[asterisk-commits] lmadsen: tag 1.6.0.27-rc1 r256158 - /tags/1.6.0.27-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Apr 5 10:04:05 CDT 2010
Author: lmadsen
Date: Mon Apr 5 10:04:03 2010
New Revision: 256158
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=256158
Log:
Importing files for 1.6.0.27-rc1 release.
Added:
tags/1.6.0.27-rc1/.lastclean (with props)
tags/1.6.0.27-rc1/.version (with props)
tags/1.6.0.27-rc1/ChangeLog (with props)
Added: tags/1.6.0.27-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.0.27-rc1/.lastclean?view=auto&rev=256158
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Added: tags/1.6.0.27-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.0.27-rc1/ChangeLog?view=auto&rev=256158
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--- tags/1.6.0.27-rc1/ChangeLog (added)
+++ tags/1.6.0.27-rc1/ChangeLog Mon Apr 5 10:04:03 2010
@@ -1,0 +1,57401 @@
+2010-04-05 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.0.27-rc1 Released
+
+2010-04-02 23:47 +0000 [r256011-256016] Russell Bryant <russell at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 256015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r256015 | russell | 2010-04-02 18:46:45 -0500
+ (Fri, 02 Apr 2010) | 16 lines 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 256010 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010)
+ | 9 lines 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 20:20 +0000 [r255953] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 255952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 |
+ tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines
+ 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 ........
+
+2010-04-01 18:21 +0000 [r255674-255813] Tilghman Lesher <tlesher at digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010)
+ | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue
+ #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt
+ uploaded by tilghman (license 14) ........
+
+ * apps/app_voicemail.c, /: Recorded merge of revisions 255592 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r255592 | tilghman | 2010-03-31 14:13:02 -0500
+ (Wed, 31 Mar 2010) | 22 lines 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 17:54 +0000 [r255507] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_dial.c, configs/sip.conf.sample: Merged revisions 255504
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31
+ Mar 2010) | 5 lines 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 20:57 +0000 [r255324-255411] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_h323.c: Merged revisions 255410 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r255410 | russell | 2010-03-30 15:56:26 -0500
+ (Tue, 30 Mar 2010) | 9 lines 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 255323 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010)
+ | 9 lines 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. ........ ................
+
+2010-03-26 19:24 +0000 [r255054] Leif Madsen <lmadsen at digium.com>
+
+ * /, configs/sip.conf.sample: Merged revisions 255021 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010)
+ | 8 lines 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
+ ........
+
+2010-03-25 20:42 +0000 [r254803] Jason Parker <jparker at digium.com>
+
+ * utils/Makefile, /: Merged revisions 254802 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) |
+ 9 lines 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 20:09 +0000 [r254719] Russell Bryant <russell at digium.com>
+
+ * channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010)
+ | 2 lines chan_usbradio depends on alsa. ........
+
+2010-03-25 19:59 +0000 [r254716] Jason Parker <jparker at digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS
+ issue with out-of-tree modules. Take 2, without ABI breakage this
+ time. Review: https://reviewboard.asterisk.org/r/588/
+
+2010-03-25 17:45 +0000 [r254549-254554] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500
+ (Thu, 25 Mar 2010) | 11 lines 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/chan_sip.c: Undo unnecessary commit. Sean Bright beat me
+ to the punch on this one.
+
+ * channels/chan_sip.c: Fix potential use of uninitialized value.
+
+2010-03-25 17:19 +0000 [r254546] Sean Bright <sean at malleable.com>
+
+ * channels/chan_sip.c: Initialize stream to avoid a compilation
+ error.
+
+2010-03-25 17:05 +0000 [r254540] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix crashes resulting from reading
+ non-existent RTP streams. Specifically, when using the CHANNEL
+ dialplan function, it was possible to crash Asterisk by trying to
+ get the rtpdest of a stream type that is not in use by the
+ channel. This commit fixes that issue.
+
+2010-03-25 17:02 +0000 [r254539] Leif Madsen <lmadsen at digium.com>
+
+ * contrib/scripts/safe_asterisk, /: Make safe_asterisk work on
+ dash/sh/bash etc. Merged from the change to trunk via issue
+ #13111. For some reason the changes there were only done on
+ trunk, and thus were available for 1.6.1 and 1.6.2 when they were
+ branched. Because this change is available on both 1.6.1 and
+ 1.6.2, it makes sense to allow it on the 1.6.0 branch as well.
+ (closes issue #17094) Reported by: stuarth Much thanks to
+ Tilghman and Sean Bright for the help on this merge. Merged
+ revisions 135061 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r135061 |
+ mvanbaak | 2008-08-01 07:17:33 -0500 (Fri, 01 Aug 2008) | 8 lines
+ Make safe_asterisk work on dash/sh/bash etc. (closes issue
+ #13111) Reported by: pabelanger Patches:
+ 2008071901_issue13111_safe_asterisk.diff uploaded by mvanbaak
+ (license 7) Tested by: mvanbaak, pabelanger ........
+
+2010-03-25 16:57 +0000 [r254538] Sean Bright <sean at malleable.com>
+
+ * /: Unblock r135061
+
+2010-03-25 16:19 +0000 [r254466] Terry Wilson <twilson at digium.com>
+
+ * /, main/file.c: Merged revisions 254453 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010)
+ | 9 lines 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 16:11 +0000 [r254455] Mark Michelson <mmichelson at digium.com>
+
+ * main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500
+ (Thu, 25 Mar 2010) | 50 lines 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 15:22 +0000 [r254449] Leif Madsen <lmadsen at digium.com>
+
+ * /, res/res_agi.c: Merged revisions 254446 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 |
+ lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines
+ 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 ........
+
+2010-03-24 17:17 +0000 [r254061-254278] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 254277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010)
+ | 78 lines 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)
+ ........ ................
+
+ * main/channel.c, /: Merged revisions 254050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 |
+ jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines
+ 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 [r254044] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * tests/Makefile, /: Merged revisions 254001 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk ........ r254001 |
+ tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines
+ Change the name of the category 'TEST' to match the name of the
+ subdir ........
+
+2010-03-22 19:57 +0000 [r253803] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, main/features.c: Merged revisions 253800 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar
+ 2010) | 11 lines 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) ........
+ ................
+
+2010-03-20 18:29 +0000 [r253625-253630] Russell Bryant <russell at digium.com>
+
+ * main/sched.c, main/manager.c, main/features.c,
+ apps/app_waituntil.c, main/logger.c: Resolve 1.6.0 compilation
+ issues on FreeBSD.
+
+ * apps/app_dial.c, channels/chan_dahdi.c, main/tcptls.c, /,
+ main/features.c, pbx/pbx_dundi.c, cdr/cdr_pgsql.c,
+ main/stdtime/localtime.c, apps/app_followme.c: Merged revisions
+ 253536-253538,253540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r253536 |
+ russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010) | 4 lines
+ Use SHRT_MAX instead of MAXSHORT. These changes fix build issues
+ I had with this module on FreeBSD. ........ r253537 | russell |
+ 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve a
+ compiler warning on FreeBSD. ........ r253538 | russell |
+ 2010-03-20 06:43:08 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve
+ compiler warnings on FreeBSD. ........ r253540 | russell |
+ 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve
+ more compiler warnings on FreeBSD. ........
+
+ * main/utils.c: Resolve compiler warnings on FreeBSD.
+
+2010-03-18 17:56 +0000 [r253259-253348] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_userevent.c: Slightly different fix for UserEvent docs
+ update. (issue #16961)
+
+ * doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010)
+ | 9 lines 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) ........
+
+2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson at digium.com>
+
+ * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_mgcp.c, channels/chan_sip.c,
+ include/asterisk/rtp.h: Revert API change in release branches
+ This re-renames ast_rtp_update_source to ast_rtp_new_source
+
+2010-03-17 00:31 +0000 [r253031] Leif Madsen <lmadsen at digium.com>
+
+ * /, configs/say.conf.sample: Merged revisions 253028 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500
+ (Tue, 16 Mar 2010) | 13 lines 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) ........ ................
+
+2010-03-16 23:54 +0000 [r252979] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_stack.c, /: Merged revisions 252976 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 |
+ tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines
+ 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 19:01 +0000 [r252768] Russell Bryant <russell at digium.com>
+
+ * utils/Makefile, /: Merged revisions 252767 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010)
+ | 13 lines 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 18:49 +0000 [r252765] Leif Madsen <lmadsen at digium.com>
+
+ * /, configs/extensions.ael.sample: Merged revisions 252762 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500
+ (Tue, 16 Mar 2010) | 15 lines 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) ........ ................
+
+2010-03-15 21:59 +0000 [r252624] Sean Bright <sean at malleable.com>
+
+ * /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 |
+ seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4
+ lines Resolve a crash in SLATrunk when the specified trunk
+ doesn't exist. Reported by philipp64 in #asterisk-dev. ........
+
+2010-03-15 21:53 +0000 [r252620] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions
+ 252619 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010)
+ | 9 lines 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 20:54 +0000 [r252537] Leif Madsen <lmadsen at digium.com>
+
+ * configs/extensions.ael.sample: Merged revisions 252534 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500
+ (Mon, 15 Mar 2010) | 15 lines 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 01:37 +0000 [r252363] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, Makefile,
+ contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged
+ revisions 252362 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010)
+ | 11 lines 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 17:45 +0000 [r252315] Sean Bright <sean at malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar
+ 2010) | 8 lines 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
+ ........
+
+2010-03-13 00:30 +0000 [r252134-252176] Terry Wilson <twilson at digium.com>
+
+ * main/rtp.c: Remove unused field
+
+ * main/rtp.c, channels/chan_mgcp.c, main/channel.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ include/asterisk/rtp.h, channels/chan_h323.c,
+ configs/sip.conf.sample, include/asterisk/frame.h: Merged
+ revisions 252089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 |
+ twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
+ 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 19:53 +0000 [r251995] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Forward declaring dahdi_pri was already
+ done.
+
+2010-03-12 19:49 +0000 [r251992] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010)
+ | 8 lines 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 19:42 +0000 [r251988] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 251987 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r251987 | rmudgett | 2010-03-12 13:40:16 -0600
+ (Fri, 12 Mar 2010) | 9 lines 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. ........ ................
+
+2010-03-11 21:08 +0000 [r251882-251885] Tilghman Lesher <tlesher at digium.com>
+
+ * /, apps/app_exec.c: Merged revisions 251884 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 |
+ tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines
+ 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 ........
+
+ * /, apps/app_system.c: Merged revisions 251877 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 |
+ tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines
+ 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 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.0.26 released
+
+2010-03-04 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.0.26-rc1 released
+
+2010-03-03 21:27 +0000 [r250612] Leif Madsen <lmadsen at digium.com>
+
+ * doc/tex/localchannel.tex: Merged revisions 250609 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r250609 | lmadsen | 2010-03-03 16:22:55 -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. (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 19:07 +0000 [r250482] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600
+ (Wed, 03 Mar 2010) | 22 lines 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 18:06 +0000 [r250265-250398] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 250395 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600
+ (Wed, 03 Mar 2010) | 22 lines 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 ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 |
+ dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines
+ fixes signed to unsigned int comparision issue for FaxMaxDatagram
+ value. ........
+
+2010-03-02 21:11 +0000 [r250040-250054] Leif Madsen <lmadsen at digium.com>
+
+ * doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010)
+ | 8 lines 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 250045 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500
+ (Tue, 02 Mar 2010) | 15 lines 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 ........ ................
+
+ * doc/tex/configuration.tex: Merged revisions 250037 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02 Mar 2010)
+ | 4 lines Update documentation to not imply we support overriding
+ options. (closes issue #16855) Reported by: davidw ........
+
+2010-03-02 19:45 +0000 [r249948] Alec L Davis <sivad.a at 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 19:20 +0000 [r249907] David Vossel <dvossel at digium.com>
+
+ * 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,
+ channels/chan_unistim.c, configs/console.conf.sample,
+ configs/chan_dahdi.conf.sample, channels/chan_local.c,
+ configs/oss.conf.sample, channels/chan_sip.c, /,
+ configs/usbradio.conf.sample, configs/misdn.conf.sample,
+ channels/chan_gtalk.c, channels/chan_console.c: Merged revisions
+ 249893 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 |
+ dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
+ 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 09:16 +0000 [r249846] Alec L Davis <sivad.a at 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 (closes issue #16880) Reported by:
+ alecdavis Patches: based on echo_exit_1-6-1.diff.txt uploaded by
[... 56702 lines stripped ...]
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