticated using a peer entry. Previously the channel name
+ was created using either the username setting from the sip.conf
+ entry or the IP address that the call came from. Now the channel
+ name will be created using the peer name itself. This commit will
+ not change the way the channel name is generated for users or
+ friends. (closes issue #14256) Reported by: Nick_Lewis Patches:
+ chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by:
+ Nick_Lewis, file ........ ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri,
+ 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4
+ lines Fix a situation where the DAHDI channel private structure
+ lock was not unlocked when it should have been. (issue AST-210)
+ ........ ................
+
+2009-04-16 22:05 +0000 [r188776-188838] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009)
+ | 14 lines Merged revisions 188835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
+ | 7 lines Only update realtime, if global option rtupdate !=
+ false (closes issue #14885) Reported by: deepesh Patches:
+ 20090413__bug14885.diff.txt uploaded by tilghman (license 14)
+ Tested by: deepesh ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500
+ (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009)
+ | 4 lines Umask should not be exported into global namespace.
+ (closes issue #14912) Reported by: jcapp ........
+ ................
+
+2009-04-15 22:12 +0000 [r188649] David Vossel <dvossel at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500
+ (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009)
+ | 12 lines National prefix inserted even when caller ID not
+ available When the caller ID is restricted, the expected behavior
+ is for the caller id to be blank. In chan_dahdi, the national
+ prefix is placed onto the callers number even if its restricted
+ (empty) causing the caller id to be the national prefix rather
+ than blank. (closes issue #13207) Reported by: shawkris Patches:
+ national_prefix.diff uploaded by dvossel (license 671) Review:
+ http://reviewboard.digium.com/r/220/ ........ ................
+
+2009-04-15 20:20 +0000 [r188473-188596] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/file.c: Merged revisions 188585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr
+ 2009) | 13 lines Merged revisions 188582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
+ 2009) | 7 lines Update ast_readvideo_callback to match
+ ast_readaudio_callback. This fixes potential refcount errors that
+ may occur on ast_filestreams. AST-208 ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 188470 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 |
+ mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3
+ lines Fix a couple of queue member reference leaks. ........
+
+2009-04-14 17:43 +0000 [r188254-188415] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c, /: Merged revisions 188413 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 |
+ file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix
+ an incorrect clock rate when sending T140 text. (closes issue
+ #14029) Reported by: epicac ........
+
+ * /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 |
+ file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix
+ a bug with the change I made yesterday to outbound proxy support.
+ Per discussion with oej on IRC we need the actual IP address, not
+ the outbound proxy IP address, in the sa field. Upon further
+ inspection this should make the behaviour of all other uses of
+ the outbound proxy in the code. ........
+
+2009-04-14 05:46 +0000 [r188208-188212] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c, /: Merged revisions 188210 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 |
+ tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines
+ As suggested by Russell, warn users when their dialplan arguments
+ contain pipes, but not commas. ........
+
+ * /, utils/smsq.c: Merged revisions 188206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 |
+ tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines
+ Application delimiter is ',', not '|'. (closes issue #14881)
+ Reported by: stegro Patches: smsq.patch uploaded by stegro
+ (license 752) ........
+
+2009-04-13 19:33 +0000 [r188104] Mark Michelson <mmichelson at digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 188102 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr
+ 2009) | 5 lines Fix another crash related to cached realtime
+ music on hold. This was another off-by-one problem caused by
+ moh_register. ........
+
+2009-04-13 16:32 +0000 [r188069] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 |
+ file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
+ Fix a bug where using an outbound proxy would cause the local
+ address to be 127.0.0.1. Copy the outbound proxy IP address into
+ the SIP dialog structure as the IP address we will be sending to.
+ This has to be done because the logic that determines what local
+ IP address to use in the SIP messages is not aware of an outbound
+ proxy being in place. It only knows what IP address we are
+ sending to. (closes issue #12006) Reported by: mnicholson
+ ........
+
+2009-04-13 14:20 +0000 [r188038] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 188032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 |
+ mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6
+ lines Set all queue variables on both the caller and member
+ channels. This allows for the variables to be accessed if a
+ member macro is run. Thanks to Grigoriy Puzankin for bringing
+ this up on the -dev list. ........
+
+2009-04-10 20:28 +0000 [r187914] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/Makefile, /: Merged revisions 187906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 |
+ jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
+ Fix module embedding for chan_h323. Include libchanh323.a in the
+ modules.link file so that all the symbols can be resolved at link
+ time. (closes issue #11966) Reported by: dome Patches:
+ issue_11966.patch uploaded by kpfleming (license 421) Tested by:
+ jpeeler ........
+
+2009-04-10 17:30 +0000 [r187767] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/scripts/sip-friends.sql,
+ contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500
+ (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10
+ Apr 2009) | 2 lines Add lastms column to the contributed table
+ designs ........ ................
+
+2009-04-10 16:54 +0000 [r187723] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, build_tools/embed_modules.xml: Merged revisions 187721 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10
+ Apr 2009) | 5 lines clean up some patterns for files to remove
+ add embedding support for bridge and test modules ........
+
+2009-04-10 16:03 +0000 [r187678] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 |
+ tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines
+ Ensure pvt is not NULL before dereferencing it. (closes issue
+ #14784) Reported by: pj ........
+
+2009-04-10 16:00 +0000 [r187676] Russell Bryant <russell at digium.com>
+
+ * tests/test_heap.c, /: Merged revisions 187675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187675 |
+ russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines
+ Disable test modules by default. ........
+
[... 56164 lines stripped ...]
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