[asterisk-commits] lmadsen: tag 1.8.0-rc3 r290746 - /tags/1.8.0-rc3/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Oct 7 11:52:24 CDT 2010
Author: lmadsen
Date: Thu Oct 7 11:52:22 2010
New Revision: 290746
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=290746
Log:
Importing files for 1.8.0-rc3 release.
Added:
tags/1.8.0-rc3/.lastclean (with props)
tags/1.8.0-rc3/.version (with props)
tags/1.8.0-rc3/ChangeLog (with props)
Added: tags/1.8.0-rc3/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-rc3/.lastclean?view=auto&rev=290746
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==============================================================================
--- tags/1.8.0-rc3/ChangeLog (added)
+++ tags/1.8.0-rc3/ChangeLog Thu Oct 7 11:52:22 2010
@@ -1,0 +1,25030 @@
+2010-10-07 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.0-rc3 Released.
+
+2010-10-07 11:00 +0000 [r290713] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c, /: Merged revisions 290712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010)
+ | 4 lines Don't crash when Set() is called without a value.
+ Review: https://reviewboard.asterisk.org/r/949/ ........
+
+2010-10-06 21:22 +0000 [r290648-290674] David Vossel <dvossel at digium.com>
+
+ * channels/chan_gtalk.c: Fixes commented out code to use #if 0
+ instead. Thanks to rmudgett for catching this!
+
+ * channels/chan_gtalk.c: Fixes gtalk outbound DTMF to work
+ properly. Outbound DTMF with gtalk needs to be done within the
+ RTP stream. I discovered this after investigating a packet
+ capture from the gmail client. Instead of performing jingle
+ signaling DTMF, the gtalk servers expect all DTMF to arrive on
+ the RTP stream using RFC2833 way of doing things. Chan_gtalk also
+ had an issue with negotiating RTP payload type 106 for the
+ telephony-event and then sending DTMF as payload 101. This has
+ been resolved by always negotiating 101 as the payload type like
+ we do everywhere else. With this patch, incoming google voice
+ calls forwarded to Asterisk via gtalk work.
+
+2010-10-06 18:50 +0000 [r290614] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_dial.c: Merged revision 290613 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed,
+ 06 Oct 2010) | 5 lines Eliminate a redundant test for
+ AST_CONTROL_REDIRECTING. Eliminate redundant test for
+ AST_CONTROL_REDIRECTING that prevents running the redirecting
+ interception macro if it is defined. ..........
+
+2010-10-06 13:49 +0000 [r290576] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/file.c: Merged revisions 290575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010)
+ | 8 lines Allow streaming audio from a pipe. (closes issue
+ #18001) Reported by: jamicque Patches:
+ 20100926__issue18001.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamicque ........
+
+2010-10-06 04:35 +0000 [r290542] Terry Wilson <twilson at digium.com>
+
+ * res/res_rtp_asterisk.c: Don't try to send RTP when remote_address
+ is null It is possible for ast_rtp_stop() to be called which will
+ clear the remote address and cause the sendto to fail and spam
+ warnings. Don't send in this case.
+
+2010-10-05 22:23 +0000 [r290479-290506] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Fixes uninitialized memory problem in 'iax2
+ set debug peer' option.
+
+ * include/asterisk/jingle.h, channels/chan_gtalk.c,
+ res/res_jabber.c, include/asterisk/jabber.h: Fixes chan_gtalk to
+ work with gmail client This patch was written by Philippe Sultan
+ (phsultan). Thanks for keeping this up to date!
+
+2010-10-05 20:23 +0000 [r290408] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_jabber.c, /: Merged revisions 290396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500
+ (Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
+ | 8 lines Fix a crash by ensuring that we don't alter memory
+ after it's freed. (closes issue #17387) Reported by: jmls
+ Patches: 20100726__issue17387.diff.txt uploaded by tilghman
+ (license 14) Tested by: jmls ........ ................
+
+2010-10-05 20:09 +0000 [r290376-290378] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Resolves dnsmgr memory corruption in
+ chan_iax2. (closes issue #17902) Reported by: afried Patches:
+ issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+ afried, russell, dvossel Review:
+ https://reviewboard.asterisk.org/r/965/
+
+ * /, apps/app_directed_pickup.c: Merged revisions 290375 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010)
+ | 10 lines Fixes PickupChan() not working with full channel name.
+ (closes issue #18011) Reported by: schern Patches:
+ app_directed_pickup.c.2.patch uploaded by schern (license 995)
+ app_directed_pickup.c.trunk.patch uploaded by schern (license
+ 995) Tested by: schern, dvossel ........
+
+2010-10-05 14:15 +0000 [r290066-290289] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, configure.ac: Restore run directory for OS X, as well
+ as standardizing some other paths to Mac OS X.
+
+ * pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
+ pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c,
+ pbx/ael/ael-test/ref.ael-vtest17, /,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3:
+ Merged revisions 290254 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010)
+ | 11 lines Change new pattern matcher to regard dashes the same
+ as the old pattern matcher -- as visual candy to be ignored. Also
+ change the AEL parser to not generate dashes within extensions,
+ as those dashes would be ignored. Update the AEL tests to match
+ this behavior. (closes issue #17366) Reported by: murf Patches:
+ 20100727__issue17366.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+ * /, configure, configure.ac: Merged revisions 290201 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500
+ (Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
+ Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
+ ................
+
+ * /, configure, configure.ac: Merged revisions 290101 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500
+ (Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
+ Oct 2010) | 2 lines Automatically re-run configure test for
+ menuselect, when the relevant makeopts settings change. ........
+ ................
+
+ * pbx/pbx_spool.c: Get notification only when file is closed, not
+ when created. (closes issue #17924) Reported by: mkeuter Patches:
+ asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946)
+ Tested by: abelbeck
+
+2010-10-02 17:57 +0000 [r290026] Kevin P. Fleming <kpfleming at digium.com>
+
+ * contrib/scripts/get_mp3_source.sh: Allow users to pass additional
+ arguments to the Subversion command that obtains the MP-3 source
+ code. (reported on IRC by jmls)
+
+2010-10-02 08:56 +0000 [r289951] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c, /: Merged revisions 289950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör,
+ 02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
+ lines Add documentation for undocumented option to AMI action
+ originate ........ ................
+
+2010-10-02 04:46 +0000 [r289875] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 289874 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500
+ (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010)
+ | 8 lines When forwarding a message, a prepend means that the
+ filesystem will always have a better copy. (closes issue #17803)
+ Reported by: dpetersen Patches: 20100923__issue17803.diff.txt
+ uploaded by tilghman (license 14) Tested by: dpetersen ........
+ ................
+
+2010-10-02 02:43 +0000 [r289840] Jeff Peeler <jpeeler at digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, /, channels/chan_sip.c: Merged revisions
+ 289798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500
+ (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
+ | 15 lines Change RFC2833 DTMF event duration on end to report
+ actual elapsed time. The scenario here is with a non P2P early
+ media session. The reported time length of DTMF presses are
+ coming up short when sending to the remote side. Currently the
+ event duration is a running total that is incremented when
+ sending continuation packets. These continuation packets are only
+ triggered upon incoming media from the remote side, which means
+ that the running total probably is not going to end up matching
+ the actual length of time Asterisk received DTMF. This patch
+ changes the end event duration to be lengthened if it is detected
+ that the end event is going to come up short. Review:
+ https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
+ ................
+
+2010-10-01 17:19 +0000 [r289718] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
+ 289704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400
+ (Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
+ 2010) | 6 lines Disable debugging by default and reformat .config
+ file. Review: https://reviewboard.asterisk.org/r/929/ ........
+ ................
+
+2010-10-01 16:22 +0000 [r289701] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500
+ (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
+ | 14 lines Ensure user portion of SIP URI matches dialplan when
+ using encoded characters. This commit takes a simliar approach to
+ 288112 and checks the dialplan to determine the proper action for
+ an incoming contact header as to whether or not it should be
+ decoded or not. sip_new was blindly always decoding the
+ extension, which also caused the outgoing contact header to be
+ incorrect as well as failing to match the encoded extension in
+ the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
+ bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
+ wdoekes ........ ................
+
+2010-10-01 09:42 +0000 [r289622] Stefan Schmidt <sst at sil.at>
+
+ * channels/chan_sip.c: don't iterate through all dialogs to find
+ and delete old subscribes On every incoming subscribe there is a
+ iteration through all dialogs to find old subscribes and delete
+ them. This is slow and not RFC conform. This was only needed in
+ 1.2 cause a subscribe was not deleted when a dialog was
+ destroyed, after 1.4 a subscribe get removed when its dialog is
+ destroyed. (closes issue #17950) Reported by: schmidts Tested by:
+ schmidts Review: https://reviewboard.asterisk.org/r/901/
+
+2010-09-30 20:23 +0000 [r289581] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_env.c: Solaris fixes.
+
+2010-09-30 19:53 +0000 [r289554] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289553 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep
+ 2010) | 4 lines Properly handle channel allocation failures duing
+ invites with replaces. ABE-2588 ........
+
+2010-09-30 19:28 +0000 [r289549] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Merged revision 289547 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu,
+ 30 Sep 2010) | 10 lines In chan_misdn, the
+ DivertingLegInformation2 DivertingNr is garbage when the number
+ is restricted. The same thing happens with
+ DivertingLegInformation1 DivertedTo number. The
+ misdn_PresentedNumberUnscreened_extract() extracted the
+ Unscreened PartyNumber field unconditionally. It now checks the
+ presented number unscreened type to see if the PartyNumber was
+ even present. JIRA ABE-2595 ..........
+
+2010-09-30 17:50 +0000 [r289543] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/localtime.h, main/stdtime/localtime.c,
+ tests/test_time.c, tests/test_utils.c, res/res_agi.c: More
+ Solaris compatibility fixes
+
+2010-09-30 15:39 +0000 [r289426] Russell Bryant <russell at digium.com>
+
+ * apps/app_sms.c, /: Merged revisions 289425 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289425 | russell | 2010-09-30 10:37:29 -0500
+ (Thu, 30 Sep 2010) | 15 lines Merged revisions 289424 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010)
+ | 8 lines Fix a crash in app_sms. Since the data being passed to
+ the generator callback is on the stack of the SMS() application,
+ we must ensure that the generator is stopped before the
+ application exits. ABE-2587 ........ ................
+
+2010-09-29 21:12 +0000 [r289340] Jason Parker <jparker at digium.com>
+
+ * main/channel.c, /, main/features.c: Merged revisions 289339 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289339 | qwell | 2010-09-29 16:03:47 -0500
+ (Wed, 29 Sep 2010) | 15 lines Merged revisions 289338 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) |
+ 8 lines Allow a manager originate to succeed on forwarded
+ devices. The timeout to wait for an answer was being set to 0
+ when a device forwarded to another extension. We don't always
+ need the timeout set like this, so make it an optional parameter,
+ and don't use it in this case. ABE-2544 ........ ................
+
+2010-09-29 20:27 +0000 [r289336] Leif Madsen <lmadsen at digium.com>
+
+ * configs/res_ldap.conf.sample, /: Merged revisions 289334 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29 Sep 2010)
+ | 1 line Update sample documentation to note md5secret
+ requirements. ........
+
+2010-09-29 20:20 +0000 [r289333] Russell Bryant <russell at digium.com>
+
+ * res/res_config_ldap.c, /: Merged revisions 289332 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29
+ Sep 2010) | 4 lines Don't completely ignore md5secret from LDAP
+ if the value does not begin with {md5}. This fixes a problem that
+ lmadsen ran in to where md5secret was not working for him.
+ ........
+
+2010-09-29 17:53 +0000 [r289268-289300] Matthew Nicholson <mnicholson at digium.com>
+
+ * configs/res_fax.conf.sample: Add 'ecm' to the sample fax config
+ file
+
+ * main/channel.c: Update the CDR record when
+ ast_channel_set_caller_event() is called (related to issue
+ #17569) Reported by: tbelder
+
+2010-09-29 16:16 +0000 [r289253] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c: Make development error message indicate which
+ channel.
+
+2010-09-29 15:04 +0000 [r289179] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c, /: Merged revisions 289178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500
+ (Wed, 29 Sep 2010) | 15 lines Merged revisions 289177 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep
+ 2010) | 8 lines Set the caller id on CDRs when it is set on the
+ parent channel. (closes issue #17569) Reported by: tbelder
+ Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
+ tbelder ........ ................
+
+2010-09-28 18:18 +0000 [r289104] Tilghman Lesher <tlesher at digium.com>
+
+ * makeopts.in, apps/app_voicemail.c, Makefile, tests/test_time.c,
+ configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, tests/test_utils.c,
+ configure.ac: Solaris compatibility fixes Review:
+ https://reviewboard.asterisk.org/r/942/
+
+2010-09-28 18:18 +0000 [r289099] Brett Bryant <bbryant at digium.com>
+
+ * main/channel.c, /: Merged revisions 289095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289095 | bbryant | 2010-09-28 14:14:19 -0400
+ (Tue, 28 Sep 2010) | 21 lines Merged revisions 289094 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010)
+ | 14 lines Fixes an issue with the Newchannel AMI event during
+ the Masquerading process. Fixes an issue with the Newchannel AMI
+ event during the Masquerading process, where no Newchannel AMI
+ event was generated for the psuedo channel used during the
+ masquerading process. (closes issue #17987) Reported by:
+ RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish
+ (license 1122) Tested by: RadicAlish Review:
+ https://reviewboard.asterisk.org/r/937/ ........ ................
+
+2010-09-28 01:04 +0000 [r289054-289057] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: Avoid deadlock processing incoming AOC-E
+ messages. Deadlock avoidance for the owner channel was not done
+ when processing incoming AOC-E messages.
+
+ * channels/sig_pri.c: Revert stuff not ready for commit in
+ -r289054.
+
+ * channels/sig_pri.c, channels/chan_sip.c: Break up long
+ ast_manager_event_multichan() event lines.
+
+2010-09-27 18:37 +0000 [r288961] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Still build SIP, even if res_crypto cannot
+ be built (use, not depend). (closes issue #18062) Reported by: a
+ user on the mailing list
+
+2010-09-27 13:03 +0000 [r288925-288927] Russell Bryant <russell at digium.com>
+
+ * res/res_agi.c: Fix some documentation typos and spelling errors.
+
+ * res/res_agi.c: Fix a documentation spelling error.
+
+2010-09-24 17:58 +0000 [r288821-288852] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Append Retry-After header on 500 error
+ response to Re-INVITE according to RFC3261 section 14.2. ABE-2301
+
+ * channels/chan_sip.c: Inspect Require header on BYE transaction
+ according to RFC3261 section 8.2.2.3. ABE-2293
+
+2010-09-24 16:02 +0000 [r288748] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 288747 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288747 | twilson | 2010-09-24 08:37:39 -0700
+ (Fri, 24 Sep 2010) | 12 lines Merged revisions 288746 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010)
+ | 5 lines Don't fail a masquerade if it is already being hung up
+ This avoids noise on some Local channel situations where we don't
+ use /n. Thanks to Alec Davis for the suggestion. ........
+ ................
+
+2010-09-24 13:54 +0000 [r288606-288713] Tilghman Lesher <tlesher at digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 288712 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24
+ Sep 2010) | 5 lines Solaris won't printf a NULL. (closes issue
+ #18041) Reported by: asgaroth ........
+
+ * main/asterisk.exports.in: Export timersub for platforms which do
+ not have it
+
+ * include/asterisk/channel.h, cdr/cdr_pgsql.c, /, configure,
+ include/asterisk/autoconfig.h.in, include/asterisk/compat.h,
+ main/strcompat.c, configure.ac: Merged revisions 288637 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288637 | tilghman | 2010-09-23 22:36:01 -0500
+ (Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23
+ Sep 2010) | 2 lines Solaris compatibility fixes ........
+ ................
+
+ * CHANGES: Add note about the checkhangup option of ${CHANNEL()}
+
+2010-09-23 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.0-rc2 Released.
+
+2010-09-23 18:05 +0000 [r288507-288572] Terry Wilson <twilson at digium.com>
+
+ * main/manager.c: Make AMI honor enabled=no (closes issue #18040)
+ Reported by: twilson Review:
+ https://reviewboard.asterisk.org/r/938/
+
+ * channels/chan_local.c, /: Merged revisions 288500 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288500 | twilson | 2010-09-22 16:10:09 -0700
+ (Wed, 22 Sep 2010) | 15 lines Merged revisions 288499 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010)
+ | 8 lines Don't let a Local channel get bridged to itself If a
+ local channel gets bridged to itself, it becomes orphaned with no
+ devices left to actually tell it to hang up. This patch modifies
+ local_fixup() to detect this case and deny it. Review:
+ https://reviewboard.asterisk.org/r/934 ........ ................
+
+2010-09-22 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.0-rc1 Released.
+
+2010-09-22 17:49 +0000 [r288345-288418] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288417 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500
+ (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
+ | 5 lines RFC3261 section 12.2 explicitly says out of order
+ requests are responded with a 500 Server Internal Error response.
+ ABE-2458 ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 288344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500
+ (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22
+ Sep 2010) | 2 lines During check_pendings, if the dialog is
+ terminated with a CANCEL, change the invitestate to INV_CANCEL
+ like in sip_hangup. ........ ................
+
+2010-09-22 16:45 +0000 [r288341] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 288340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288340 | russell | 2010-09-22 11:44:13 -0500
+ (Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
+ | 11 lines Fix a 100% CPU consumption problem when setting
+ console=yes in asterisk.conf. The handling of -c and console=yes
+ should be the same, but they were not. When you specify -c, it
+ sets both a flag for console module and for asterisk not to
+ fork() off into the background. The handling of console=yes only
+ set console mode, so you would end up with a background process()
+ trying to run the Asterisk console and freaking out since it
+ didn't have anything to read input from. Thanks to beagles for
+ reporting and helping debug the problem! ........
+ ................
+
+2010-09-22 15:14 +0000 [r288268] Tilghman Lesher <tlesher at digium.com>
+
+ * UPGRADE.txt, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /:
+ Merged revisions 288267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288267 | tilghman | 2010-09-22 10:11:09 -0500
+ (Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
+ | 9 lines Allow the encoding to be set, in case local charset
+ does not agree with database. (closes issue #16940) Reported by:
+ jamicque Patches: 20100827__issue16940.diff.txt uploaded by
+ tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+ r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
+ | 5 lines Document addition of encoding parameter. (issue #16940)
+ Reported by: jamicque ........ ................
+
+2010-09-22 00:06 +0000 [r288194] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 288193 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500
+ (Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010)
+ | 26 lines In chan_iax2.c:schedule_delivery() calls
+ ast_bridged_channel() on an unlocked channel. Near the beginning
+ of schedule_delivery(), ast_bridged_channel() is called on
+ iaxs[fr->callno]->owner. However, the channel is not locked,
+ which can result in ast_bridged_channel() crashing should
+ owner->tech change to a technology that doesn't implement
+ bridged_channel. I also fixed the other calls to
+ ast_bridged_channel() in chan_iax2.c since the owner lock was not
+ held there either. Converted the existing channel deadlock
+ avoidance to use iax2_lock_owner(). Using the new function
+ simplified some awkward code. In the process of fixing the
+ locking on ast_bridged_channel(), I also found a memory leak in
+ socket_process() for v1.6.2 and v1.8. The local struct variable
+ ies.vars is not freed on early/abnormal function exits. (closes
+ issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
+ uploaded by rmudgett (license 664) Review:
+ https://reviewboard.asterisk.org/r/926/ ........ ................
+
+2010-09-21 22:57 +0000 [r288159] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 288113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500
+ (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
+ | 15 lines Try both the encoded and unencoded subscription URI
+ for a match in hints. When a phone sends an encoded URI for a
+ subscription, the URI is not matched with the actual hint that is
+ in decoded format. For example, if we have an extension with a
+ hint that is named: "#5601" or "*5601", the subscription will
+ work fine if the phone subscribes with an already decoded URI,
+ but when it's decoded like "%255601" or "%2A5601", Asterisk is
+ unable to match it with the correct hint. (closes issue #17785)
+ Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
+ uploaded by tilghman (license 14) Tested by: ramonpeek ........
+ ................
+
+2010-09-21 22:26 +0000 [r288157] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 288147 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue,
+ 21 Sep 2010) | 9 lines Setup timer before set_config(). (closes
+ issue #18019) Reported by: Netview Patches: issue_0018019.patch
+ uploaded by pabelanger (license 224) Tested by: Netview ........
+
+2010-09-21 21:03 +0000 [r288079-288082] Richard Mudgett <rmudgett at digium.com>
+
+ * doc/tex/partymanip.tex: Add note in party manipulation chapter on
+ interception macros.
+
+ * apps/app_queue.c, apps/app_dial.c: Simplify locking code for
+ REDIRECTING interception macro when forwarding a call. Simplified
+ the locking code by using a local copy of the redirecting party
+ information in app_dial.c:do_forward() and
+ app_queue.c:wait_for_answer() for launching the REDIRECTING
+ interception macro when a call is forwarded. Reduced the lock
+ time of the 'o->chan' and 'in' channels.
+
+ * main/channel.c: Protect channel access in CONNECTED_LINE and
+ REDIRECTING interception macro launch code.
+
+2010-09-21 19:48 +0000 [r288007] Brett Bryant <bbryant at digium.com>
+
+ * main/channel.c, /: Merged revisions 288006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400
+ (Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010)
+ | 8 lines Add a check to fix a rare segmentation fault you'd get
+ if ast_frdup couldn't allocate memory on the first frame being
+ queued in ast_queue_frame. (closes issue #17882) Reported by:
+ seanbright Tested by: seanbright ........ ................
+
+2010-09-21 19:08 +0000 [r287935] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 287934 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287934 | tilghman | 2010-09-21 14:07:53 -0500
+ (Tue, 21 Sep 2010) | 9 lines Merged revisions 287933 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21
+ Sep 2010) | 2 lines Less than zero is an error, not any non-zero
+ value. ........ ................
+
+2010-09-21 19:02 +0000 [r287931] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c: Revert change in favor of a more targeted fix
+
+2010-09-21 18:32 +0000 [r287929] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Send a "415 Unsupported Media Type" after
+ failure to process sdp due to unknown Content-Encoding header.
+ ABE-2258
+
+2010-09-21 15:53 +0000 [r287897] Richard Mudgett <rmudgett at digium.com>
+
+ * main/features.c: Cut-n-paste error in builtin_blindtransfer().
+
+2010-09-21 15:43 +0000 [r287895] Russell Bryant <russell at digium.com>
+
+ * res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
+ main/acl.c: Don't use ast_strdupa() from within the arguments to
+ a function. (closes issue #17902) Reported by: afried Patches:
+ issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+ russell Review: https://reviewboard.asterisk.org/r/927/
+
+2010-09-21 15:24 +0000 [r287893] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Anonymous callerid needs a "sip:" uri
+ prefix. (closes issue #17981) Reported by: avalentin Patches:
+ sip-anonymous-aastra.patch uploaded by avalentin (license 1107)
+ (plus an additional fix by me) Tested by: avalentin
+
+2010-09-21 13:41 +0000 [r287863] Russell Bryant <russell at digium.com>
+
+ * main/logger.c: Fix a regression in verbose logger processing.
+
+2010-09-21 04:37 +0000 [r287833] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c: Don't generate connected line buffer twice for
+ comparison
+
+2010-09-21 00:00 +0000 [r287760] Brett Bryant <bbryant at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 287759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400
+ (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010)
+ | 16 lines Fix misvalidation of meetme pins in conjunction with
+ the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a
+ user and admin pin setup for your conference, using the user pin
+ would gain you admin priviledges. Also, when no user pin was set,
+ an admin pin was, the 'a' MeetMe flag wasn't used, and the user
+ tried to enter a conference then they were still prompted for a
+ pin and forced to hit #. (closes issue #17908) Reported by: kuj
+ Patches: pins_2.patch uploaded by kuj (license 1111) Tested by:
+ kuj Review: [full review board URL with trailing slash] ........
+ ................
+
+2010-09-20 23:51 +0000 [r287757] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c: Avoid infinite loop with certain local channel
+ connected line updates Compare connected line data before sending
+ a connected line indication to avoid possible loops. Review:
+ https://reviewboard.asterisk.org/r/932/
+
+2010-09-20 23:20 +0000 [r287701] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/channel.c, /: Merged revisions 287685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep
+ 2010) | 18 lines ast_channel_masquerade: Avoid recursive
+ masquerades. Check all 4 combinations of (original/clonechan) *
+ (masq/masqr). Initially original->masq and clonechan->masqr were
+ only checked. It's possible with multiple masq's planned - and
+ not yet executed, that the 'original' chan could already have
+ another masq'd into it - thus original->masqr would be set, that
+ masqr would lost. Likewise for the clonechan->masq. (closes issue
+ #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
+ based on bug16057.diff4.txt uploaded by alecdavis (license 585)
+ Tested by: ramonpeek, davidw, alecdavis ........
+
+2010-09-20 23:14 +0000 [r287683] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: The inalarm flag was not set in sig_analog
+ struct if the port is initially in alarm. Fixed initial inalarm
+ value for sig_analog ports. Along with -r261007, this gets the
+ inalarm flag in sync with chan_dahdi for sig_analog ports.
+ (closes issue #16983)
+
+2010-09-20 22:21 +0000 [r287661] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/channel.c: ast_do_masquerade. Keep channels ao2_container
+ locked while unlink and linking channels. Previously, Masquerade
+ would unlock 'original' and 'clonechan' and allow another masq
+ thread to run. End result would be corrupted memory, and the
+ frequent report 'Bad Magic Number'. (closes issue #17801,#17710)
+ Reported by: notthematrix Patches: Based on bug17801.diff1.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/928
+
+2010-09-20 22:09 +0000 [r287645-287647] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/channel.h, CHANGES, include/asterisk/framehook.h
+ (added), main/channel.c, main/framehook.c (added),
+ funcs/func_frame_trace.c (added): Addition of the FrameHook API
+ (AKA AwesomeHooks) So far all our tools for viewing and
+ manipulating media streams within Asterisk have been entirely
+ focused on audio. That made sense then, but is not scalable now.
+ The FrameHook API lets us tap into and manipulate _ANY_ type of
+ media or signaling passed on a channel present today or in the
+ future. This tool is a step in the direction of expanding
+ Asterisk's boundaries and will help generate some rather
+ interesting applications in the future. In addition to the
+ FrameHook API, a simple dialplan function exercising the api has
+ been included as well. This function is called FRAME_TRACE().
[... 24287 lines stripped ...]
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