[svn-commits] bebuild: tag 1.8.16.0-rc1 r370605 - /tags/1.8.16.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 31 08:25:18 CDT 2012
Author: bebuild
Date: Tue Jul 31 08:25:14 2012
New Revision: 370605
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=370605
Log:
Importing files for 1.8.16.0-rc1 release.
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tags/1.8.16.0-rc1/.version (with props)
tags/1.8.16.0-rc1/ChangeLog (with props)
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--- tags/1.8.16.0-rc1/ChangeLog (added)
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+2012-07-31 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.16.0-rc1 Released.
+
+2012-07-30 16:47 +0000 [r370563] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Release B channel allocation on error path
+ in chan_misdn.
+
+2012-07-25 21:00 +0000 [r370494] 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
+
+2012-07-24 16:53 +0000 [r370429] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/frame.c: Rewrite a comment that didn't adequately explain
+ the code it was documenting.
+
+2012-07-24 16:49 +0000 [r370428] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * channels/chan_oss.c: chan_oss: fix "sample rate" error message
+
+2012-07-23 21:09 +0000 [r370360-370383] Kevin P. Fleming <kpfleming at digium.com>
+
+ * funcs/func_shell.c: Improve documentation for the SHELL()
+ dialplan function.
+
+ * 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)
+
+2012-07-19 22:07 +0000 [r370275] Richard Mudgett <rmudgett at digium.com>
+
+ * main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has problems
+ casting away constness.
+
+2012-07-19 22:00 +0000 [r370252-370273] 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
+
+ * 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
+
+2012-07-18 19:12 +0000 [r370183-370205] 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/
+
+ * 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,
+ funcs/func_lock.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.
+
+2012-07-16 19:50 +0000 [r370131] 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
+
+2012-07-16 13:44 +0000 [r370081] Kinsey Moore <kmoore at digium.com>
+
+ * UPGRADE.txt, CHANGES: Add comments about the BUILD_NATIVE change
+ This is a significant change and mention of it should have gone
+ into UPGRADE.txt and CHANGES.
+
+2012-07-12 20:15 +0000 [r370017] 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.
+
+2012-07-12 20:05 +0000 [r369993-370014] 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
+
+ * 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
+
+2012-07-12 14:23 +0000 [r369970] 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)
+
+2012-07-11 17:08 +0000 [r369937] 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/
+
+2012-07-30 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.15.0 Released.
+
+2012-07-11 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.15.0-rc1 Released.
+
+2012-07-10 13:33 +0000 [r369869] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_stack.c, main/pbx.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
+
+2012-07-09 17:05 +0000 [r369818] 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)
+
+2012-07-09 14:38 +0000 [r369792] 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.
+
+2012-07-06 20:54 +0000 [r369750] 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)
+
+2012-07-06 18:40 +0000 [r369708-369731] 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
+
+ * 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
+
+2012-07-05 19:01 +0000 [r369652] 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)
+
+2012-07-05 17:01 +0000 [r369626] 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)
+
+2012-07-03 16:58 +0000 [r369557-369579] 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)
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: 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/
+
+2012-06-29 16:52 +0000 [r369471-369490] 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.
+
+ * 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/
+
+2012-06-27 20:58 +0000 [r369436] Terry Wilson <twilson at digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: 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
+
+2012-06-26 13:21 +0000 [r369366-369390] 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.
+
+ * main/cdr.c: Tweak CDR change in r369351 As Tilghman pointed out
+ on review 1996, the check to see if a CDR end time has been set
+ is sufficient to know whether or not the duration value can be
+ used. The check-in done for r369351 forgot to include this
+ change.
+
+2012-06-25 19:13 +0000 [r369352] Mark Michelson <mmichelson at digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: 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
+
+2012-06-25 19:12 +0000 [r369351] Matthew Jordan <mjordan at digium.com>
+
+ * 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/
+
+2012-06-25 15:57 +0000 [r369327] 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
+
+2012-06-25 15:50 +0000 [r369302-369324] Mark Michelson <mmichelson at digium.com>
+
+ * main/adsi.c (added): Forgot to svn add this file in my last
+ commit.
+
+ * res/res_adsi.exports.in (removed), include/asterisk/adsi.h,
+ main/Makefile, res/res_adsi.c: 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
+
+ * 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)
+
+2012-06-23 00:04 +0000 [r369235-369282] 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.
+
+ * apps/app_queue.c: Explicitly check caller hangup in app Queue
+ rather than a polluted res2 value.
+
+ * apps/app_dial.c: Check if PBX was started and fix F and F(x)
+ action logic in Dial application.
+
+ * main/ccss.c: Check if PBX was started for generic CCSS recall.
+
+ * channels/chan_sip.c: Change incorrect chan_sip zombie hangup
+ debug message. They are all zombies now.
+
+2012-06-22 19:28 +0000 [r369214] 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
+
+2012-06-22 17:14 +0000 [r369195] 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.
+
+2012-06-20 17:33 +0000 [r369130-369146] 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
+
+ * addons/chan_ooh323.c: fix compile error (1.8 don't have
+ ast_channel_name macro)
+
+2012-06-20 02:03 +0000 [r369108] Michael L. Young <elgueromexicano at gmail.com>
+
+ * include/asterisk/netsock2.h, main/netsock2.c: 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/
+
+2012-06-19 23:28 +0000 [r369090] 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-ooh323-2.patch
+
+2012-06-19 15:30 +0000 [r369066] 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)
+
+2012-06-18 18:07 +0000 [r369043] 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
+
+2012-06-15 15:57 +0000 [r369001-369002] Kevin P. Fleming <kpfleming at digium.com>
+
+ * build_tools/find_missing_support_level (added): Add a script to
+ enable finding source files without support-levels defined.
+
+ * main/devicestate.c, main/astfd.c, main/ssl.c,
+ main/taskprocessor.c, main/astobj2.c, main/indications.c,
+ main/config.c, main/loader.c, main/term.c, main/cli.c,
+ channels/sig_analog.c, main/framehook.c, main/strcompat.c,
+ main/plc.c, res/ais/evt.c, main/fskmodem_int.c, main/syslog.c,
+ main/stdtime/localtime.c, main/db.c, main/bridging.c,
+ channels/sig_ss7.c, main/datastore.c, main/sched.c,
+ channels/sip/sdp_crypto.c, main/pbx.c, main/strings.c,
+ channels/vcodecs.c, channels/iax2-provision.c, main/aoc.c,
+ pbx/dundi-parser.c, main/cel.c, channels/iax2-parser.c,
+ main/chanvars.c, main/netsock.c, main/data.c, main/srv.c,
+ channels/chan_misdn.c, main/privacy.c,
+ channels/sip/dialplan_functions.c, main/fixedjitterbuf.c,
+ main/test.c, main/audiohook.c, main/alaw.c, main/asterisk.c,
+ main/timing.c, main/global_datastores.c, main/fskmodem_float.c,
+ main/ccss.c, channels/sip/reqresp_parser.c,
+ channels/misdn/isdn_msg_parser.c, main/utils.c, main/xml.c,
+ main/autochan.c, main/enum.c, channels/misdn/isdn_lib.c,
+ main/fskmodem.c, channels/misdn_config.c, main/io.c,
+ res/ael/pval.c, main/channel.c, main/cdr.c, main/ulaw.c,
+ main/dial.c, main/tdd.c, main/heap.c, channels/console_gui.c,
+ channels/misdn/ie.c, main/logger.c, channels/console_board.c,
+ main/app.c, main/image.c, main/dns.c, main/lock.c, main/stun.c,
+ main/dnsmgr.c, channels/sip/srtp.c, main/translate.c,
+ main/slinfactory.c, main/jitterbuf.c, main/acl.c,
+ channels/sig_pri.c, main/tcptls.c, main/hashtab.c,
+ main/abstract_jb.c, main/callerid.c, main/file.c,
+ res/snmp/agent.c, main/astmm.c, channels/misdn/portinfo.c,
+ main/event.c, channels/sip/config_parser.c, channels/vgrabbers.c,
+ main/xmldoc.c, main/dsp.c, main/udptl.c, main/netsock2.c,
+ main/autoservice.c, main/rtp_engine.c, main/frame.c,
+ main/security_events.c, res/ais/clm.c, main/threadstorage.c,
+ main/say.c, channels/console_video.c: 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.
+
+2012-06-14 15:23 +0000 [r368898-368927] 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.
+
+ * 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)
+
+2012-06-13 20:26 +0000 [r368894] 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
+
+2012-06-13 19:00 +0000 [r368873] 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
+
+2012-06-13 14:27 +0000 [r368830-368852] 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.
+
+ * 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,
+ Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded
+ by mjordan)
+
+2012-06-12 15:36 +0000 [r368807] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Set the Caller ID "tag" on peers even if
+ remote party information is present. On incoming calls, we were
+ setting the cid_tag on the dialog only if there was no remote
+ party information (Remote-Party-ID or P-Asserted-Identity)
+ present. The Caller ID tag is an invented parameter, though, and
+ should be set no matter the circumstance. (closes issue
+ ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884)
+ Reported by Trey Blancher
+
+2012-06-11 17:03 +0000 [r368759] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/channel.h, channels/chan_iax2.c, main/channel.c,
+ channels/chan_dahdi.c, channels/sig_analog.c,
+ channels/chan_sip.c: Fix deadlock potential with
+ ast_set_hangupsource() calls. Calling ast_set_hangupsource() with
+ the channel lock held can result in a deadlock because the
+ function also locks the bridged channel. (issue ASTERISK-19537)
+ (closes issue AST-891) Reported by: Guenther Kelleter Tested by:
+ Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec
+ Davis
+
+2012-06-11 15:13 +0000 [r368719-368738] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_queue.c, main/loader.c, channels/chan_dahdi.c,
+ res/res_config_odbc.c, channels/sip/dialplan_functions.c,
+ pbx/pbx_config.c, apps/app_directory.c, res/res_odbc.c,
+ res/res_speech.c, apps/app_voicemail.c, main/udptl.c,
+ channels/sip/sdp_crypto.c, channels/chan_sip.c, res/res_fax.c,
+ main/say.c, funcs/func_strings.c, channels/sip/reqresp_parser.c:
+ Fix coverity UNUSED_VALUE findings in core support level files
+ Most of these were just saving returned values without using them
+ and in some cases the variable being saved to could be removed as
+ well. (issue ASTERISK-19672)
+
+ * main/md5.c: Fix compilation in dev-mode Backport a compilation
+ fix in md5.c from trunk that only showed up in dev-mode under
+ certain compiler versions.
+
+2012-07-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.14.0 Released.
+
+2012-07-06 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.14.0-rc2 Released.
+
+ * AST-2012-010: Possible Resource Leak on Uncompleted Re-INVITE
+ transactions
+
+ * AST-2012-011: Remote Crash Vulnerability in VoiceMail Application
+
+ * Fix 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 is is allowed.
+
+ (closes issue ASTERISK-20040)
+
+ * 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 esnt, 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)
+
+ * 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)
+
+2012-06-08 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.14.0-rc1 Released.
+
+2012-06-06 21:27 +0000 [r368644] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c: Fix POTS flash hook
+ to orignate a second call deadlock. A deadlock can occur when a
+ POTS phone tries to flash hook to originate a second call for
+ 3-way or transfer. If another process is scanning the channels
+ container when the POTS line flash hooks then a deadlock will
+ occur. * Release the channel and private locks when creating a
+ new channel as a result of a flash hook. (closes issue
+ ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett
+
+2012-06-06 19:13 +0000 [r368625] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix a specific scenario where ACKs are not
+ matched. If a dialog-starting INVITE contains a to-tag, then
+ Asterisk will respond with a 481. In this case, the resulting
+ incoming ACK would not be matched, so Asterisk would continue
+ retransmitting the 481 until the transaction times out. There
+ were two issues. Asterisk, upon creating a sip_pvt would generate
+ a local tag. However, when the time came to transmit the 481,
+ since there was a to-tag in the INVITE, Asterisk would place this
+ original to-tag in the 481 response. When the ACK came in,
+ Asterisk would attempt to match the to-tag in the ACK to the
+ generated local tag. Unfortunately, Asterisk never actually
+ transmitted a response with the generated local tag, so the
+ to-tag in the ACK would not match. The other problem was that
+ when the 481 was sent, nothing was set on the sip_pvt to indicate
+ what CSeq is expected in the ACK. To fix the first problem, we
+ zero out the to-tag seen in the incoming INVITE. This way,
+ Asterisk, when time to send a response, will send its generated
+ local tag instead. To fix the second problem, we set the
+ sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
+ 481. (closes issue ASTERISK-19892) Reported by Mark Michelson
+
+2012-06-06 17:20 +0000 [r368604] Matthew Jordan <mjordan at digium.com>
+
+ * build_tools/make_version: Add feature modifier to versions
+ produced from branches Certain branches, such as Certified
+ Asterisk, may have a modifier added to them that specifies the
+ features available in that branch. For branches, this modifier is
+ expected to be reflected in the location of the branch in
+ subversion. For example, a subversion of URL of
+ /certified/branches/1.8.11 would have a feature modifier of
+ 'certified'. This is slightly different then how features are
+ determined for tags, where the feature is part of the actual tag
+ name, e.g., "10.5.0-digiumphones". In keeping with the
+ nomenclature used for tags, the feature specifier for branches is
+ translated and placed after the revision numbers. For the example
+ given previously, this would result in a branch version of
+ "Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
+
+2012-06-06 16:07 +0000 [r368586] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Ensure overlapping hold flags do not
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