[svn-commits] bebuild: tag 10.7.0-digiumphones-rc1 r369933 - /tags/10.7.0-digiumphones-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Jul 11 10:53:34 CDT 2012
Author: bebuild
Date: Wed Jul 11 10:53:29 2012
New Revision: 369933
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369933
Log:
Importing files for 10.7.0-digiumphones-rc1 release.
Added:
tags/10.7.0-digiumphones-rc1/.lastclean (with props)
tags/10.7.0-digiumphones-rc1/.version (with props)
tags/10.7.0-digiumphones-rc1/ChangeLog (with props)
Added: tags/10.7.0-digiumphones-rc1/.lastclean
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URL: http://svnview.digium.com/svn/asterisk/tags/10.7.0-digiumphones-rc1/ChangeLog?view=auto&rev=369933
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--- tags/10.7.0-digiumphones-rc1/ChangeLog (added)
+++ tags/10.7.0-digiumphones-rc1/ChangeLog Wed Jul 11 10:53:29 2012
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+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
+ ................
+
+ * addons/chan_ooh323.c, /: Merged revisions 369091 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10 ........
+ r369091 | may | 2012-06-19 18:32:06 -0500 (Tue, 19 Jun 2012) | 9
+ lines 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 ........
+
+ * /, channels/chan_sip.c: Merged revisions 369067 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369067 | mmichelson | 2012-06-19 10:37:37 -0500
+ (Tue, 19 Jun 2012) | 17 lines 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
+ ................
+
+ * main/features.c, /: Merged revisions 369044 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369044 | rmudgett | 2012-06-18 13:11:30 -0500
+ (Mon, 18 Jun 2012) | 12 lines 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 16:30 +0000 [r369026] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.c, /: Fix voicemail API tests by using the
+ correct argument order for create/destroy. ........ Merged
+ revisions 369024 from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+2012-06-15 16:25 +0000 [r369023] Automerge script <automerge at asterisk.org>
+
+ * main/translate.c, channels/vcodecs.c,
+ channels/sip/security_events.c, main/jitterbuf.c,
+ main/autochan.c, pbx/dundi-parser.c, main/aoc.c, main/cel.c,
+ main/enum.c, channels/iax2-parser.c, main/fskmodem.c,
+ main/config.c, channels/misdn_config.c, main/netsock.c,
+ build_tools/find_missing_support_level (added), main/loader.c,
+ main/ulaw.c, main/dial.c, channels/sig_analog.c, main/srv.c,
+ main/heap.c, main/privacy.c, channels/misdn/ie.c, res/ais/evt.c,
+ main/syslog.c, res/snmp/agent.c, main/event.c, main/astmm.c,
+ channels/sip/config_parser.c, channels/vgrabbers.c, main/db.c,
+ main/udptl.c, main/lock.c, channels/sip/sdp_crypto.c,
+ main/stun.c, main/frame.c, channels/sip/srtp.c,
+ main/threadstorage.c, channels/console_video.c,
+ channels/iax2-provision.c, main/xml.c, main/astfd.c,
+ main/taskprocessor.c, utils/astdb2bdb.c,
+ apps/confbridge/conf_config_parser.c, main/channel.c, main/cdr.c,
+ res/ael/pval.c, channels/chan_misdn.c, main/framehook.c,
+ main/tdd.c, main/strcompat.c, channels/console_gui.c,
+ channels/sip/dialplan_functions.c, main/fixedjitterbuf.c,
+ main/callerid.c, main/file.c, main/app.c,
+ main/stdtime/localtime.c, main/dns.c, main/message.c,
+ main/datastore.c, main/sched.c, main/timing.c, main/netsock2.c,
+ main/fskmodem_float.c, /, main/slinfactory.c, main/acl.c,
+ channels/sip/reqresp_parser.c, channels/sig_pri.c,
+ channels/misdn/isdn_lib.c, main/term.c, main/io.c,
+ main/hashtab.c, main/format_cap.c, main/abstract_jb.c,
+ main/fskmodem_int.c, main/logger.c, main/audiohook.c,
+ main/bridging.c, main/dsp.c, main/global_datastores.c,
+ main/autoservice.c, main/dnsmgr.c, main/security_events.c,
+ main/say.c, main/utils.c, channels/misdn/isdn_msg_parser.c,
+ utils/astdb2sqlite3.c, main/devicestate.c, main/ssl.c,
+ main/format_pref.c, main/astobj2.c, main/indications.c,
+ main/chanvars.c, main/cli.c, main/tcptls.c, main/data.c,
+ main/plc.c, main/test.c, channels/console_board.c,
+ channels/misdn/portinfo.c, main/image.c, main/alaw.c,
+ channels/sig_ss7.c, main/asterisk.c, main/xmldoc.c,
+ main/format.c, main/strings.c, main/pbx.c, main/rtp_engine.c,
+ main/ccss.c, res/ais/clm.c: Merged revisions 369005 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10
+ ................ r369005 | kpfleming | 2012-06-15 11:07:08 -0500
+ (Fri, 15 Jun 2012) | 22 lines 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
+ ................
+
+2012-06-15 15:32 +0000 [r368963-368999] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.exports.in, /: Remove some symbol exports that
+ got missed in the removal of global symbols. (issue AST-807)
+ (issue AST-901) (issue AST-908) ........ Merged revisions 368998
+ from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+ * apps/app_voicemail.c, /: These functions that were moved need to
+ be static. Also wrap test functions in a #ifdef. (issue AST-807)
+ (issue AST-901) (issue AST-908) ........ Merged revisions 368964
+ from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+ * tests/test_voicemail_api.c, main/app.c,
+ include/asterisk/app_voicemail.h, apps/app_voicemail.c,
+ include/asterisk/app.h, /: Remove global symbol requirement from
+ app_voicemail. This uses the existing "function installation"
+ stuff that already existed for other functions, like getting
+ message counts. (closes issue AST-807) (issue AST-901) (issue
+ AST-908) Review: https://reviewboard.asterisk.org/r/1965/
+ ........ Merged revisions 368962 from
+ http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+2012-06-14 18:20 +0000 [r368872-368960] Automerge script <automerge at asterisk.org>
+
+ * /, channels/chan_skinny.c: Merged revisions 368947 via svnmerge
+ from file:///srv/subversion/repos/asterisk/branches/10 ........
+ r368947 | mjordan | 2012-06-14 12:31:33 -0500 (Thu, 14 Jun 2012)
+ | 21 lines 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) ........
+
+ * /, main/Makefile: Merged revisions 368928 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r368928 | mmichelson | 2012-06-14 10:25:23 -0500
+ (Thu, 14 Jun 2012) | 10 lines 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
+ ................
+
+ * res/res_adsi.c, res/res_smdi.c, /, funcs/func_volume.c: Merged
+ revisions 368895,368899 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r368895 | mjordan | 2012-06-13 15:27:28 -0500
+ (Wed, 13 Jun 2012) | 21 lines Mark res_smdi/res_adsi as 'core'
+ supported modules Recently, various issues surrounding weak
+ attributes 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 attributes. (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
+ ................ r368899 | mmichelson | 2012-06-13 16:13:30 -0500
+ (Wed, 13 Jun 2012) | 19 lines 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
+ ................
+
+ * /, main/Makefile: Merged revisions 368885 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r368885 | mmichelson | 2012-06-13 14:36:39 -0500
+ (Wed, 13 Jun 2012) | 11 lines 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
+ ................
+
+ * Makefile, /: Merged revisions 368831,368853 via svnmerge from
+ file:///srv/subversion/repos/asterisk/branches/10
+ ................ r368831 | mjordan | 2012-06-12 13:30:06 -0500
+ (Tue, 12 Jun 2012) | 24 lines 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) ........ Merged revisions 368830 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ................ r368853 | mjordan | 2012-06-13 09:30:34 -0500
+ (Wed, 13 Jun 2012) | 11 lines 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
+ ................
+
+2012-06-12 16:22 +0000 [r368810-368826] Jason Parker <jparker at digium.com>
+
+ * /: Let's fix the 1.8-merged prop, to give automerge the best
+ chance at succeeding.
+
+ * funcs/func_strings.c, channels/sip/reqresp_parser.c,
+ include/asterisk/channel.h, apps/app_queue.c,
+ channels/chan_iax2.c, main/loader.c, main/channel.c,
+ channels/chan_dahdi.c, channels/sig_analog.c,
+ res/res_config_odbc.c, channels/sip/dialplan_functions.c,
+ apps/app_directory.c, pbx/pbx_config.c, main/md5.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: Multiple revisions
+ 368721,368739,368760,368808 ........ r368721 | kmoore |
+ 2012-06-11 09:11:14 -0500 (Mon, 11 Jun 2012) | 8 lines 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. ........ Merged revisions 368719 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368739 | kmoore | 2012-06-11 10:15:07 -0500 (Mon, 11 Jun 2012) |
+ 10 lines 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) ........ Merged revisions
+ 368738 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+ ........ r368760 | rmudgett | 2012-06-11 12:08:50 -0500 (Mon, 11
+ Jun 2012) | 17 lines 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 ........ Merged revisions 368759 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ r368808 | mmichelson | 2012-06-12 10:37:38 -0500 (Tue, 12 Jun
+ 2012) | 15 lines 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
[... 25720 lines stripped ...]
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