[svn-commits] lmadsen: tag 1.6.1.0-rc4 r186326 - /tags/1.6.1.0-rc4/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Apr 3 11:12:35 CDT 2009
Author: lmadsen
Date: Fri Apr 3 11:12:31 2009
New Revision: 186326
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=186326
Log:
Importing files for 1.6.1.0-rc4 release.
Added:
tags/1.6.1.0-rc4/.lastclean (with props)
tags/1.6.1.0-rc4/.version (with props)
tags/1.6.1.0-rc4/ChangeLog (with props)
Added: tags/1.6.1.0-rc4/.lastclean
URL: http://svn.digium.com/svn-view/asterisk/tags/1.6.1.0-rc4/.lastclean?view=auto&rev=186326
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--- tags/1.6.1.0-rc4/ChangeLog (added)
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+2009-04-03 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.1.0-rc4 released.
+
+2009-04-03 15:54 +0000 [r186323] Joshua Colp <jcolp at digium.com>
+
+ * include/asterisk/crypto.h, /: Merged revisions 186321 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri,
+ 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
+ lines Fix a problem with the crypto variable definitions not
+ actually being defined properly. (closes issue #14804) Reported
+ by: jvandal ........ ................
+
+2009-04-03 14:33 +0000 [r186288] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr
+ 2009) | 20 lines Fix the ability to retrieve voicemail messages
+ from IMAP. A recent change made interactive vm_states no longer
+ get added to the list of vm_states and instead get stored in
+ thread-local storage. In trunk and all the 1.6.X branches, the
+ problem is that when we search for messages in a voicemail box,
+ we would attempt to update the appropriate vm_state struct by
+ directly searching in the list of vm_states instead of using the
+ get_vm_state_by_imap_user function. This meant we could not find
+ the interactive vm_state that we wanted. (closes issue #14685)
+ Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
+ (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........
+
+2009-04-03 02:06 +0000 [r186232] Russell Bryant <russell at digium.com>
+
+ * cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009)
+ | 29 lines Merged revisions 186229 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
+ | 21 lines Fix a memory leak in cdr_radius. I came across this
+ while doing some testing of my ast_channel_ao2 branch. After
+ running a test overnight that generated over 5 million calls,
+ Asterisk had taken up about 1 GB of my system memory. So, I
+ re-ran the test with MALLOC_DEBUG turned on. However, it showed
+ no leaks in Asterisk during the test, even though Asterisk was
+ still consuming it somehow. Instead, I turned to valgrind, which
+ when run with --leak-check=full, told me exactly where the leak
+ came from, which was from allocations inside the radiusclient-ng
+ library. This explains why MALLOC_DEBUG did not report it. After
+ a bit of analysis, I found that we were leaking a little bit of
+ memory every time a CDR record was passed to cdr_radius. I don't
+ actually have a radius server set up to receive CDR records.
+ However, I always have my development systems compile and install
+ all modules. In addition to making sure there are not build
+ errors across modules, always loading modules helps find bugs
+ like this, too, so it is strongly recommend for all developers.
+ ........ ................
+
+2009-04-02 21:59 +0000 [r186177] Mark Michelson <mmichelson at digium.com>
+
+ * configs/features.conf.sample, /: Merged revisions 186175 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500
+ (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
+ 2009) | 5 lines Fix instructions in one-step parking comment to
+ make more sense. Changed a capital K to a lowercase k. ........
+ ................
+
+2009-04-02 17:27 +0000 [r186108] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500
+ (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
+ Apr 2009) | 3 lines ensure that the buffer passed to
+ DAHDI_SET_BUFINFO is fully initialized ........ ................
+
+2009-04-02 17:14 +0000 [r186062] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
+ 186060 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009)
+ | 16 lines Merged revisions 186059 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
+ (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
+ Apr 2009) | 2 lines Fix for AST-2009-003 ........
+ ................ ................
+
+2009-04-02 13:53 +0000 [r185956] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500
+ (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr
+ 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
+ DAHDI_GET_PARAMS ioctls were recently corrected to show that they
+ do, in fact, read data from userspace as part of their work. due
+ to this fix, valgrind now reports a number of cases where
+ chan_dahdi passed an uninitialized (or partially) buffer to these
+ ioctls, which could lead to unexpected behavior. this patch
+ corrects chan_dahdi to ensure that buffers passed to these ioctls
+ are always fully initialized. ........ ................
+
+2009-04-01 19:06 +0000 [r185848] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009)
+ | 16 lines Merged revisions 185845 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
+ | 10 lines Fixes issue with dropped calles due to re-Invite glare
+ and re-Invites never executing after a 491 Acknowledgement for
+ 491 responses were never being processed because it didn't match
+ our pending invite's seqno. Since the ACK was never processed,
+ the 491 frame would continue to be retransmitted until eventually
+ the call was dropped due to max retries. Now during a pending
+ invite, if we receive another invite, we send an 491 and hold on
+ to that glare invite's seqno in the "glareinvite" variable for
+ that sip_pvt struct. When ACK's are received, we first check to
+ see if it is in response to our pending invite, if not we check
+ to see if it is in response to a glare invite. In this case, it
+ is in response to the glare invite and must be dealt with or the
+ call is dropped. I've changed the wait time for resending the
+ re-Invite after receving a 491 response to comply with RFC 3261.
+ Before this patch the scheduled re-Invite would only change a
+ flag indicating that the re-Invite should be sent out, now it
+ actually sends it out as well. (closes issue #12013) Reported by:
+ alx Review: http://reviewboard.digium.com/r/213/ ........
+ ................
+
+2009-04-01 13:50 +0000 [r185774] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, /: Merged revisions 185772 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009)
+ | 14 lines Merged revisions 185771 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009)
+ | 6 lines Fix a case where DTMF could bypass audiohooks. This
+ change fixes a situation where an audiohook that wants DTMF would
+ not actually get it. This is in the code path where we end DTMF
+ digit length emulation while handling a NULL frame. ........
+ ................
+
+2009-03-31 22:38 +0000 [r185666] Kevin P. Fleming <kpfleming at digium.com>
+
+ * utils, /: Merged revisions 185664 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 |
+ kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line
+ ignore copied (generated) file ........
+
+2009-03-31 22:05 +0000 [r185471-185602] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185600 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar
+ 2009) | 12 lines Merged revisions 185599 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar
+ 2009) | 6 lines Fix crash that would occur if an empty member was
+ specified in queues.conf. (closes issue #14796) Reported by: pida
+ ........ ................
+
+ * apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500
+ (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar
+ 2009) | 8 lines Fix Russian voicemail intro to say the word
+ "messages" properly. (closes issue #14736) Reported by: chappell
+ Patches: voicemail_no_messages.diff uploaded by chappell (license
+ 8) ........ ................
+
+2009-03-31 17:48 +0000 [r185427] David Brooks <dbrooks at digium.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 185363 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500
+ (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009)
+ | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains
+ extra whitespaces To drill into the xmpp to find the capabilities
+ between channels, chan_gtalk calls iks_child() and iks_next().
+ iks_child() and iks_next() are functions in the iksemel xml
+ parsing library that traverse xml nodes. The bug here is that
+ both iks_child() and iks_next() will return the next iks_struct
+ node *regardless* of type. chan_gtalk expects the next node to be
+ of type IKS_TAG, which in most cases, it is, but in this case (a
+ call being made from the Empathy IM client), there exists
+ iks_struct nodes which are not IKS_TAG data (they are extraneous
+ whitespaces), and chan_gtalk doesn't handle that case, so
+ capabilities don't match, and a call cannot be made.
+ iks_first_tag() and iks_next_tag(), on the other hand, will not
+ return the very next iks_struct, but will check to see if the
+ next iks_struct is of type IKS_TAG. If it isn't, it will be
+ skipped, and the next struct of type IKS_TAG it finds will be
+ returned. This assures that chan_gtalk will find the iks_struct
+ it is looking for. This fix simply changes all calls to
+ iks_child() and iks_next() to become calls to iks_first_tag() and
+ iks_next_tag(), which resolves the capability matching. The
+ following is a payload listing from Empathy, which, due to the
+ extraneous whitespace, will not be parsed correctly by iksemel:
+ <iq from='dbrooksjab at 235-22-24-10/Telepathy'
+ to='astjab at 235-22-24-10/asterisk' type='set' id='542757715704'>
+ <session xmlns='http://www.google.com/session'
+ initiator='dbrooksjab at 235-22-24-10/Telepathy' type='initiate'
+ id='1837267342'> <description
+ xmlns='http://www.google.com/session/phone'> <payload-type
+ clockrate='16000' name='speex' id='96'/> <payload-type
+ clockrate='8000' name='PCMA' id='8'/> <payload-type
+ clockrate='8000' name='PCMU' id='0'/> <payload-type
+ clockrate='90000' name='MPA' id='97'/> <payload-type
+ clockrate='16000' name='SIREN' id='98'/> <payload-type
+ clockrate='8000' name='telephone-event' id='99'/> </description>
+ </session> </iq> Review: http://reviewboard.digium.com/r/181/
+ ........ ................
+
+2009-03-31 14:57 +0000 [r185263] Russell Bryant <russell at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185261 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 |
+ russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines
+ Don't free() an astobj2 object. (closes issue #14672) Reported
+ by: makoto ........
+
+2009-03-31 14:10 +0000 [r185199] Joshua Colp <jcolp at digium.com>
+
+ * /, main/audiohook.c: Merged revisions 185197 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) |
+ 15 lines Merged revisions 185196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8
+ lines Fix crash when moving audiohooks between channels. Handle
+ the scenario where we are called to move audiohooks between
+ channels and the source channel does not actually have any on it.
+ (closes issue #14734) Reported by: corruptor ........
+ ................
+
+2009-03-30 20:50 +0000 [r185126-185127] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn_config.c, /, configs/misdn.conf.sample: Merged
+ revisions 185123 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009)
+ | 9 lines Merged revisions 185121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009)
+ | 1 line Update the channel allocation method documentation.
+ ........ ................
+
+ * channels/misdn/isdn_lib.c, /: Merged revisions 185122 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500
+ (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009)
+ | 19 lines Make chan_misdn BRI TE side normally defer channel
+ selection to the NT side. Channel allocation collisions are not
+ handled by chan_misdn very well. This patch simply avoids the
+ problem for BRI only. For PRI, allocation collisions are still
+ possible but less likely since there are simply more channels
+ available and each end could use a different allocation strategy.
+ misdn.conf options available: te_choose_channel - Use to force
+ the TE side to allocate channels. method - Specify the channel
+ allocation strategy. (closes issue #13488) Reported by:
+ Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich
+ Tested by: crich, siepkes, festr ........ ................
+
+2009-03-30 16:47 +0000 [r185088] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185072 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar
+ 2009) | 45 lines Merged revisions 185031 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar
+ 2009) | 39 lines Fix queue weight behavior so that calls in
+ low-weight queues are not inappropriately blocked. (This is
+ copied and pasted from the review request I made for this patch)
+ Asterisk has some odd behavior when queue weights are used. The
+ current logic used when potentially calling a queue member is: If
+ the member we are going to call is part of another queue and
+ _that other queue has any callers in it_ and has a higher weight
+ than the queue we are calling from, then don't try to contact
+ that member. The issue here is what I have marked with
+ underscores. If the higher-weighted queue has any callers in it
+ at all, then the queue member will be unreachable from the
+ lower-weighted queue. This has the potential to be really really
+ bad if using a queue strategy, such as leastrecent or
+ fewestcalls, with the potential to call the same member
+ repeatedly. The fix proposed by garychen on issue 13220 is very
+ simple and, as far as I can see, works well for this situation.
+ With this set of changes, the logic used becomes: If the member
+ we are going to call is part of another queue, the other queue
+ has a higher weight than the queue we are calling from, and the
+ higher weight queue has at least as many callers as available
+ members, then do not try to contact the queue member. If the
+ higher weighted queue has fewer callers than available members,
+ then there is no reason to deny the call to this member since the
+ other queue can afford to spare a member. Since the fix involved
+ writing a generic function for determining the number of
+ available members in the queue, I also modified the is_our_turn
+ function to make use of the new num_available_members function to
+ determine if it is our turn to try calling a member. There is one
+ small behavior change. Before writing this patch, if you had
+ autofill disabled, then if you were the head caller in a queue,
+ you would automatically be told that it was your turn to try
+ calling a member. This did not take into account whether there
+ were actually any queue members available to take the call. Now
+ we actually make sure there is at least one member available to
+ take the call if autofill is disabled. (closes issue #13220)
+ Reported by: garychen Review:
+ http://reviewboard.digium.com/r/202/ ........ ................
+
+2009-03-30 14:41 +0000 [r184950] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) |
+ 21 lines Merged revisions 184947 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) |
+ 14 lines Improve our handling of T38 in the initial INVITE from a
+ device. We now answer with matching media streams to what is
+ requested. If an INVITE is received with both a T38 and RTP media
+ stream this means we answer with both. For any outgoing calls
+ created as a result of this inbound one no T38 is requested in
+ the initial INVITE. Instead if we start receiving udptl packets
+ we trigger a reinvite on the outbound side. (closes issue #12437)
+ Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu
+ Review: http://reviewboard.digium.com/r/208/ ........
+ ................
+
+2009-03-30 13:57 +0000 [r184912] Russell Bryant <russell at digium.com>
+
+ * channels/h323/Makefile.in, /: Merged revisions 184910 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30
+ Mar 2009) | 4 lines Fix build error when chan_h323 is not being
+ built. (reported by cai1982 in #asterisk-dev) ........
+
+2009-03-29 05:52 +0000 [r184840-184845] Russell Bryant <russell at digium.com>
+
+ * apps/app_followme.c, /: Merged revisions 184843 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009)
+ | 13 lines Merged revisions 184842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009)
+ | 5 lines Ensure targs variable is fully initialized. (closes
+ issue #14758) Reported by: tim_ringenbach ........
+ ................
+
+ * channels/Makefile, /: Merged revisions 184838 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 |
+ russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines
+ Simplify chan_h323 build to not require a second run of "make".
+ (closes issue #14715) Reported by: jthurman Patches:
+ h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license
+ 614) Tested by: tzafrir, russell ........
+
+2009-03-27 19:17 +0000 [r184765] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_iax2.c, main/timing.c, main/channel.c, /,
+ include/asterisk/timing.h, include/asterisk/channel.h: Merged
+ revisions 184762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 |
+ kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12
+ lines Improve timing interface to remember which provider
+ provided a timer The ability to load/unload timing interfaces is
+ nice, but it means that when a timer is allocated, it may come
+ from provider A, but later provider B becomes the 'preferred'
+ provider. If this happens, all timer API calls on the timer that
+ was provided by provider A will actually be handed to provider B,
+ which will say WTF and return an error. This patch changes the
+ timer API to include a pointer to the provider of the timer
+ handle so that future operations on the timer will be forwarded
+ to the proper provider. (closes issue #14697) Reported by: moy
+ Review: http://reviewboard.digium.com/r/211/ ........
+
+2009-03-27 18:09 +0000 [r184728] Russell Bryant <russell at digium.com>
+
+ * main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27
+ Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure
+ we use the best RNG available. ........
+
+2009-03-27 15:54 +0000 [r184675] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_agi.c: Merged revisions 184673 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 |
+ file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix
+ speech structure leak in the AGI speech recognition integration.
+ The AGI dialplan applications did not destroy the speech
+ structure automatically if it was not destroyed by the running
+ AGI script. They will now do this. (issue LUMENVOX-15) ........
+
+2009-03-27 14:04 +0000 [r184631] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /,
+ res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions
+ 184630 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 |
+ russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines
+ Change g_eid to ast_eid_default. ........
+
+2009-03-27 13:22 +0000 [r184587] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) |
+ 16 lines Merged revisions 184565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9
+ lines Fix an issue where nat=yes would not always take effect for
+ the RTP session on outgoing calls. If calls were placed using an
+ IP address or hostname the global nat setting was copied over but
+ was not set on the RTP session itself. This caused the RTP stack
+ to not perform symmetric RTP actions. (closes issue #14546)
+ Reported by: acunningham ........ ................
+
+2009-03-27 02:25 +0000 [r184513-184547] Russell Bryant <russell at digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009)
+ | 20 lines Fix some issues with rwlock corruption that caused
+ deadlock like symptoms. When dvossel and I were doing some load
+ testing last week, we noticed that we could make Asterisk trunk
+ lock up instantly when we started generating a bunch of calls.
+ The backtraces of locked threads were bizarre, and many were
+ stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a
+ number of places where a backtrace would be loaded into an
+ invalid index of the backtrace array. It's an off by one error,
+ which ends up writing over the rwlock itself. 2) Ensure that in
+ the array of held locks, we NULL out an index once it is not
+ being used so that it's not confusing when analyzing its
+ contents. 3) Remove a bunch of logging referring to an rwlock
+ operating being done with "deep reentrancy". It is normal for
+ _many_ threads to hold a read lock on an rwlock. ........
+
+ * /, main/file.c: Merged revisions 184515 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 |
+ russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines
+ Don't act surprised if we get a -1 indication. ........
+
+ * include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26
+ Mar 2009) | 2 lines Pass more useful information through to lock
+ tracking when DEBUG_THREADS is on. ........
+
+2009-03-26 22:19 +0000 [r184451] Kevin P. Fleming <kpfleming at digium.com>
+
+ * sounds/Makefile, /: Merged revisions 184448 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar
+ 2009) | 9 lines Merged revisions 184447 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar
+ 2009) | 3 lines use new, improved 8kHz prompts ........
+ ................
+
+2009-03-26 21:18 +0000 [r184394] David Vossel <dvossel at digium.com>
+
+ * /, apps/app_test.c: Merged revisions 184389 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184389 | dvossel | 2009-03-26 16:09:37 -0500 (Thu, 26 Mar 2009)
+ | 14 lines Merged revisions 184388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009)
+ | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF
+ 8 app_test was failing when sending the last DTMF digit, 8,
+ because of the 100ms pause issued after DTMF is sent. During this
+ pause the other side would hang up causing the test to look like
+ it failed. Now the other side waits a second before hanging up.
+ (closes issue #12442) Reported by: tzafrir ........
+ ................
+
+2009-03-25 22:13 +0000 [r184325-184345] Russell Bryant <russell at digium.com>
+
+ * /, main/event.c: Merged revisions 184344 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 |
+ russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines
+ Remove unneeded AST_LIST_ENTRY() and comment on the purpose of
+ ast_event_ref. ........
+
+ * channels/chan_iax2.c, channels/chan_dahdi.c,
+ include/asterisk/event.h, channels/chan_skinny.c, res/ais/evt.c,
+ main/event.c, include/asterisk/strings.h, main/asterisk.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c,
+ channels/chan_unistim.c, include/asterisk/devicestate.h, /,
+ channels/chan_sip.c, main/devicestate.c,
+ include/asterisk/_private.h: Merged revisions 184339 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009)
+ | 35 lines Improve performance of the ast_event cache
+ functionality. This code comes from
+ svn/asterisk/team/russell/event_performance/. Here is a summary
+ of the changes that have been made, in order of both invasiveness
+ and performance impact, from smallest to largest. 1) Asterisk
+ 1.6.1 introduces some additional logic to be able to handle
+ distributed device state. This functionality comes at a cost. One
+ relatively minor change in this patch is that the extra
+ processing required for distributed device state is now
+ completely bypassed if it's not needed. 2) One of the things that
+ I noticed when profiling this code was that a _lot_ of time was
+ spent doing string comparisons. I changed the way strings are
+ represented in an event to include a hash value at the front. So,
+ before doing a string comparison, we do an integer comparison on
+ the hash. 3) Finally, the code that handles the event cache has
+ been re-written. I tried to do this in a such a way that it had
+ minimal impact on the API. I did have to change one API call,
+ though - ast_event_queue_and_cache(). However, the way it works
+ now is nicer, IMO. Each type of event that can be cached (MWI,
+ device state) has its own hash table and rules for hashing and
+ comparing objects. This by far made the biggest impact on
+ performance. For additional details regarding this code and how
+ it was tested, please see the review request. (closes issue
+ #14738) Reported by: russell Review:
+ http://reviewboard.digium.com/r/205/ ........
+
+ * /: add reviewboard:url property.
+
+2009-03-25 19:26 +0000 [r184282] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 |
+ file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix
+ issue with a T38 reinvite being sent even if not configured to do
+ so. If we receive a T38 request negotiate control frame we should
+ only attempt to do so if the option is enabled on the dialog.
+ ........
+
+2009-03-25 15:12 +0000 [r184223] Eliel C. Sardanons <eliels at gmail.com>
+
+ * main/asterisk.c, /: Merged revisions 184220 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) |
+ 19 lines Merged revisions 184188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) |
+ 13 lines Avoid destroying the CLI line when moving the cursor
+ backward and trying to autocomplete. When moving the cursor
+ backward and pressing TAB to autocomplete, a NULL is put in the
+ line and we are loosing what we have already wrote after the
+ actual cursor position. (closes issue #14373) Reported by: eliel
+ Patches: asterisk.c.patch uploaded by eliel (license 64) Tested
+ by: lmadsen ........ ................
+
+2009-03-25 01:55 +0000 [r184149] Russell Bryant <russell at digium.com>
+
+ * main/timing.c, utils/Makefile, /, include/asterisk/compat.h:
+ Merged revisions 184147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 |
+ russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines
+ Fix build issues on Mac OSX. (closes issue #14714) Reported by:
+ ygor ........
+
+2009-03-24 22:42 +0000 [r184081] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar
+ 2009) | 15 lines Merged revisions 184078 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar
+ 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero.
+ The 'digit' variable is guaranteed to be non-NULL, so the if
+ statement could never evaluate true. Changing to ast_strlen_zero
+ makes the logic correct. This was found while reviewing
+ ast_channel_ao2 code review. ........ ................
+
+2009-03-24 21:47 +0000 [r184039] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009)
+ | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low
+ and =medium The default codec configuration for chan_iax2 is
+ bandwidth=low. I noticed slin16 being negotiated as the codec in
+ some test calls, but that no longer happens after this change.
+ ........
+
+2009-03-24 15:28 +0000 [r183867-183916] Tilghman Lesher <tlesher at digium.com>
+
+ * /, configs/voicemail.conf.sample: Merged revisions 183914 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500
+ (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009)
+ | 3 lines Additionally note that the operator option needs an 'o'
+ extension. (Related to issue #14731) ........ ................
+
+ * /, main/http.c: Merged revisions 183865 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 |
+ tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines
+ Allow browsers to cache images and other static content. (This is
+ a regression over 1.4) ........
+
+2009-03-23 18:59 +0000 [r183768] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 183766 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar
+ 2009) | 13 lines Merged revisions 183700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar
+ 2009) | 7 lines Fix a memory leak in res_monitor.c The only way
+ that this leak would occur is if Monitor were started using the
+ Manager interface and no File: header were given. Discovered
+ while reviewing the ast_channel_ao2 review request. ........
+ ................
+
+2009-03-23 18:12 +0000 [r183703] Leif Madsen <lmadsen at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009)
+ | 7 lines Fixes a documentation error introduced during the CLI
+ cleanup at AstriDevCon 2008. (closes issue #14655) Reported by:
+ ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728)
+ Tested by: lmadsen ........
+
+2009-03-20 17:08 +0000 [r183563] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183560 | russell | 2009-03-20 12:00:58 -0500
+ (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009)
+ | 2 lines Fix a crash in IAX2 registration handling found during
+ load testing with dvossel. ........ ................
+
+2009-03-19 20:33 +0000 [r183438] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/features.h, apps/app_dial.c, /, main/features.c:
+ Merged revisions 183436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009)
+ | 13 lines Merged revisions 183386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009)
+ | 6 lines Cleaning up a few things in detect disconnect patch
+ Initialized ast_call_feature in detect_disconnect to avoid
+ accessing uninitialized memory. Cleaned up /param tags in
+ features.h. No longer send dynamic features in
+ ast_feature_detect. issue #11583 ........ ................
+
+2009-03-19 19:19 +0000 [r183333] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500
+ (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via
+ svnmerge from
[... 55221 lines stripped ...]
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