[svn-commits] bebuild: tag 10.8.0-rc1 r370611 - /tags/10.8.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 31 08:32:14 CDT 2012
Author: bebuild
Date: Tue Jul 31 08:32:10 2012
New Revision: 370611
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=370611
Log:
Importing files for 10.8.0-rc1 release.
Added:
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tags/10.8.0-rc1/.version (with props)
tags/10.8.0-rc1/ChangeLog (with props)
Added: tags/10.8.0-rc1/.lastclean
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+2012-07-31 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.8.0-rc1 Released.
+
+2012-07-30 16:49 +0000 [r370564] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c, /: Release B channel allocation on error
+ path in chan_misdn. ........ Merged revisions 370563 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-07-30 14:50 +0000 [r370547] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_meetme.c: app_meetme: Change app_meetme support level to
+ extended from deprecated (closes issue ASTERISK-20134) Reported
+ by: Leif Madsen
+
+2012-07-25 21:12 +0000 [r370495] Jonathan Rose <jrose at digium.com>
+
+ * /, res/res_agi.c: 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
+
+2012-07-24 17:08 +0000 [r370432] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * /, channels/chan_oss.c: chan_oss: fix "sample rate" error message
+ Merged revisions 370428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-07-24 16:54 +0000 [r370361-370430] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/frame.c, /: 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
+
+ * /, funcs/func_shell.c: Improve documentation for the SHELL()
+ dialplan function. ........ Merged revisions 370383 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/channel.c, /: 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 22:11 +0000 [r370277] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/cel.c: 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
+
+2012-07-19 22:01 +0000 [r370271-370274] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/cel.c: 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
+
+ * res/res_rtp_asterisk.c, /: 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
+
+2012-07-18 19:14 +0000 [r370184-370206] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, main/cel.c: 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
+
+ * main/channel.c, addons/app_mysql.c, main/pbx.c,
+ funcs/func_curl.c, /, main/ccss.c, funcs/func_odbc.c,
+ 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: 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 19:52 +0000 [r370132] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: 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
+
+2012-07-16 13:51 +0000 [r370082] Kinsey Moore <kmoore at digium.com>
+
+ * /, UPGRADE.txt, CHANGES, UPGRADE-1.8.txt: 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 20:20 +0000 [r370025] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /: 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
+
+2012-07-12 20:05 +0000 [r369994-370015] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: 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
+
+ * /, channels/chan_sip.c: 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
+
+2012-07-12 14:25 +0000 [r369971] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, funcs/func_math.c: 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
+
+2012-07-11 17:12 +0000 [r369938] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * funcs/func_realtime.c, /: 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 Released.
+
+2012-07-11 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.7.0-rc1 Released.
+
+2012-07-10 13:35 +0000 [r369871] Kinsey Moore <kmoore at digium.com>
+
+ * main/pbx.c, /, apps/app_stack.c: 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 17:06 +0000 [r369819] Jason Parker <jparker at digium.com>
+
+ * configs/sip_notify.conf.sample, /: 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
+
+2012-07-09 14:43 +0000 [r369793] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c: 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 21:02 +0000 [r369751] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c: 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
+
+2012-07-06 18:47 +0000 [r369709-369732] Mark Michelson <mmichelson at digium.com>
+
+ * main/tcptls.c, /: 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: 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
+
+2012-07-05 19:12 +0000 [r369653] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_voicemail.c, /: AST-2012-011: 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) Reported by: Dan Delaney Tested by:
+ Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by
+ kmoore (license 6273) ........ Merged revisions 369652 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-07-05 17:02 +0000 [r369627] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: 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 17:02 +0000 [r369558-369580] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: 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/chan_sip.c, channels/sip/include/sip.h: 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 20:28 +0000 [r369511] Mark Michelson <mmichelson at digium.com>
+
+ * main/rtp_engine.c: Fix apparent copy and paste error where
+ incorrect "glue" is used.
+
+2012-06-29 16:54 +0000 [r369472-369491] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: 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: 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 21:10 +0000 [r369437] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: AST-2012-010:
+ 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. Review:
+ https://reviewboard.asterisk.org/r/2009/ (closes issue
+ ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
+ Davies, Terry Wilson ........ Merged revisions 369436 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-26 13:22 +0000 [r369369-369391] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/adsi.c: 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, /: 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
+
+2012-06-25 19:16 +0000 [r369353] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: 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
+
+2012-06-25 15:59 +0000 [r369328] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/features.c: 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
+
+2012-06-25 15:52 +0000 [r369303-369325] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/adsi.h, /, main/Makefile, res/res_adsi.c,
+ main/adsi.c (added), res/res_adsi.exports.in (removed): 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
+
+ * /, channels/chan_sip.c: 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 00:12 +0000 [r369236-369283] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/features.c: 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: 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
+
+ * apps/app_dial.c, /: 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
+
+ * /, main/ccss.c: 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: 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
+
+2012-06-22 19:34 +0000 [r369215] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: 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. (closes issue
+ ASTERISK-20040) Reported by: Terry Wilson ........ Merged
+ revisions 369214 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-22 17:23 +0000 [r369206] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: 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 17:36 +0000 [r369147] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c, /: 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
+
+2012-06-20 02:04 +0000 [r369109] Michael L. Young <elgueromexicano at gmail.com>
+
+ * main/netsock2.c, /, include/asterisk/netsock2.h: 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
+
+2012-06-19 23:32 +0000 [r369091] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c: check rtptimeouts in ooh323 channels as per
+ config file (rtp voice, video, udptl except rtcp) (closes issue
+ ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
+ 19179-ooh323-ast10.patch
+
+2012-06-19 15:37 +0000 [r369067] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Fix request routing issue when
+ outboundproxy is used. Asterisk was incorrectly setting the
+ destination of CANCELs and ACKs for error responses to the URI of
+ the initial INVITE. This resulted in further requests, such as
+ INVITEs with authentication credentials, to be routed
+ incorrectly. Instead, when these CANCEL or ACKs are to be sent,
+ we should simply keep the destination the same as what it
+ previously was. There is no need to alter it any. (closes issue
+ ASTERISK-20008) Reported by Marcus Hunger Patches:
+ ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
+ ........ Merged revisions 369066 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-18 18:11 +0000 [r369044] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/features.c: Fix monitoring calls put in a parking lot. *
+ Fix a regression that was introduced by -r366167 which
+ effectively disabled monitoring parked calls. (closes issue
+ ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett
+ ........ Merged revisions 369043 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-15 16:07 +0000 [r369005] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/sip/sdp_crypto.c, main/slinfactory.c, main/translate.c,
+ main/jitterbuf.c, main/acl.c, channels/iax2-provision.c,
+ channels/sig_pri.c, utils/astdb2bdb.c, main/chanvars.c,
+ main/data.c, main/hashtab.c, channels/chan_misdn.c,
+ main/abstract_jb.c, main/fixedjitterbuf.c,
+ channels/sip/dialplan_functions.c, main/test.c, res/snmp/agent.c,
+ main/event.c, main/astmm.c, channels/sip/config_parser.c,
+ channels/vgrabbers.c, main/alaw.c, main/asterisk.c, main/dsp.c,
+ main/timing.c, main/udptl.c, main/autoservice.c,
+ main/fskmodem_float.c, main/frame.c, main/security_events.c,
+ main/ccss.c, main/threadstorage.c, main/say.c,
+ channels/console_video.c, channels/sip/reqresp_parser.c,
+ main/devicestate.c, main/astfd.c, main/ssl.c,
+ main/taskprocessor.c, main/autochan.c, channels/misdn/isdn_lib.c,
+ main/enum.c, main/format_pref.c, main/astobj2.c,
+ main/indications.c, main/fskmodem.c, channels/misdn_config.c,
+ apps/confbridge/conf_config_parser.c, main/io.c, main/cli.c,
+ main/ulaw.c, main/dial.c, main/framehook.c, main/format_cap.c,
+ main/strcompat.c, main/heap.c, channels/misdn/ie.c, main/plc.c,
+ main/logger.c, main/stdtime/localtime.c, channels/sig_ss7.c,
+ main/sched.c, main/datastore.c, main/lock.c, main/strings.c,
+ main/pbx.c, main/stun.c, channels/sip/srtp.c, main/dnsmgr.c,
+ channels/vcodecs.c, channels/sip/security_events.c,
+ utils/astdb2sqlite3.c, main/aoc.c, pbx/dundi-parser.c,
+ main/cel.c, channels/iax2-parser.c,
+ build_tools/find_missing_support_level (added), main/netsock.c,
+ main/tcptls.c, main/srv.c, main/privacy.c, main/callerid.c,
+ main/file.c, channels/misdn/portinfo.c, main/audiohook.c,
+ main/xmldoc.c, main/netsock2.c, main/format.c,
+ main/global_datastores.c, main/rtp_engine.c, /, res/ais/clm.c,
+ main/utils.c, channels/misdn/isdn_msg_parser.c, main/xml.c,
+ main/config.c, main/loader.c, main/term.c, main/channel.c,
+ main/cdr.c, res/ael/pval.c, channels/sig_analog.c, main/tdd.c,
+ channels/console_gui.c, res/ais/evt.c, main/fskmodem_int.c,
+ channels/console_board.c, main/syslog.c, main/app.c,
+ main/image.c, main/dns.c, main/message.c, main/db.c,
+ main/bridging.c: Multiple revisions 369001-369002 ........
+ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun
+ 2012) | 11 lines Add support-level indications to many more
+ source files. Since we now have tools that scan through the
+ source tree looking for files with specific support levels, we
+ need to ensure that every file that is a component of a 'core' or
+ 'extended' module (or the main Asterisk binary) is explicitly
+ marked with its support level. This patch adds support-level
+ indications to many more source files in tree, but avoids adding
+ them to third-party libraries that are included in the tree and
+ to source files that don't end up involved in Asterisk itself.
+ ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15
+ Jun 2012) | 3 lines Add a script to enable finding source files
+ without support-levels defined. ........ Merged revisions
+ 369001-369002 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-14 17:31 +0000 [r368947] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_skinny.c: AST-2012-009: Fix crash in chan_skinny
+ due to Key Pad Button Message handling AST-2012-008 (r367844)
+ fixed a denial of service attack exploitable in the Skinny
+ channel driver that occurred when certain messages are sent after
+ a previously registered station sends an Off Hook message.
+ Unresolved in that patch is an issue in the Asterisk 10 releases,
+ wherein, if a Station Key Pad Button Message is processed after
+ an Off Hook message, the channel driver will inappropriately
+ dereference a NULL pointer. This patch fixes those places where
+ the message handling or the channel callback functions would
+ attempt to dereference the line's pointer to the device. (issue
+ ASTERISK-19905) Reported by: Christoph Hebeisen Tested by:
+ mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff
+ uploaded by mjordan (license 6283)
+
+2012-06-14 15:25 +0000 [r368899-368928] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/Makefile: Revert Makefile change to remove embedding
+ res_adsi.so The change has resulted in a linking error for
+ certain versions of GCC. This is much worse than the original
+ issue, so for now, temporarily revert the change. A more thorough
+ change will be sought out. ........ Merged revisions 368927 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, funcs/func_volume.c: Fix a deadlock that occurs when
+ func_volume is used on a local channel. This was discovered by
+ trying to perform a call forward to an extension that makes use
+ of func_volume. When the local channel is optimized away, the
+ datastore on the local;2 channel would have its audiohook
+ destroyed rather than detaching the audiohook from the channel
+ and then destroying it. With this patch, func_volume's datastore
+ destructor takes the proper route of detaching the audiohook and
+ then destroying it. (closes issue ASTERISK-19611) reported by
+ Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark
+ Michelson (license #5049) ........ Merged revisions 368898 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-13 20:27 +0000 [r368895] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_smdi.c, /, res/res_adsi.c: Mark res_smdi/res_adsi as
+ 'core' supported modules Recently, various issues surrounding
+ weak symbols have caused problems with modules that rely on that
+ feature to be enabled in menuselect. This includes app_voicemail
+ and chan_dahdi, as they both rely upon res_smdi and res_adsi,
+ which, in certain circumstances, may not be enabled by default in
+ menuselect. Because res_smdi/res_adsi are dependencies for
+ chan_dahdi/app_voicemail, this patch marks both as 'core'
+ supported modules. This will allow both app_voicemail and
+ chan_dahdi to be enabled as well, regardless of whether or not
+ that system supports weak symbols. (issue AST-900) Reported by:
+ Thomas Arimont (issue AST-885) Reported by: Denis Alberto
+ Martinez ........ Merged revisions 368894 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-13 19:36 +0000 [r368885] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+
+ the result is that Asterisk has a phantom module loaded at
+ startup, claiming to be res_adsi. (closes issue ASTERISK-19920)
+ reported by Leif Madsen ........ Merged revisions 368873 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-06-13 14:30 +0000 [r368831-368853] Matthew Jordan <mjordan at digium.com>
+
+ * Makefile, /: Do not install empty directories; add ASTLIBDIR
+ r368830 modified the installation script to only create a
+ directory if that directory does not exist. If some directory
+ variable was empty, it would attempt to create the empty
+ location. It also failed to create the ASTLIBDIR directory. This
+ patch fixes it such that the correct directories are made and
+ only created if a value specifying them actually exists. ........
+ Merged revisions 368852 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * Makefile, /: Do not perform install on existing directories If a
+ directory already exists, performing a 'make install' will remove
+ the permissions associated with the current directory and replace
+ them with the permissions of the user executing the install. This
+ patch changes this behavior to only perform an install on the
+ directory if the directory does not exist. Thus, if a user later
+ changes the permissions on that directory, those permissions will
+ be preserved in subsequent installs. Review:
+ https://reviewboard.asterisk.org/r/1986 Review:
+ https://reviewboard.asterisk.org/r/1864 (closes issue
+ ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger,
[... 25191 lines stripped ...]
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