[svn-commits] lmadsen: tag 1.6.0.11-rc1 r202930 - /tags/1.6.0.11-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Jun 24 13:16:11 CDT 2009
Author: lmadsen
Date: Wed Jun 24 13:16:07 2009
New Revision: 202930
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=202930
Log:
Importing files for 1.6.0.11-rc1 release.
Added:
tags/1.6.0.11-rc1/.lastclean (with props)
tags/1.6.0.11-rc1/.version (with props)
tags/1.6.0.11-rc1/ChangeLog (with props)
Added: tags/1.6.0.11-rc1/.lastclean
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.0.11-rc1/.lastclean?view=auto&rev=202930
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--- tags/1.6.0.11-rc1/ChangeLog (added)
+++ tags/1.6.0.11-rc1/ChangeLog Wed Jun 24 13:16:07 2009
@@ -1,0 +1,53308 @@
+2009-06-24 Leif Madsen <lmadsen at digium.com>
+
+ * Released Asterisk 1.6.0.11-rc1
+
+2009-06-24 18:09 +0000 [r202926] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 |
+ file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines
+ Ensure the default settings are applied for T.38 when we set it
+ up for a peer. ........
+
+2009-06-23 22:09 +0000 [r202763] Matthew Fredrickson <creslin at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) |
+ 1 line I could have sworn I committed this patch ages ago, but...
+ bug fix with setting NAI properly on linksets in certain
+ situations. ........
+
+2009-06-23 21:26 +0000 [r202754] Ryan Brindley <rbrindley at digium.com>
+
+ * main/config.c, /: Merged revisions 202753 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 |
+ rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9
+ lines If we delete the info, lets also delete the lines (closes
+ issue #14509) Reported by: timeshell Patches:
+ 20090504__bug14509.diff.txt uploaded by tilghman (license 14)
+ Tested by: awk, timeshell ........
+
+2009-06-23 16:40 +0000 [r202675] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009)
+ | 18 lines Merged revisions 202671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009)
+ | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to
+ non-standard port and transport (closes issue #14659) Reported
+ by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded
+ by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded
+ by dvossel (license 671) Tested by: dvossel, klaus3000 Review:
+ https://reviewboard.asterisk.org/r/288/ ........ ................
+
+2009-06-22 20:12 +0000 [r202498] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, /: Merged revisions 202497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009)
+ | 11 lines Merged revisions 202496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009)
+ | 4 lines Report CallerID change during a masquerade. Reported
+ by: markster ........ ................
+
+2009-06-22 16:30 +0000 [r202471] Sean Bright <sean at malleable.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun
+ 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid
+ potential crashes during reload. Pointed out by Russell while
+ working on the CEL branch. ........
+
+2009-06-22 16:06 +0000 [r202416] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009)
+ | 9 lines Merged revisions 202414 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009)
+ | 2 lines Make Polycom subscription type override check more
+ explicit. ........ ................
+
+2009-06-22 15:05 +0000 [r202338-202344] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun
+ 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun
+ 2009) | 26 lines Fix a situation in which Asterisk would not stop
+ retransmitting 487s. If a CANCEL were received by Asterisk, we
+ would send a 487 in response to the original INVITE and a 200 OK
+ for the CANCEL. If there were a network hiccup which caused the
+ 200 OK and the 487 to be lost, then the UA communicating with
+ Asterisk may try to retransmit its CANCEL. Asterisk's response to
+ this used to be to try sending another 487 to the canceled INVITE
+ and another 200 OK to the CANCEL. The problem here is that the
+ originally-sent 487 was sent "reliably" meaning that it will be
+ retransmitted until it is received properly. So when we receive
+ the second CANCEL it is likely that the first batch of 487s we
+ sent is still going strong and reaches the UA. The result was
+ that the second set of 487s would be retransmitted constantly
+ until the maximum number of retries had been reached. The fix for
+ this is that if we receive a second CANCEL for an INVITE, then we
+ cancel the retransmission of the first set of 487s and start a
+ second set. This causes the dialog to be terminated reasonably.
+ (closes issue #14584) Reported by: klaus3000 Patches:
+ 14584_v2.patch uploaded by mmichelson (license 60) Tested by:
+ klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58
+ -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line
+ left from previous commit. ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun
+ 2009) | 31 lines Merged revisions 202336 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun
+ 2009) | 25 lines Fix a possible infinite loop in SDP parsing
+ during glare situation. There was a while loop in
+ get_ip_and_port_from_sdp which was controlled by a call to
+ get_sdp_iterate. The loop would exit either if what we were
+ searching for was found or if the return was NULL. The problem is
+ that get_sdp_iterate never returns NULL. This means that if what
+ we were searching for was not present, the loop would run
+ infinitely. This modification of the loop fixes the problem.
+ (closes issue #15213) Reported by: schmidts (closes issue #15349)
+ Reported by: samy (closes issue #14464) Reported by: pj (closes
+ issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
+ uploaded by mmichelson (license 60) Tested by: aragon ........
+ ................
+
+2009-06-21 16:14 +0000 [r202259-202263] Russell Bryant <russell at digium.com>
+
+ * cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 |
+ russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines
+ Fix possibility of crashiness during reload in custom fields
+ handling. ........
+
+ * cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 |
+ russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines
+ Standardize return values of load_config() so reload() doesn't
+ report an error on success. ........
+
+2009-06-20 19:14 +0000 [r202184] Sean Bright <sean at malleable.com>
+
+ * /, apps/app_fax.c: Merged revisions 202183 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 |
+ seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5
+ lines Fix version detection for API changes in spandsp. (closes
+ issue #15355) Reported by: deuffy ........
+
+2009-06-19 21:07 +0000 [r202006] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Added deadlock protection to
+ try_suggested_sip_codec in chan_sip.c. Review:
+ https://reviewboard.asterisk.org/r/287/
+
+2009-06-19 20:27 +0000 [r201997] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 201994 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500
+ (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009)
+ | 8 lines timestamp was being converted to host order as a short
+ rather than a long (closes issue #15361) Reported by: ffloimair
+ Patches: ts_issue.diff uploaded by dvossel (license 671) ........
+ ................
+
+2009-06-19 00:44 +0000 [r201786-201830] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/features.c: Merged revisions 201829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009)
+ | 13 lines Merged revisions 201828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009)
+ | 6 lines If the "h" extension fails, give it another chance in
+ main/pbx.c. If the "h" extension fails, give it another chance in
+ main/pbx.c, when it returns from the bridge code. Fixes an issue
+ where the "h" extension may occasionally not fire, when a Dial is
+ executed from a Macro. Debugged in #asterisk with user tompaw.
+ ........ ................
+
+ * /, apps/Makefile: Merged revisions 201783 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 |
+ tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines
+ One of the changes in 1.6.1 was to allow app_directory to use
+ functionality within app_voicemail for directory functions. It is
+ therefore no longer necessary for app_directory to be linked
+ against the ODBC libraries (and it never was necessary for
+ app_directory to be linked against IMAP, though it was). ........
+
+2009-06-18 16:58 +0000 [r201682] David Vossel <dvossel at digium.com>
+
+ * channels/misdn/isdn_lib.c, main/asterisk.c, utils/conf2ael.c,
+ main/ast_expr2.c, utils/stereorize.c,
+ codecs/gsm/src/gsm_destroy.c, /, channels/h323/ast_h323.cxx,
+ main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c,
+ utils/extconf.c, pbx/pbx_config.c, res/res_config_ldap.c: Merged
+ revisions 201678 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 |
+ dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines
+ fixes some memory leaks and redundant conditions (closes issue
+ #15269) Reported by: contactmayankjain Patches: patch.txt
+ uploaded by contactmayankjain (license 740)
+ memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
+ Tested by: contactmayankjain, dvossel ........
+
+2009-06-18 15:32 +0000 [r201612] Russell Bryant <russell at digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 201610 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201610 | russell | 2009-06-18 10:27:10 -0500
+ (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009)
+ | 29 lines Fix memory corruption and leakage related reloads of
+ non files mode MoH classes. For Music on Hold classes that are
+ not files mode, meaning that we are executing an application that
+ will feed us audio data, we use a thread to monitor the external
+ application and read audio from it. This thread also makes use of
+ the MoH class object. In the MoH class destructor, we used
+ pthread_cancel() to ask the thread to exit. Unfortunately, the
+ code did not wait to ensure that the thread actually went away.
+ What needed to be done is a pthread_join() to ensure that the
+ thread fully cleans up before we proceed. By adding this one
+ line, we resolve two significant problems: 1) Since the thread
+ was never joined, it never fully goes away. So, on every reload
+ of non-files mode MoH, an unused thread was sticking around. 2)
+ There was a race condition here where the application monitoring
+ thread could still try to access the MoH class, even though the
+ thread executing the MoH reload has already destroyed it. (issue
+ #15109) Reported by: jvandal (issue #15123) Reported by:
+ axisinternet (issue #15195) Reported by: amorsen (issue AST-208)
+ ........ ................
+
+2009-06-17 20:10 +0000 [r201459-201463] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 |
+ mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12
+ lines Fix problem with no audio due to ignoring the SDP. A recent
+ change to our SDP version comparison made audio not function on
+ some calls. This was because of a test wherein we were trying to
+ see if an unsigned value was less than 0. This is a dumb
+ comparison and arguably the compiler should have warned about it.
+ Alas, though, it slipped past. Now it's fixed by changing the
+ variable to be a signed type. Found by several developers. Tested
+ by mnicholson and dbrooks. ........
+
+ * main/channel.c, /: Merged revisions 201458 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun
+ 2009) | 15 lines Merged revisions 201450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun
+ 2009) | 9 lines Change the datastore traversal in
+ ast_do_masquerade to use a safe list traversal. It is possible
+ for datastore fixup functions to remove the datastore from the
+ list and free it. In particular, the queue_transfer_fixup in
+ app_queue does this. While I don't yet know of this causing any
+ crashes, it certainly could. Found while discussing a separate
+ issue with Brian Degenhardt. ........ ................
+
+2009-06-17 19:55 +0000 [r201449] David Vossel <dvossel at digium.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500
+ (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009)
+ | 19 lines StopMixMonitor race condition (not giving up file
+ immediately) StopMixMonitor only indicates to the MixMonitor
+ thread to stop writing to the file. It does not guarantee that
+ the recording's file handle is available to the dialplan
+ immediately after execution. This results in a race condition. To
+ resolve this, the filestream pointer is placed in a datastore on
+ the channel. When StopMixMonitor is called, the datastore is
+ retrieved from the channel and the filestream is closed
+ immediately before returning to the dialplan. Documentation
+ indicating the use of StopMixMonitor to free files has been
+ updated as well. (closes issue #15259) Reported by: travisghansen
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/283/ ........ ................
+
+2009-06-17 19:35 +0000 [r201443] David Brooks <dbrooks at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009)
+ | 16 lines Merged revisions 201380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009)
+ | 9 lines Checks for NULL sip_pvt pointer in
+ chan_sip.c->acf_channel_read() Zombie channels could be passed,
+ and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
+ checking for NULL pointer. (closes issue #15330) Reported by:
+ okrief Tested by: dbrooks ........ ................
+
+2009-06-17 12:05 +0000 [r201263] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 201262 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500
+ (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun
+ 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list
+ to be appended is empty. When the list to be appended is empty,
+ and the list to be appended to is *not*, AST_LIST_APPEND_LIST
+ would actually cause the target list to become broken, and no
+ longer have a pointer to its last entry. This patch fixes the
+ problem. (reported by Stanislaw Pitucha on the asterisk-dev
+ mailing list) ........ ................
+
+2009-06-16 22:31 +0000 [r201226] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 |
+ dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines
+ fix issue with build_contact introduced by the "SIP trasnport
+ type issues" commit ........
+
+2009-06-16 19:34 +0000 [r201093] Kevin P. Fleming <kpfleming at digium.com>
+
+ * apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c,
+ main/autoservice.c, main/frame.c, /, apps/app_meetme.c,
+ configure, main/slinfactory.c, autoconf/ast_gcc_attribute.m4,
+ configure.ac, include/asterisk/linkedlists.h, main/file.c,
+ include/asterisk/channel.h, include/asterisk/frame.h: Merged
+ revisions 201056,201090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun
+ 2009) | 18 lines Merged revisions 200991 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun
+ 2009) | 11 lines Improve support for media paths that can
+ generate multiple frames at once. There are various media paths
+ in Asterisk (codec translators and UDPTL, primarily) that can
+ generate more than one frame to be generated when the application
+ calling them expects only a single frame. This patch addresses a
+ number of those cases, at least the primary ones to solve the
+ known problems. In addition it removes the broken TRACE_FRAMES
+ support, fixes a number of bugs in various frame-related API
+ functions, and cleans up various code paths affected by these
+ changes. https://reviewboard.asterisk.org/r/175/ ........
+ ................ r201090 | kpfleming | 2009-06-16 14:27:12 -0500
+ (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler
+ attribute checking. Defaulting to 'static' for the function scope
+ was bad... so remove it. ................
+
+2009-06-16 17:11 +0000 [r200992] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 |
+ dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines
+ SIP transport type issues What this patch addresses: 1.
+ ast_sip_ouraddrfor() by default binds to the UDP address/port
+ reguardless if the sip->pvt is of type UDP or not. Now when no
+ remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
+ transport type, attempting to set the address and port to the
+ correct TCP/TLS bindings if necessary. 2. It is not necessary to
+ send the port number in the Contact header unless the port is
+ non-standard for the transport type. This patch fixes this and
+ removes the todo note. 3. In sip_alloc(), the default dialog
+ built always uses transport type UDP. Now sip_alloc() looks at
+ the sip_request (if present) and determines what transport type
+ to use by default. 4. When changing the transport type of a
+ sip_socket, the file descriptor must be set to -1 and in some
+ cases the tcptls_session's ref count must be decremented and set
+ to NULL. I've encountered several issues associated with this
+ process and have created a function, set_socket_transport(), to
+ handle the setting of the socket type. (closes issue #13865)
+ Reported by: st Patches: dont_add_port_if_tls.patch uploaded by
+ Kristijan (license 753) 13865.patch uploaded by mmichelson
+ (license 60) tls_port_v5.patch uploaded by vrban (license 756)
+ transport_issues.diff uploaded by dvossel (license 671) Tested
+ by: mmichelson, Kristijan, vrban, jmacz, dvossel Review:
+ https://reviewboard.asterisk.org/r/278/ ........
+
+2009-06-16 16:34 +0000 [r200986] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
+ revisions 200985 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 |
+ kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7
+ lines Fix problems with new compiler attribute checking in
+ configure script. The last changes to ast_gcc_attribute.m4 caused
+ some problems checking for various attributes, because the scope
+ of the symbol the attribute is applied to can be important; this
+ patch allows the scope to be specified for the check. ........
+
+2009-06-16 16:02 +0000 [r200945] Michiel van Baak <michiel at vanbaak.info>
+
+ * apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009)
+ | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail
+ can only use one storage module at the moment. Because it's
+ unclear that selecting one of the storage modules in menuselect
+ will disable filesystem storage we now have a FILE_STORAGE option
+ that conflicts with the other modules. (closes issue #15333)
+ ........
+
+2009-06-16 01:33 +0000 [r200724-200767] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in,
+ autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15
+ Jun 2009) | 11 lines Ensure that configure-script testing for
+ compiler attributes actually works. The configure script tests
+ for compiler attributes didn't actually enable enough warnings or
+ provide a proper test harness to determine whether the compiler
+ supports the attribute in question or not; this caused gcc 4.1 to
+ report that it supports 'weakref', but it doesn't actually
+ support it in the way that is needed for our optional API
+ mechanism. The new configure script test will properly
+ distinguish between full support and partial support for this
+ attribute, among others. ........
+
+ * /, CHANGES: Merged revisions 200726 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 |
+ kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6
+ lines Document the new automatic 'ignoresdpversion' behavior.
+ Asterisk will now automatically ignore incorrect incoming SDP
+ version numbers when necessary to complete a T.38 re-INVITE
+ operation. ........
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 165180,200689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 |
+ mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14
+ lines This patch adds a new 'ignoresdpversion' option to
+ sip.conf. When this is enabled (either globally or for a specific
+ peer), chan_sip will treat any SDP data it receives as new data
+ and update the media stream accordingly. By default, Asterisk
+ will only modify the media stream if the SDP session version
+ received is different from the current SDP session version. This
+ option is required to interoperate with devices that have
+ non-standard SDP session version implementations (observed by toc
+ on the bug tracker with Microsoft OCS which always uses 0 as the
+ session version). http://reviewboard.digium.com/r/94/ (closes
+ issue #13958) Reported by: toc Tested by: toc ........ r200689 |
+ kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12
+ lines Accept T.38 re-INVITE responses with invalid SDP versions.
+ This commit changes the 'incoming SDP version' check logic a bit
+ more; when 'ignoresdpversion' is *not* set for a peer, if we
+ initiate a re-INVITE to switch to T.38, we'll always accept the
+ peer's SDP response, even if they don't properly increment the
+ SDP version number as they should. If this situation occurs, a
+ warning message will be generated suggesting that the peer's
+ configuration be changed to include the 'ignoresdpversion'
+ configuration option (although ideally they'd fix their SIP
+ implementation to be RFC compliant). AST-221 ........
+
+2009-06-15 15:22 +0000 [r200515] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun
+ 2009) | 11 lines Merged revisions 200513 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun
+ 2009) | 5 lines Add INFO to our allowed methods so that endpoints
+ know they may send it to us. AST-223 ........ ................
+
+2009-06-12 19:08 +0000 [r200362] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c, /: Merged revisions 200361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun
+ 2009) | 16 lines Merged revisions 200360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun
+ 2009) | 10 lines Suppress a warning message and give a better
+ return code when generating inband ringing after a call is
+ answered. (closes issue #15158) Reported by: madkins Patches:
+ 15158.patch uploaded by mmichelson (license 60) Tested by:
+ madkins ........ ................
+
+2009-06-11 22:42 +0000 [r200228] Sean Bright <sean at malleable.com>
+
+ * Makefile, /: Merged revisions 199781 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 |
+ seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2
+ lines Fix all of the parallel build warnings issued when running
+ make -j#. ........
+
+2009-06-11 21:18 +0000 [r200149] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 |
+ mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5
+ lines Fix a crash due to a potentially NULL p->options. Thanks to
+ mnicholson for pointing it out. ........
+
+2009-06-11 12:16 +0000 [r200040] Leif Madsen <lmadsen at digium.com>
+
+ * /, build_tools/make_version_c, build_tools/make_version_h: Merged
+ revisions 200039 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 |
+ lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
+ Fix path for .flavor and .version (issue #14737) Reported by:
+ davidw Patches: flavor.patch uploaded by davidw (license 780)
+ Tested by: davidw ........
+
+2009-06-10 20:29 +0000 [r199994] David Brooks <dbrooks at digium.com>
+
+ * main/pbx.c, /: Fixes the argument order in definition of
+ new_find_extension(). In the definition of new_find_extension(),
+ the arguments 'callerid' and 'label' were swapped. The prototype
+ declaration and all calls to the function are ordered 'callerid'
+ then 'label', but the function itself was ordered 'label' then
+ 'callerid'. (closes issue #15303) Reported by: JimDickenson
+
+2009-06-10 20:20 +0000 [r199975] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: The 1.6.0 branch was missing all
+ invite_branch logic. It has now been added.
+
+2009-06-10 16:13 +0000 [r199858] Sean Bright <sean at malleable.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400
+ (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
+ 10 Jun 2009) | 2 lines __WORDSIZE is not available on all
+ platforms, so use sizeof(void *) instead. ........
+ ................
+
+2009-06-09 20:54 +0000 [r199821] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 |
+ dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
+ CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command
+ only used UDP rather than copying the transport type from the
+ peer. (closes issue #15283) Reported by: jthurman Patches:
+ sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
+ Tested by: jthurman, dvossel ........
+
+2009-06-08 19:39 +0000 [r199632] Sean Bright <sean at malleable.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400
+ (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
+ 2009) | 21 lines Increase the size of our thread stack on 64 bit
+ processors. We were setting the stack size for each thread to
+ 240KB regardless of architecture, which meant that in some
+ scenarios we actually had less available stack space on 64 bit
+ processors (pointers use 8 bytes instead of 4). So now we
+ calculate the stack size we reserve based on the platform's
+ __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
+ bit -> 1008KB (that's right, we're ready for 128 bit processors)
+ Patch typed by me but written by several members of
+ #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
+ issue #14932) Reported by: jpiszcz Patches:
+ 06052009_issue14932.patch uploaded by seanbright (license 71)
+ Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
+ 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
+ stack size calculation just introduced. ........ ................
+
+2009-06-05 21:37 +0000 [r199301] David Vossel <dvossel at digium.com>
+
+ * main/pbx.c, /: Merged revisions 199298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009)
+ | 21 lines Merged revisions 199297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
+ | 14 lines Fixes issue with hints giving unexpected results.
+ Hints with two or more devices that include ONHOLD gave
+ unexpected results. (closes issue #15057) Reported by:
+ p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
+ (license 671) pbx.c.1.4.patch uploaded by p (license 558)
+ devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
+ p_lindheimer, dvossel Review:
+ https://reviewboard.asterisk.org/r/254/ ........ ................
+
+2009-06-05 13:51 +0000 [r199228] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun
+ 2009) | 14 lines Correct "dahdi show channels" output when
+ specifying a group. Since a DAHDI channel may belong to multiple
+ groups, we need to use a bitwise and instead of equivalence to
+ determine whether to display the channel information. (closes
+ issue #15248) Reported by: gentian Patches: 15248.patch uploaded
+ by mmichelson (license 60) Tested by: gentian ........
+
+2009-06-04 19:16 +0000 [r199142] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 199139 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500
+ (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
+ Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
+ ................
+
+2009-06-04 14:53 +0000 [r199052] Sean Bright <sean at malleable.com>
+
+ * main/asterisk.c, main/loader.c, /, include/asterisk/_private.h:
+ Merged revisions 199051 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun
+ 2009) | 47 lines Merged revisions 199022 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
+ 2009) | 40 lines Safely handle AMI connections/reload requests
+ that occur during startup. During asterisk startup, a lock on the
+ list of modules is obtained by the primary thread while each
+ module is initialized. Issue 13778 pointed out a problem with
+ this approach, however. Because the AMI is loaded before other
+ modules, it is possible for a module reload to be issued by a
+ connected client (via Action: Command), causing a deadlock. The
+ resolution for 13778 was to move initialization of the manager to
+ happen after the other modules had already been lodaded. While
+ this fixed this particular issue, it caused a problem for users
+ (like FreePBX) who call AMI scripts via an #exec in a
+ configuration file (See issue 15189). The solution I have come up
+ with is to defer any reload requests that come in until after the
+ server is fully booted. When a call comes in to ast_module_reload
+ (from wherever) before we are fully booted, the request is added
+ to a queue of pending requests. Once we are done booting up, we
+ then execute these deferred requests in turn. Note that I have
+ tried to make this a bit more intelligent in that it will not
+ queue up more than 1 request for the same module to be reloaded,
+ and if a general reload request comes in ('module reload') the
+ queue is flushed and we only issue a single deferred reload for
+ the entire system. As for how this will impact existing
+ installations - Before 13778, a reload issued before module
+ initialization was completed would result in a deadlock. After
+ 13778, you simply couldn't connect to the manager during startup
+ (which causes problems with #exec-that-calls-AMI configuration
+ files). I believe this is a good general purpose solution that
+ won't negatively impact existing installations. (closes issue
+ #15189) (closes issue #13778) Reported by: p_lindheimer Patches:
+ 06032009_15189_deferred_reloads.diff uploaded by seanbright
+ (license 71) Tested by: p_lindheimer, seanbright Review:
+ https://reviewboard.asterisk.org/r/272/ ........ ................
+
+2009-06-03 15:27 +0000 [r198825-198889] David Vossel <dvossel at digium.com>
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ Merged revisions 198856 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 |
+ dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
+ Generic call forward api, ast_call_forward() The function
+ ast_call_forward() forwards a call to an extension specified in
+ an ast_channel's call_forward string. After an ast_channel is
+ called, if the channel's call_forward string is set this function
+ can be used to forward the call to a new channel and terminate
+ the original one. I have included this api call in both
+ channel.c's ast_request_and_dial() and feature.c's
+ feature_request_and_dial(). App_dial and app_queue already
+ contain call forward logic specific for their application and
+ options. (closes issue #13630) Reported by: festr Review:
+ https://reviewboard.asterisk.org/r/271/ ........
+
+ * /, channels/chan_iax2.c: Merged revisions 198824 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
[... 52644 lines stripped ...]
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