[svn-commits] bebuild: tag 10.8.0-digiumphones-rc1 r370615 - /tags/10.8.0-digiumphones-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 31 08:35:23 CDT 2012
Author: bebuild
Date: Tue Jul 31 08:35:19 2012
New Revision: 370615
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=370615
Log:
Importing files for 10.8.0-digiumphones-rc1 release.
Added:
tags/10.8.0-digiumphones-rc1/.lastclean (with props)
tags/10.8.0-digiumphones-rc1/.version (with props)
tags/10.8.0-digiumphones-rc1/ChangeLog (with props)
Added: tags/10.8.0-digiumphones-rc1/.lastclean
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--- tags/10.8.0-digiumphones-rc1/ChangeLog (added)
+++ tags/10.8.0-digiumphones-rc1/ChangeLog Tue Jul 31 08:35:19 2012
@@ -1,0 +1,26671 @@
+2012-07-31 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.8.0-digiumphones-rc1 Released.
+
+2012-07-30 17:24 +0000 [r370555-370584] Automerge script <automerge at asterisk.org>
+
+ * channels/chan_misdn.c, /: Merged revisions 370564 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370564 | rmudgett | 2012-07-30 11:49:12 -0500
+ (Mon, 30 Jul 2012) | 5 lines Release B channel allocation on
+ error path in chan_misdn. ........ Merged revisions 370563 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, apps/app_meetme.c: Merged revisions 370547 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r370547 | jrose | 2012-07-30 09:50:34 -0500 (Mon, 30 Jul 2012) |
+ 5 lines app_meetme: Change app_meetme support level to extended
+ from deprecated (closes issue ASTERISK-20134) Reported by: Leif
+ Madsen ........
+
+2012-07-25 21:22 +0000 [r370509] Automerge script <automerge at asterisk.org>
+
+ * res/res_agi.c, /: Merged revisions 370495 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370495 | jrose | 2012-07-25 16:12:50 -0500
+ (Wed, 25 Jul 2012) | 14 lines res_agi: Add message indicating
+ need for \n character in verbose message The while loop
+ responsible for reading AGI messages from a fastAGI service can
+ end up looping indefinitely when an AGI script fails to indicate
+ the end of a message with a \n character. This patch adds an
+ indication that we are expecting a \n character to end the
+ message to make it more clear to users that this is necessary if
+ they are receiving this warning over and over. (issue
+ ASTERISK-20061) Reported by: Eike Kuiper ........ Merged
+ revisions 370494 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-25 03:45 +0000 [r370473] Terry Wilson <twilson at digium.com>
+
+ * main/pbx.c, /: Revert a change that broke compilation 1) There is
+ no such function as ast_ref() 2) The patch was originally
+ credited as the one uploaded by Guenther Kelleter (license 6372)
+ via issue AST-921, but the patch committed was not the patch
+ referenced on the issue. 3) Guenther Kelleter's patch was
+ actually correct. It moved the ast_free above the
+ presencechange_cleanup label. I am not committing his change as
+ it is not technically necesary--calling ast_free(NULL) is
+ perfectly safe and I worry that moving the ast_free outside of
+ the label could lead to future bugs if someone ever adds another
+ failure conditional and expects 'goto presencechange_cleanup;' to
+ clean up after everything.
+
+2012-07-24 21:08 +0000 [r370465] Jonathan Rose <jrose at digium.com>
+
+ * main/pbx.c, /: Don't attempt free of NULL ptr in pbx.c
+ handle_presencechange (closes issue AST-921) Reported by:
+ Guenther Kelleter Patches: nullptr.patch uploaded by Guenther
+ Kelleter (license 6372)
+
+2012-07-24 17:24 +0000 [r370381-370452] Automerge script <automerge at asterisk.org>
+
+ * channels/chan_oss.c, main/frame.c, /: Merged revisions
+ 370430,370432 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370430 | kpfleming | 2012-07-24 11:54:01 -0500
+ (Tue, 24 Jul 2012) | 5 lines Rewrite a comment that didn't
+ adequately explain the code it was documenting. ........ Merged
+ revisions 370429 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r370432 | tzafrir | 2012-07-24 12:08:40 -0500
+ (Tue, 24 Jul 2012) | 4 lines chan_oss: fix "sample rate" error
+ message Merged revisions 370428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, funcs/func_shell.c: Merged revisions 370384 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370384 | kpfleming | 2012-07-23 16:09:53 -0500
+ (Mon, 23 Jul 2012) | 5 lines Improve documentation for the
+ SHELL() dialplan function. ........ Merged revisions 370383 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/channel.c, /: Merged revisions 370361 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370361 | kpfleming | 2012-07-23 09:51:21 -0500
+ (Mon, 23 Jul 2012) | 13 lines Free any datastores attached to
+ dummy channels. Revision 370205 added the use of a datastore
+ attached to a dummy channel to resolve a memory leak, but
+ ast_dummy_channel_destructor() in this branch did not free
+ datastores, resulting in a continued (but slightly smaller)
+ memory leak. This patch backports the change to free said
+ datastores from the Asterisk trunk. (related to issue AST-916)
+ ........ Merged revisions 370360 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-19 22:24 +0000 [r370297] Automerge script <automerge at asterisk.org>
+
+ * main/cel.c, res/res_rtp_asterisk.c, /: Merged revisions
+ 370271,370274,370277 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370271 | mjordan | 2012-07-19 16:37:09 -0500
+ (Thu, 19 Jul 2012) | 49 lines Handle extremely out of order RFC
+ 2833 DTMF The current implementation of RFC 2833 DTMF handling in
+ res_rtp_asterisk will, if a packet arrives out of order, drop the
+ packet. This is to prevent duplicate ton generation in the
+ Asterisk core. Since the RTP layer does not buffer data itself,
+ this is the only option the RTP layer currently has for handling
+ packets that arrive out of order. For the most part, this doesn't
+ matter. For a particular digit, so long as a BEGIN packet arrives
+ before the first END packet, the digit will be produced. If
+ subsequent BEGIN packets arrive interleaved with the ENDs, they
+ will be dropped; likewise, if the BEGIN or END packets themselves
+ are out of order, those packets are dropped but sufficient
+ information is conveyed to the Asterisk core to produce the
+ appropriate digit. For certain sequences of DTMF packets - most
+ notably when, for a particular digit, an END packet arrives
+ before any BEGIN packet for that digit - this is a real problem.
+ When an END arrives before any BEGINs, the END packet is dropped
+ - but at the same time, it causes subsequent BEGIN packets for
+ that digit to be ignored. When the next in order END packet
+ arrives, it too is dropped - Asterisk believes that there was no
+ initial BEGIN. The solution this patch provides is to trust the
+ END packet to convey the information needed for the Asterisk core
+ to produce the DTMF digit. If we receive an END packet, and it: *
+ Has a timestamp greater then the last timestamp received from an
+ END packet * Does not have the same sequence number as the last
+ received sequence number (and is thus not an END packet
+ retransmission) Then we send the END frame up to the Asterisk
+ core. It contains enough DTMF information for Asterisk to produce
+ the digit. On the other hand, if we receive a BEGIN or
+ continuation packet that occurs with a timestamp equal to or less
+ then the last END timestamp, then we've received something out of
+ order - but we already have received enough information to
+ produce the digit. These packets are dropped. Much thanks goes to
+ Olle Johansson (oej) for providing the idea for this solution.
+ Review: https://reviewboard.asterisk.org/r/2033/ (issue
+ ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt
+ Jordan ........ Merged revisions 370252 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r370274 | mjordan | 2012-07-19 17:01:32 -0500
+ (Thu, 19 Jul 2012) | 17 lines Fix compilation error when
+ MALLOC_DEBUG is enabled To fix a memory leak in CEL, a channel
+ datastore was introduced whose destruction function pointer was
+ pointed to the ast_free macro. Without MALLOC_DEBUG enabled this
+ compiles as fine, as ast_free is defined as free. With
+ MALLOC_DEBUG enabled, however, ast_free takes on a definition
+ from a different place then utils.h, and became undefined. This
+ patch resolves this by using a reference to ast_free_ptr. When
+ MALLOC_DEBUG is enabled, this calls ast_free; when MALLOC_DEBUG
+ is not enabled, this is defined to be ast_free, which is defined
+ to be free. (issue AST-916) Reported by: Thomas Arimont ........
+ Merged revisions 370273 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r370277 | rmudgett | 2012-07-19 17:11:48 -0500
+ (Thu, 19 Jul 2012) | 7 lines Fix compiler warnings. gcc (GCC)
+ 4.2.4 has problems casting away constness. ........ Merged
+ revisions 370275 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-18 19:23 +0000 [r370202-370224] Automerge script <automerge at asterisk.org>
+
+ * main/cel.c, /: Merged revisions 370206 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370206 | kpfleming | 2012-07-18 14:14:09 -0500
+ (Wed, 18 Jul 2012) | 19 lines Resolve severe memory leak in CEL
+ logging modules. A customer reported a significant memory leak
+ using Asterisk 1.8. They have tracked it down to
+ ast_cel_fabricate_channel_from_event() in main/cel.c, which is
+ called by both in-tree CEL logging modules (cel_custom.c and
+ cel_sqlite3_custom.c) for each and every CEL event that they log.
+ The cause was an incorrect assumption about how data attached to
+ an ast_channel would be handled when the channel is destroyed;
+ the data is now stored in a datastore attached to the channel,
+ which is destroyed along with the channel at the proper time.
+ (closes issue AST-916) Review:
+ https://reviewboard.asterisk.org/r/2053/ ........ Merged
+ revisions 370205 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * funcs/func_lock.c, apps/app_macro.c, channels/chan_iax2.c,
+ apps/app_mixmonitor.c, apps/app_stack.c, funcs/func_global.c,
+ res/res_odbc.c, main/channel.c, addons/app_mysql.c, main/pbx.c,
+ funcs/func_curl.c, /, main/ccss.c, funcs/func_odbc.c: Merged
+ revisions 370184 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370184 | kpfleming | 2012-07-18 12:13:07 -0500
+ (Wed, 18 Jul 2012) | 10 lines Ensure that all ast_datastore_info
+ structures are 'const'. While addressing a bug, I came across a
+ instance of 'struct ast_datastore_info' that was not declared
+ 'const'. Since the API already expects them to be 'const', this
+ patch changes the declarations of all existing instances that
+ were not already declared that way. ........ Merged revisions
+ 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-16 20:24 +0000 [r370101-370151] Automerge script <automerge at asterisk.org>
+
+ * /, channels/chan_sip.c: Merged revisions 370132 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370132 | wdoekes | 2012-07-16 14:52:45 -0500
+ (Mon, 16 Jul 2012) | 11 lines Code cleanup and bugfix in chan_sip
+ outboundproxy parsing. The bug was clearing the global
+ outboundproxy when a peer-specific outboundproxy was bad. The
+ cleanup reduces duplicate code. Review:
+ https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark
+ Michelson ........ Merged revisions 370131 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * UPGRADE.txt, CHANGES, UPGRADE-1.8.txt, /: Merged revisions 370082
+ via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370082 | kmoore | 2012-07-16 08:51:57 -0500
+ (Mon, 16 Jul 2012) | 8 lines Add comments about the BUILD_NATIVE
+ change This is a significant change and mention of it should have
+ gone into UPGRADE.txt and CHANGES. ........ Merged revisions
+ 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-12 20:24 +0000 [r369958-370036] Automerge script <automerge at asterisk.org>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/chan_sip.c: Merged revisions 370015,370025 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r370015 | kmoore | 2012-07-12 15:05:45 -0500
+ (Thu, 12 Jul 2012) | 11 lines Include Expires header for SIP
+ PUBLISH requests RFC3903 requres SIP PUBLISH requests to have
+ Expires headers, so add them. Review:
+ https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth
+ ........ Merged revisions 370014 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r370025 | rmudgett | 2012-07-12 15:20:02 -0500
+ (Thu, 12 Jul 2012) | 8 lines Add missing ast_hangup() calls on
+ some analog exception paths. Make starting analog_ss_thread() or
+ __analog_ss_thread() failure paths hangup the channel. ........
+ Merged revisions 370017 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369994 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369994 | kmoore | 2012-07-12 13:55:17 -0500
+ (Thu, 12 Jul 2012) | 12 lines Prevent double uri_escaping in
+ chan_sip when pedantic is enabled If pedantic mode is enabled,
+ outbound invites will have double-escaped contacts. This avoids
+ setting an already-escaped string into a field where it is
+ expected to be unescaped. (closes issue ASTERISK-20023)
+ Reported-by: Walter Doekes ........ Merged revisions 369993 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * funcs/func_math.c, /: Merged revisions 369971 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369971 | elguero | 2012-07-12 09:25:45 -0500
+ (Thu, 12 Jul 2012) | 14 lines Correct Documentation For DEC
+ Function The documentation for DEC in func_math.c was incorrect.
+ Looks like a copy and paste error. (Closes issue ASTERISK-20095)
+ Reported by: Billy Chia Tested by: Michael L. Young Patches:
+ func_math.patch uploaded by Billy Chia (license 6381) ........
+ Merged revisions 369970 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * funcs/func_realtime.c, /: Merged revisions 369938 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369938 | tilghman | 2012-07-11 12:12:28 -0500
+ (Wed, 11 Jul 2012) | 11 lines Allow the REALTIME() function to
+ report errors back to the caller. Also, do more error checking on
+ the arguments specified to the REALTIME() function and clarify
+ the documentation. While I was editing the file, a few coding
+ guidelines fixups, as well. Review:
+ https://reviewboard.asterisk.org/r/2031/ ........ Merged
+ revisions 369937 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-30 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.7.0-digiumphones Released.
+
+2012-07-11 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.7.0-digiumphones-rc1 Released.
+
+2012-07-10 14:22 +0000 [r369889] Automerge script <automerge at asterisk.org>
+
+ * apps/app_stack.c, main/pbx.c, /: Merged revisions 369871 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369871 | kmoore | 2012-07-10 08:35:30 -0500
+ (Tue, 10 Jul 2012) | 12 lines Improve Goto and GotoIf related
+ documentation Correct documentation on labeliftrue and
+ labeliffalse parameters of GotoIf() and update several other
+ locations that use the same syntax. (closes issue ASTERISK-20007)
+ Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged
+ revisions 369869 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-09 19:51 +0000 [r369846] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Add support for exposing the received
+ contact URI and also for setting the request URI in messages.
+ (closes issue AST-911)
+
+2012-07-09 17:22 +0000 [r369810-369836] Automerge script <automerge at asterisk.org>
+
+ * configs/sip_notify.conf.sample, /: Merged revisions 369819 via
+ svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369819 | qwell | 2012-07-09 12:06:40 -0500
+ (Mon, 09 Jul 2012) | 9 lines Add Digium phones context to
+ sip_notify sample config. This makes it so that they can be
+ reconfigured remotely. (closes issue ASTERISK-19910) ........
+ Merged revisions 369818 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369793 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369793 | jrose | 2012-07-09 09:43:49 -0500
+ (Mon, 09 Jul 2012) | 9 lines chan_sip: Fix small behavioral
+ change accidentally introduced in r369750 When removing the
+ warning for AST_CONTROL_FLASH from sip_indicate, I also
+ inadvertently changed the return value, which would likely make
+ the indication not be sent in audio. This fixes that while still
+ removing the warning message. ........ Merged revisions 369792
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-06 21:21 +0000 [r369643-369763] Automerge script <automerge at asterisk.org>
+
+ * /, channels/chan_sip.c: Merged revisions 369751 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369751 | jrose | 2012-07-06 16:02:37 -0500
+ (Fri, 06 Jul 2012) | 12 lines chan_sip: Add case for FLASH
+ control frames so that we don't display a warning. chan_sip
+ channels can receive flash control frames when connected to
+ analog phones and possibly for other reasons. There really isn't
+ a reason to warn when these frames are received, we can safely
+ ignore them. Patches: dahdi_sip_flash.diff uploaded by Jonathan
+ Rose (license 6182) ........ Merged revisions 369750 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/tcptls.c, /: Merged revisions 369732 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369732 | mmichelson | 2012-07-06 13:47:05 -0500
+ (Fri, 06 Jul 2012) | 21 lines Remove a superfluous and dangerous
+ freeing of an SSL_CTX. The problem here is that multiple server
+ sessions share a SSL_CTX. When one session ended, the SSL_CTX
+ would be freed and set NULL, leaving the other sessions unable to
+ function. The code being removed is superfluous because the
+ SSL_CTX structures for servers will be properly freed when
+ ast_ssl_teardown is called. (closes issue ASTERISK-20074)
+ Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded
+ by Mark Michelson (license #5049) Testers: Trevor Helmsley
+ ........ Merged revisions 369731 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/bridging.c, /: Merged revisions 369709 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369709 | mmichelson | 2012-07-06 10:23:28 -0500
+ (Fri, 06 Jul 2012) | 14 lines Fix bridging thread leak. The
+ bridge thread was exiting but was never being reaped using
+ pthread_join(). This has been fixed now by calling pthread_join()
+ in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported
+ by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012
+ ........ Merged revisions 369708 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 369653 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369653 | kmoore | 2012-07-05 14:12:33 -0500
+ (Thu, 05 Jul 2012) | 20 lines Resolve heap corruption issue with
+ voicemail The heard and deleted arrays in the voicemail state
+ structure were not handled properly following the memory leak fix
+ in r354890 and a fix for an invalid free in r356797. This could
+ result in accessing and writing into freed memory. The allocation
+ for these arrays has been reworked to avoid the possibility of
+ invalid frees, access of freed memory, and crashes that were
+ occurring as a result of this. Locking around accesses and
+ modifications of the voicemail state structure members
+ dh_arraysize, heard, and deleted has been added to prevent
+ simultaneous modification and access when IMAP storage is in use.
+ If IMAP storage is not in use, this locking is not compiled in.
+ Review: https://reviewboard.asterisk.org/r/1994/ (closes issue
+ ASTERISK-19923) ........ Merged revisions 369652 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369627 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369627 | mjordan | 2012-07-05 12:02:53 -0500
+ (Thu, 05 Jul 2012) | 18 lines Do not send a BYE when a
+ provisional response arrives during a re-INVITE Commits r369557
+ and r369579 were done to improve handling of re-INVITEs when the
+ UA that was supposed to receive the re-INVITE fails to respond. A
+ limitation of those patches occurred when a UA sent a provisional
+ response to the re-INVITE. This triggered a sending of a BYE in
+ check_pending. This patch tweaks the handling of the re-INVITE
+ such that a BYE is not sent in response to those messages. (issue
+ ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
+ patches: (reinvite_tweak.diff license #5012 by Steve Davies)
+ ........ Merged revisions 369626 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-07-03 17:23 +0000 [r369578-369598] Automerge script <automerge at asterisk.org>
+
+ * /, channels/chan_sip.c: Merged revisions 369580 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369580 | twilson | 2012-07-03 12:02:18 -0500
+ (Tue, 03 Jul 2012) | 11 lines More improvements to re-INVITEs
+ timing out after a provisional response There is no need to call
+ check_pendings() on a final response to an INVITE when destroying
+ the scheduler entry as it will be done later during normal
+ processing. (issue ASTERISK-19992) ........ Merged revisions
+ 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 369558 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369558 | twilson | 2012-07-03 09:34:22 -0500
+ (Tue, 03 Jul 2012) | 14 lines Better handle re-INVITEs with
+ provisional but no final repsonses A previous attempt at fixing
+ this issue had negative side effects related to attended
+ transfers which this patch should resolve. Many thanks to Steve
+ Davies for all of the good suggestions and testing. (closes issue
+ ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
+ Davies, Terry Wilson Review:
+ https://reviewboard.asterisk.org/r/2009/ ........ Merged
+ revisions 369557 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-29 21:19 +0000 [r369488-369516] Automerge script <automerge at asterisk.org>
+
+ * main/rtp_engine.c, /: Merged revisions 369511 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10 ........
+ r369511 | mmichelson | 2012-06-29 15:28:10 -0500 (Fri, 29 Jun
+ 2012) | 3 lines Fix apparent copy and paste error where incorrect
+ "glue" is used. ........
+
+ * /, channels/chan_sip.c: Merged revisions 369491 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369491 | file | 2012-06-29 11:54:11 -0500 (Fri,
+ 29 Jun 2012) | 5 lines With some configurations a transport is
+ not actually specified so assume UDP in these cases. ........
+ Merged revisions 369490 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369472 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369472 | file | 2012-06-29 10:30:47 -0500 (Fri,
+ 29 Jun 2012) | 10 lines Make the address family filter specific
+ to the transport. (closes issue ASTERISK-16618) Reported by: Leif
+ Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........
+ Merged revisions 369471 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-27 21:22 +0000 [r369453] Automerge script <automerge at asterisk.org>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 369437 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369437 | twilson | 2012-06-27 16:10:01 -0500
+ (Wed, 27 Jun 2012) | 16 lines Clean up after a reinvite that
+ never gets a final response The basic problem is that if a
+ re-INVITE is sent by Asterisk and it receives a provisional
+ response, but no final response, then the dialog is never torn
+ down. In addition to leaking memory, this also leaks file
+ descriptors and will eventually lead to Asterisk no longer being
+ able to process calls. This patch just keeps track of whether
+ there is an outstanding re-INVITE, and if there is goes ahead and
+ cleans up everything as though there was no outstanding reinvite.
+ (closes issue ASTERISK-19992) ........ Merged revisions 369436
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-26 14:21 +0000 [r369322-369406] Automerge script <automerge at asterisk.org>
+
+ * main/adsi.c, /: Merged revisions 369391 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369391 | mjordan | 2012-06-26 08:22:42 -0500
+ (Tue, 26 Jun 2012) | 15 lines Fix crash in unloading of res_adsi
+ module When res_adsi is unloaded, it removes the ADSI functions
+ that it previously installed by passing a NULL adsi_funcs pointer
+ to ast_adsi_install_funcs. This function was not checking whether
+ or not the adsi_funcs pointer passed in was NULL before
+ dereferencing it to check whether or not the version of the
+ functions matches what the core was expecting it. This patch
+ makes it so that the version is only checked if a potentially
+ valid adsi_funcs pointer was passed in. Passing in NULL removes
+ the installed functions, bypassing the version check. ........
+ Merged revisions 369390 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/cdr.c, /: Merged revisions 369369 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369369 | mjordan | 2012-06-25 14:36:02 -0500
+ (Mon, 25 Jun 2012) | 29 lines Fix incorrect duration reporting in
+ CDRs created in batch mode Certain places in core/cdr.c would, if
+ the duration value were 0, calculate the duration as being the
+ delta between the current time and the time at which the CDR
+ record was started. While this does not typically cause a problem
+ in non-batch mode, this can cause an issue in batch mode where
+ CDR records are gathered and written long after those calls have
+ ended. In particular, this affects calls that were never
+ answered, as those are expected to have a duration of 0. Often,
+ this would result in CDR logs with a significant number of calls
+ with lengthy durations, but dispositions of "BUSY". Note that
+ this does not affect cdr_csv, as that backend does not use
+ ast_cdr_getvar and instead directly reports the duration value.
+ The affected core backends include cdr_apative_odbc and
+ cdr_custom; other extended or deprecated CDR backends may
+ potentially still directly manipulate the duration values. (issue
+ ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883)
+ Reported by: Thomas Arimont Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1996/ ........ Merged
+ revisions 369351 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 369353 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369353 | mmichelson | 2012-06-25 14:16:52 -0500
+ (Mon, 25 Jun 2012) | 14 lines Re-fix how local tag is generated
+ when sending a 481 to an INVITE. Match our local tag to whatever
+ to-tag was sent in the initial INVITE. Because the size of the
+ to-tag may not fit in the buffer in the sip_pvt, it has been
+ changed to a string field. (closes issue ASTERISK-19892) reported
+ by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977
+ ........ Merged revisions 369352 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * main/features.c, res/res_adsi.c, main/adsi.c (added),
+ res/res_adsi.exports.in (removed), include/asterisk/adsi.h, /,
+ main/Makefile: Merged revisions 369325,369328 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369325 | mmichelson | 2012-06-25 10:52:42 -0500
+ (Mon, 25 Jun 2012) | 20 lines Multiple revisions 369323-369324
+ ........ r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon,
+ 25 Jun 2012) | 9 lines Eliminate embedding of res_adsi.so module.
+ The way this is done is to stop using the optional API. Instead,
+ res_adsi.so, when loaded fills in a table of function pointers.
+ Review: https://reviewboard.asterisk.org/r/1991 ........ r369324
+ | mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2
+ lines Forgot to svn add this file in my last commit. ........
+ Merged revisions 369323-369324 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r369328 | rmudgett | 2012-06-25 10:59:28 -0500
+ (Mon, 25 Jun 2012) | 15 lines Fix Bridge application occasionally
+ returning to the wrong location. * Fix do_bridge_masquerade()
+ getting the resume location from the zombie channel. The code
+ must not touch a clone channel after it has masqueraded it. The
+ clone channel has become a zombie and is starting to hangup.
+ (closes issue ASTERISK-19985) Reported by: jamicque Patches:
+ jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: jamicque ........ Merged revisions 369327
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369303 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369303 | mmichelson | 2012-06-25 09:23:16 -0500
+ (Mon, 25 Jun 2012) | 14 lines Be more consistent with the return
+ code for requests received from invalid domain. When Asterisk
+ receives an INVITE from an external domain when
+ allowexternaldomains=no send a 403 instead of a 404. This is
+ consistent with Asterisk's behavior when receiving a REGISTER in
+ this situation. (Closes issue ASTERISK-19601) Reported by Matthew
+ Jordan Patches: ASTERISK-19601-no401.patch uploaded by Mark
+ Michelson (License #5049) ........ Merged revisions 369302 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-23 00:20 +0000 [r369213-369294] Automerge script <automerge at asterisk.org>
+
+ * main/features.c, /: Merged revisions 369283 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369283 | rmudgett | 2012-06-22 19:12:27 -0500
+ (Fri, 22 Jun 2012) | 22 lines Fix Bridge application and AMI
+ Bridge action error handling. * Fix AMI Bridge action
+ disconnecting the AMI link on error. * Fix AMI Bridge action and
+ Bridge application not checking if their masquerades were
+ successful. * Fix Bridge application running the h-exten when it
+ should not. * Made do_bridge_masquerade() return if the
+ masquerade was successful so the Bridge application and AMI
+ Bridge action could deal with it correctly. * Made
+ bridge_call_thread_launch() hangup the passed in channels if the
+ bridge_call_thread fails to start. Those channels would have been
+ orphaned. * Made builtin_atxfer() check the success of the
+ transfer masquerade setup. ........ Merged revisions 369282 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * apps/app_queue.c, apps/app_dial.c, /: Merged revisions
+ 369259,369263 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369259 | rmudgett | 2012-06-22 16:37:05 -0500
+ (Fri, 22 Jun 2012) | 5 lines Check if PBX was started and fix F
+ and F(x) action logic in Dial application. ........ Merged
+ revisions 369258 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r369263 | rmudgett | 2012-06-22 17:09:29 -0500
+ (Fri, 22 Jun 2012) | 5 lines Explicitly check caller hangup in
+ app Queue rather than a polluted res2 value. ........ Merged
+ revisions 369262 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c, main/ccss.c: Merged revisions
+ 369236,369239 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369236 | rmudgett | 2012-06-22 15:49:33 -0500
+ (Fri, 22 Jun 2012) | 5 lines Change incorrect chan_sip zombie
+ hangup debug message. They are all zombies now. ........ Merged
+ revisions 369235 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r369239 | rmudgett | 2012-06-22 16:04:25 -0500
+ (Fri, 22 Jun 2012) | 5 lines Check if PBX was started for generic
+ CCSS recall. ........ Merged revisions 369238 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369215 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369215 | twilson | 2012-06-22 14:34:59 -0500
+ (Fri, 22 Jun 2012) | 9 lines Don't crash on a guest directmedia
+ call A sip_pvt may not have relatedpeer set if a call doesn't
+ match up with a peer. If there is no relatedpeer, there is no
+ direct media ACL to apply, so just return that it is allowed.
+ ........ Merged revisions 369214 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 369206 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369206 | kmoore | 2012-06-22 12:23:26 -0500
+ (Fri, 22 Jun 2012) | 11 lines Don't parse media stream state for
+ SIP video streams The sendonly/recvonly/sendrecv/inactive media
+ stream attributes were parsed for video, but nothing was ever
+ done with them. With this code removed, an UNSUPPORTED message is
+ produced when these attributes are used in conjunction with a
+ video stream which is the better behavior since they were never
+ really supported in the first place. ........ Merged revisions
+ 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+2012-06-20 18:22 +0000 [r369056-369164] Automerge script <automerge at asterisk.org>
+
+ * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c, /:
+ Merged revisions 369147 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369147 | may | 2012-06-20 12:36:27 -0500 (Wed,
+ 20 Jun 2012) | 10 lines fix locking issue on empty callList
+ (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches:
+ ASTERISK-18322-2.patch ........ Merged revisions 369146 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
+ * include/asterisk/netsock2.h, main/netsock2.c, /: Merged revisions
+ 369109 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369109 | elguero | 2012-06-19 21:04:58 -0500
+ (Tue, 19 Jun 2012) | 23 lines Fix NULL pointer segfault in
+ ast_sockaddr_parse() While working with ast_parse_arg() to
+ perform a validity check, a segfault occurred. The segfault
+ occurred due to passing a NULL pointer to ast_sockaddr_parse()
+ from ast_parse_arg(). According to the documentation in config.h,
+ "result pointer to the result. NULL is valid here, and can be
+ used to perform only the validity checks." This patch fixes the
+ segfault by checking for a NULL pointer. This patch also adds
+ documentation to netsock2.h about why it is necessary to check
+ for a NULL pointer. (Closes issue ASTERISK-20006) Reported by:
+ Michael L. Young Tested by: Michael L. Young Patches:
+ asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/1990/
+ ........ Merged revisions 369108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................
+
[... 25989 lines stripped ...]
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