[svn-commits] lmadsen: tag 1.4.24-rc1 r180596 - /tags/1.4.24-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Mar 6 12:29:31 CST 2009
Author: lmadsen
Date: Fri Mar 6 12:29:24 2009
New Revision: 180596
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=180596
Log:
Importing files for 1.4.24-rc1 release
Added:
tags/1.4.24-rc1/.lastclean (with props)
tags/1.4.24-rc1/.version (with props)
tags/1.4.24-rc1/ChangeLog (with props)
Added: tags/1.4.24-rc1/.lastclean
URL: http://svn.digium.com/svn-view/asterisk/tags/1.4.24-rc1/.lastclean?view=auto&rev=180596
==============================================================================
--- tags/1.4.24-rc1/.lastclean (added)
+++ tags/1.4.24-rc1/.lastclean Fri Mar 6 12:29:24 2009
@@ -1,0 +1,1 @@
+33
Propchange: tags/1.4.24-rc1/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.4.24-rc1/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.4.24-rc1/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.4.24-rc1/.version
URL: http://svn.digium.com/svn-view/asterisk/tags/1.4.24-rc1/.version?view=auto&rev=180596
==============================================================================
--- tags/1.4.24-rc1/.version (added)
+++ tags/1.4.24-rc1/.version Fri Mar 6 12:29:24 2009
@@ -1,0 +1,1 @@
+1.4.24-rc1
Propchange: tags/1.4.24-rc1/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.4.24-rc1/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.4.24-rc1/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.4.24-rc1/ChangeLog
URL: http://svn.digium.com/svn-view/asterisk/tags/1.4.24-rc1/ChangeLog?view=auto&rev=180596
==============================================================================
--- tags/1.4.24-rc1/ChangeLog (added)
+++ tags/1.4.24-rc1/ChangeLog Fri Mar 6 12:29:24 2009
@@ -1,0 +1,23224 @@
+2009-03-06 Leif Madsen <lmadsen at digium.com>
+
+ * Released 1.4.24-rc1
+
+2009-03-06 18:23 +0000 [r180567] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Make compilation succeed in dev-mode when
+ IMAP storage is enabled.
+
+2009-03-06 17:19 +0000 [r180532] David Vossel <dvossel at digium.com>
+
+ * main/enum.c: Fix handling of backreferences for ENUM lookups
+ enum.c did not handle regex backtraces correctly. The '\1' in the
+ regex is a backreference that requires a pattern match to be
+ inserted. The way the code used to work is that it would find the
+ backreference and insert the entire input string minus the '+'.
+ This is incorrect. The regexec() function takes in a variable
+ called pmatch which is an array of structs containing the start
+ and end indexes for each backreference substring. The original
+ code actually passed the pmatch array pointer into regexec but
+ never did anything with it. Now when a backtrace is found, the
+ backtrace number is looked up in the pmatch array and the correct
+ substring is inserted. (closes issue #14576) Reported by:
+ chris-mac Review: http://reviewboard.digium.com/r/187/
+
+2009-03-05 23:26 +0000 [r180380-180464] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: [IMAP] Fix message retrieval issues when
+ identical mailbox names were defined in separate contexts. There
+ was a fix put in a while back so that an X-Asterisk-VM-Context
+ message header was added to stored IMAP voicemails. This would
+ allow for us to differentiate if the same mailbox name was used
+ in multiple contexts. The problem still left was that not all
+ places where messages were retrieved actually attempted to use
+ this header for information when retrieving messages. This commit
+ fixes that so that MWI and message retrieval from VoiceMailMain
+ work as expected. (closes issue #13853) Reported by: vicks1
+ Patches: 13853_v2.patch uploaded by mmichelson (license 60)
+ Tested by: lmadsen
+
+ * apps/app_voicemail.c, configs/voicemail.conf.sample: Fix broken
+ mailbox parsing when searchcontexts option is enabled. When using
+ the searchcontexts option in voicemail.conf, the code made the
+ assumption that all mailbox names defined were unique across all
+ contexts. However, the code did nothing to actually enforce this
+ assumption, nor did it do anything to alert a user that he may
+ have created an ambiguity in his voicemail.conf file by defining
+ the same mailbox name in multiple contexts. With this change, we
+ now will issue a nice long warning if searchcontexts is on and we
+ encounter the same mailbox name in multiple contexts and ignore
+ any duplicates after the first box. Whether searchcontexts is
+ enabled or not, if we come across a duplicate mailbox in the same
+ context, then we will issue a warning and ignore the duplicated
+ mailbox. I have also added a small note to voicemail.conf.sample
+ in the explanation for searchcontexts explaining that you cannot
+ define the same mailbox in multiple contexts if you have enabled
+ the option. (closes issue #14599) Reported by: lmadsen Patches:
+ 14599.patch uploaded by mmichelson (license 60) (with slight
+ modification) Tested by: lmadsen
+
+2009-03-05 18:22 +0000 [r180372] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/rtp.c, main/frame.c, include/asterisk/frame.h: Fix problems
+ when RTP packet frame size is changed During some code analysis,
+ I found that calling ast_rtp_codec_setpref() on an ast_rtp
+ session does not work as expected; it does not adjust the
+ smoother that may on the RTP session, in fact it summarily drops
+ it, even if it has data in it, even if the current format's
+ framing size has not changed. This is not good. This patch
+ changes this behavior, so that if the packetization size for the
+ current format changes, any existing smoother is safely updated
+ to use the new size, and if no smoother was present, one is
+ created. A new API call for smoothers,
+ ast_smoother_reconfigure(), was required to implement these
+ changes. Review: http://reviewboard.digium.com/r/184/
+
+2009-03-04 19:22 +0000 [r180194] Joshua Colp <jcolp at digium.com>
+
+ * main/callerid.c: Look for the number in a callerid string
+ starting from the end. This way a value using <> can exist in the
+ name portion. (issue #AST-194)
+
+2009-03-03 23:01 +0000 [r180010] Jason Parker <jparker at digium.com>
+
+ * channels/chan_dahdi.c: Make sure we still support zapchan in
+ users.conf, in addition to dahdichan.
+
+2009-03-03 22:48 +0000 [r180006] Mark Michelson <mmichelson at digium.com>
+
+ * configs/queues.conf.sample, apps/app_queue.c: Clarify some
+ documentation of queues.conf.sample It had always been possible
+ to explicitly specify a "blank" value for a sound file in
+ queues.conf and have no sound played back. The problem with this
+ is that it would result in some ugly CLI warnings from file.c.
+ This commit introduces a check when playing a file in app_queue
+ to see if the name of the file is zero-length and return early if
+ that is the case. Also, the ability to specify the blank sound
+ files in queues.conf is now mentioned more clearly in
+ queues.conf.sample (closes issue #14227) Reported by: caspy
+
+2009-03-03 18:27 +0000 [r179840] Joshua Colp <jcolp at digium.com>
+
+ * res/res_features.c: Do not assume that the bridge_cdr is still
+ attached to the channel when the 'h' exten is finished executing.
+ It is possible for a masquerade operation to occur when the 'h'
+ exten is operating. This operation moves the CDR records around
+ causing the bridge_cdr to no longer exist on the channel where it
+ is expected to. We can not safely modify it afterwards because of
+ this, so don't even try. (closes issue #14564) Reported by: meric
+
+2009-03-03 18:11 +0000 [r179807] Steve Murphy <murf at digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile,
+ utils/expr2.testinput, main/ast_expr2.h, main/ast_expr2.y,
+ main/ast_expr2f.c: These changes allow AEL to better check ${}
+ constructs within $[...], that are concatenated with text. I
+ modified and added rules in ast_expr2.fl to better handle the
+ concatenations. I added some default routines to ast_expr2.y so
+ the standalone would compile. It also looks like I haven't run
+ this thru bison since 2.1, so it's good to get this updated. The
+ Makefile has comments added now for check_expr2 and check_expr to
+ explain what they are for, and how to run them. The testexpr2s
+ stuff has been removed, in favor of check_expr2. expr2.testinput
+ has been updated to include the two expressions that inspired
+ these changes (from mcnobody on #asterisk this morning) The
+ regression has been run and all looks well.
+
+2009-03-03 16:45 +0000 [r179741] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Ensure chan->fdno always gets reset to -1 after
+ handling a channel fd event. Since setting fdno to -1 had to be
+ moved, a couple of other code paths that do process an fd event
+ return early and do not pass through the code path where it was
+ moved to. So, set it to -1 in a few other places, too.
+
+2009-03-03 14:38 +0000 [r179671] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c: Move where fdno is set to the default value to
+ *after* the read callback of the channel driver is called. We
+ have to do this as the underlying channel driver may need the
+ fdno value to determine what to read.
+
+2009-03-03 13:53 +0000 [r179608] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Make it easier to detect an improper call to
+ ast_read(). When you call ast_waitfor() on a channel, the index
+ into the channel fds array that holds the file descriptor that
+ poll() determines has input available is stored in fdno. This
+ patch clears out this value after a call to ast_read() and also
+ reports errors if ast_read() is called without an fdno set. From
+ a discussion on the asterisk-dev list.
+
+2009-03-02 23:54 +0000 [r179536] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c: Fix bridging regression from commit 176701 This
+ fixes a bad regression where the bridge would exit after an
+ attended transfer was made. The problem was due to nexteventts
+ getting set after the masquerade which caused the bridge to
+ return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
+ tim_ringenbach
+
+2009-03-02 23:34 +0000 [r179532] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: Move ast_waitfor() down to avoid the results
+ of the API call becoming stale. This call to ast_waitfor() was
+ being done way too soon in this section of code. Specifically,
+ there was code in between the call to waitfor and the code that
+ uses the result that puts the channel in autoservice. By putting
+ the channel in autoservice, the previous results of ast_waitfor()
+ become meaningless, as the autoservice thread will do it's own
+ ast_waitfor() and ast_read() on the channel. So, when we came
+ back out of autoservice and eventually hit the block of code that
+ calls ast_read() on the channel, there may not actually be any
+ input on the channel available. Even though the previous call to
+ ast_waitfor() in app_meetme said there was input, the autoservice
+ thread has since serviced the channel for some period of time.
+ This bug manifested itself while dvossel was doing some testing
+ of MeetMe in Asterisk trunk. He was using the timerfd timing
+ module. When the code hit ast_read() erroneously, it determined
+ that it must have been called because of input on the timer fd,
+ as chan->fdno was set to AST_TIMING_FD, since that was the cause
+ of the last legitimate call to ast_read() done by autoservice. In
+ this test, an IAX2 channel was calling into the MeetMe
+ conference. It was _much_ more likely to be seen with an IAX2
+ channel because of the way audio is handled. Every audio frame
+ that comes in results in a call to ast_queue_frame(), which then
+ uses ast_timer_enable_continuous() to notify the channel thread
+ that a frame is waiting to be handled. So, the chances of
+ ast_waitfor() indicating that a channel needs servicing due to a
+ timer event on an IAX2 event is very high. Finally, it is
+ interesting to note that if a different timing interface was
+ being used, this bug would probably not be noticed. When
+ ast_read() is called and erroneously thinks that there is a timer
+ event to handle, it calls the ast_timer_ack() function. The
+ pthread and dahdi timing modules handle the ack() function being
+ called when there is no event by simply ignoring it. In the case
+ of the timerfd module, it results in a read() on the timer fd
+ that will block forever, as there is no data to read. This caused
+ Asterisk to lock up very quickly. Thanks to dvossel and
+ mmichelson for the fun debugging session. :-)
+
+2009-03-02 23:09 +0000 [r179468] Tilghman Lesher <tlesher at digium.com>
+
+ * main/app.c: When ending a recording with silence detection,
+ remember to reduce the duration. The end of the recording is
+ correspondingly trimmed, but the duration was not trimmed by the
+ number of seconds trimmed, so the saved duration was necessarily
+ longer than the actual soundfile duration. (closes issue #14406)
+ Reported by: sasargen Patches: 20090226__bug14406.diff.txt
+ uploaded by tilghman (license 14) Tested by: sasargen
+
+2009-03-02 22:58 +0000 [r179461] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Ensure that only one thread is calling
+ ast_settimeout() on a channel at a time. For example, with an
+ IAX2 channel, you can have both the channel thread and the
+ chan_iax2 processing threads calling this function, and doing so
+ twice at the same time is a bad thing. (Found in a debugging
+ session with dvossel and mmichelson)
+
+2009-03-02 20:14 +0000 [r179395] Jason Parker <jparker at digium.com>
+
+ * main/editline/configure, main/editline/np/unvis.c,
+ main/editline/sys.h, main/editline/configure.in: Remove several
+ silly warnings in editline. One about a broken preprocessor
+ directive, and another about strlcpy/strlcat. (closes issue
+ #14264) Reported by: dimas
+
+2009-02-27 19:03 +0000 [r179056] Jason Parker <jparker at digium.com>
+
+ * doc/channelvariables.txt: Update documentation for DIALEDTIME and
+ ANSWEREDTIME variables. (closes issue #14566) Reported by:
+ klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
+ klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
+ klaus3000 (license 65)
+
+2009-02-26 21:27 +0000 [r178956] Steve Murphy <murf at digium.com>
+
+ * configs/features.conf.sample, res/res_features.c: This change
+ moves the default feature digit timeout to 1000 ms from the
+ previous default of 500. As per bug 14515, a dev discussion
+ arrived at a "mediated concensus" of a default feature digit
+ timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for
+ distracted phone users in phone booths; kpfleming put his foot
+ down at 1.0 sec. Users who found the previous default max delay
+ of 250 msec perfect, are welcome to override the new default.
+ Notice that I said that 250 msec was the default; wait a minute,
+ you might say, the config file said it was 500 msec!; well,
+ because of the bug fix for 14515, we found that 500 msec was
+ actually enforcing a max of 250. The bug fix would restore 500
+ msec, but we felt even that was a bit tight for most users...
+ 2000 msec was pushed earlier by mmichelson, so that reduces to
+ 1000 msec after the bug fix. Enjoy!
+
+2009-02-26 17:24 +0000 [r178838] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: IAX2 prune realtime fix Now prune_users()
+ and prune_peers() are called instead of reload_config() to prune
+ all users/peers that are realtime. These functions remove all
+ users/peers with the rtfriend and delme flags set.
+ iax2_prune_realtime() also lacked the code to properly delete a
+ single friend. For example. if iax2 prune realtime <friend> was
+ called, only the peer instance would be removed. The user would
+ still remain. (closes issue #14479) Reported by: mousepad99
+ Review: http://reviewboard.digium.com/r/176/
+
+2009-02-26 17:09 +0000 [r178640-178804] Steve Murphy <murf at digium.com>
+
+ * res/res_features.c: This patch prevents the feature detection
+ timeout from being cut in half. Because the ast_channel_bridge()
+ call will return 0 and pass a frame pointer for both DTMF_BEGIN
+ and DTMF_END, the feature_timer field in hte config struct is
+ getting decremented twice, which effectively cuts the
+ digittimeout in half. I added conditions to the if statement to
+ only let DTMF_END frames to flow thru, which solved the problem.
+ Also, when the frame pointer is null, let control flow thru--
+ this usually happens on timeouts. I added a comment to the code
+ to explain what's going on and why. Many thanks to sodom for
+ reporting this problem. Personnally, it always seemed like
+ something was wrong with the featuredigittimeout, but I never
+ could quite decide what... and was too busy to investigate. This
+ bug forced the issue, and now we know. Sodom had other issues in
+ 14515, but I couldn't reproduce them. If he still has problems,
+ and wants to get them solved, he is welcome to reopen 14515.
+ (closes issue #14515) Reported by: sodom Patches: 14515.patch
+ uploaded by murf (license 17) Tested by: murf, sodom
+
+ * main/ast_expr2.fl, main/ast_expr2f.c: This patch completes the
+ fixes nec. to make 1.4 asterisk dialplan expressions ($[...])
+ 8-bit transparent While I was updating ast_expr2.fl, I missed one
+ rule that would allow 8-bit chars to be caught in tokens; and in
+ so doing, it absorbs the ${ sequence and messes up the checking
+ of raw exprs by AEL. Trunk already has these changes. (closes
+ issue #14543) Reported by: klaus3000 Patches: patch.14543
+ uploaded by murf (license 17) Tested by: murf
+
+2009-02-25 12:43 +0000 [r178508] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c: Update the copyright year for the main page of
+ the doxygen documentation.
+
+2009-02-24 23:25 +0000 [r178445] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/extensions.conf.sample: Add section about the #exec
+ command in configuration files. (closes issue #14540) Reported
+ by: jtodd Patch by: jtodd, with additional notes by tilghman
+ (license 14)
+
+2009-02-24 20:36 +0000 [r178373] Russell Bryant <russell at digium.com>
+
+ * main/rtp.c: Only set dtmfcount on BEGIN, and ensure it gets reset
+ to 0 properly. (issue #14460) Reported by: moliveras Tested by:
+ russell
+
+2009-02-24 17:02 +0000 [r178266] Terry Wilson <twilson at digium.com>
+
+ * apps/app_dahdiras.c, res/res_musiconhold.c: Change include order
+ to make compile on Centos 5 with DAHDI If BIT_TYPES_DEFINED gets
+ defined before linux/types.h is included, the __s32 type doesn't
+ get defined
+
+2009-02-24 15:16 +0000 [r178205] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Skip check for extension when subscribing
+ for MWI. Since the remote side is not actually subscribing to a
+ specific extension when subscribing for MWI just skip the check
+ to see if the extension exists. They can't use it to specify the
+ mailbox either since we require configuration of that in sip.conf
+ (closes issue #14531) Reported by: festr
+
+2009-02-23 23:09 +0000 [r178141] Russell Bryant <russell at digium.com>
+
+ * main/rtp.c: Fix infinite DTMF when a BEGIN is received without an
+ END. This commit is related to rev 175124 of 1.4 where a previous
+ attempt was made to fix this problem. The problem with the
+ previous patch was that the inserted code needed to go _before_
+ setting the lastrxts to the current timestamp. Because those were
+ the same, the dtmfcount variable was never decremented, and so
+ the END was never sent. In passing, I removed the dtmfsamples
+ variable which was completed unused. I also removed a redundant
+ setting of the lastrxts variable. (closes issue #14460) Reported
+ by: moliveras
+
+2009-02-20 22:59 +0000 [r177701-177786] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c: Don't print the CR-NL combination when we aren't
+ outputting to the manager. An embedded CR-NL in a CLI command
+ screws up several AMI parsers that don't expect to see that
+ combination in the middle of output. (Closes issue #14305)
+ Reported by: martins Patch by: tilghman
+
+ * include/asterisk/threadstorage.h: This exception does not appear
+ to still be true for Solaris 10, and OpenSolaris definitely needs
+ it to be removed. Fixed for snuff-home on -dev channel.
+
+2009-02-20 20:17 +0000 [r177696] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with
+ undefined audio codecs in chan_iax2 During iax2 call negotiation,
+ supported codecs are passed in an Information Element containing
+ a 2 byte field where each bit correlates to a specific codec. In
+ 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15
+ undefined. By default all bits are enabled unless specified
+ otherwise. Since its a 2 byte field and 13-15 are not defined,
+ these bits are never turned off. In trunk, bits 13-15 are
+ defined, which means 1.4 is advertising support for codecs it
+ does not have when talking to trunk. I fixed this by adding
+ #define for undefined audio codec bits. These bits are then
+ removed from iax2's full bandwidth capabilities. (closes issue
+ #14283) Reported by: jcovert
+
+2009-02-19 22:51 +0000 [r177540] Steve Murphy <murf at digium.com>
+
+ * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This patch
+ fixes a problem with 8-bit input to the ast_expr2 scanner. The
+ real culprit was the --full argument to flex in the Makefile!
+ This causes a 7-bit scanner to be generated. I reviewed the rules
+ and found one rule where I needed to specifically include 8-bit
+ chars for a token. I tested against the text supplied by ibercom,
+ and all looks very well. This has been there a surprisingly long
+ time! (closes issue #14498) Reported by: ibercom Patches:
+ 14498.patch uploaded by murf (license 17) Tested by: murf
+
+2009-02-19 22:26 +0000 [r177536] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Fix up potential crashes, by reducing the
+ sharing between interactive and non-interactive threads. (closes
+ issue #14253) Reported by: Skavin Patches:
+ 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Skavin
+
+2009-02-19 18:58 +0000 [r177450] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Force a MWI notification after subscribe
+ request. Reported by the Resiprocate dev team. Thanks!
+
+2009-02-19 16:37 +0000 [r177383] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_speech_utils.c: If we are able to create a speech
+ structure unset the ERROR variable in case it was previously set.
+ (issue #LUMENVOX-13)
+
+2009-02-18 22:43 +0000 [r177225] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael.tab.c, pbx/ael/ael.y: This patch fixes a regression
+ of sorts that was introduced in rev 24425. It basically fixes
+ AST-190/ABE-1782. What was wrong: the user has 6000 extensions in
+ one context; and then 6000 contexts, one per extension. The
+ parser could only handle about 4893 of the 6000 extens in the
+ single context. This was due to the regression I mentioned. To
+ get rid of shift/reduce conflicts, Luigi set up right-recursive
+ lists for globals, context elements, switch lists, and
+ statements. Right recursive lists got rid of the warnings, but
+ instead, they use up a tremendous amount of stack space when the
+ lists are long. I saw this a few years back, and resolved not to
+ fix it until someone complained. That day has arrived! After the
+ changes were made, I ran the regression test suite, and there
+ were no problems. I took the test case the user provided, and
+ added 100,000 extensions to the single context, that already had
+ 6,000 extens in it. (I'll see your 6, and raise you 100!) It
+ takes a few minutes to read it all in, check it and generate code
+ for it, but no problems. So, I think I can say that
+ fundamentally, there are no longer any limits on the number of
+ items you can place in contexts, statement blocks, switches, or
+ globals, beyond your virt mem constraints.
+
+2009-02-18 20:06 +0000 [r177160] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/h323/cisco-h225.cxx, channels/h323/compat_h323.cxx,
+ autoconf/ast_check_pwlib.m4, channels/h323/cisco-h225.h,
+ channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx,
+ channels/h323/ast_ptlib.h (added), configure,
+ channels/h323/compat_h323.h, configure.ac,
+ channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
+ channels/h323/ast_h323.h, channels/h323/chan_h323.h: Modify h323
+ to build against PTLib as well as the older PWLib Several changes
+ in PTLib have occurred requiring build time detection. Changes
+ accounted for include the library name change, config option
+ change, install location change, and a boolean type change which
+ is handled by ast_ptlib.h. Also, the sed check has been modified
+ to properly work with autoconf >= 2.62. (closes issue #14224)
+ Reported by: bergolth Patches: asterisk-autoconf-sed.patch
+ uploaded by bergolth (license 661) asterisk-pwlib-v3.patch
+ uploaded by bergolth (license 661) Tested by: jpeeler
+
+2009-02-18 18:30 +0000 [r177096] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/config.h: Document the return value of the
+ update method (as requested on -dev list)
+
+2009-02-18 17:41 +0000 [r176945-177039] Doug Bailey <dbailey at digium.com>
+
+ * main/utils.c: Merged revisions 177035 manually from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 |
+ dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines
+ Fixed error where a check for an zero length, terminated string
+ was needed. ........
+
+ * main/utils.c: Need to take into account the \0 terminator of the
+ old string to determine the amount available.
+
+2009-02-18 00:34 +0000 [r176810] Shaun Ruffell <sruffell at digium.com>
+
+ * codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice
+ with G723. This commit brings in the changes that were living out
+ on the svn/asterisk/team/sruffell/asterisk-1.4-transcoder branch.
+ codec_dahdi.c now always uses signed linear as the simple codec
+ so that a soft g729 codec will not end up being preferred to the
+ hardware codec. There are also changes to allow codec_dahdi.c to
+ feed packets to the hardware in the native sample size of the
+ codec. This solves problems with choppy audio when using G723.
+
+2009-02-17 21:54 +0000 [r176701] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, res/res_features.c, include/asterisk/channel.h:
+ Modify bridging to properly evaluate DTMF after first warning is
+ played The main problem is currently if the Dial flag L is used
+ with a warning sound, DTMF is not evaluated after the first
+ warning sound. To fix this, a flag has been added in
+ ast_generic_bridge for playing the warning which ensures that if
+ a scheduled warning is missed, multiple warrnings are not played
+ back (due to a feature evaluation or waiting for digits).
+ ast_channel_bridge was modified to store the nexteventts in the
+ ast_bridge_config structure as that information was lost every
+ time ast_channel_bridge was reentered, causing a hangup due to
+ incorrect time calculations. (closes issue #14315) Reported by:
+ tim_ringenbach Reviewed on reviewboard:
+ http://reviewboard.digium.com/r/163/
+
+2009-02-17 21:21 +0000 [r176426-176661] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_local.c: Backport change to 1.4: Prior to
+ masquerade, move the group definitions to the channel performing
+ the masq, so that the group count lingers past the bridge.
+ (closes issue #14275) Reported by: kowalma Patches:
+ 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: kowalma
+
+ * channels/chan_sip.c: After a 'sip reload', qualifies for realtime
+ peers weren't immediately restarted, instead waiting until the
+ next registration. We're now caching the qualify across a
+ reload/restart and starting the qualify immediately upon loading
+ the peer. (closes issue #14196) Reported by: pdf Patches:
+ 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
+ Tested by: pdf
+
+2009-02-16 23:30 +0000 [r176354] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE not
+ being relayed correctly during bridging This should have been
+ committed with rev176247, but I missed it. srcupdate frames no
+ longer break out of the native bridge, but are not being sent to
+ the other call leg either. This fixs that. issue #13749
+
+2009-02-16 21:41 +0000 [r176254] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/utils.c: correct a logic error in the last stringfields
+ commit... don't mark additional space as allocated if the string
+ was built using already-allocated space
+
+2009-02-16 21:39 +0000 [r176249-176252] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_meetme.c: Remove unused variable and make dev-mode
+ compilation happy
+
+ * apps/app_meetme.c: Open the DAHDI pseudo device and set it to be
+ nonblocking atomically Apparently on FreeBSD, attempting to set
+ the O_NONBLOCKING flag separately from opening the file was
+ causing an "inappropriate ioctl for device" error. While I cannot
+ fathom why this would be happening, I certainly am not opposed to
+ making the code a bit more compact/efficient if it also fixes a
+ bug. (closes issue #14482) Reported by: ys Patches: meetme.patch
+ uploaded by ys (license 281) Tested by: ys
+
+2009-02-16 21:28 +0000 [r176247] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE
+ breaking out of native bridge In iax2, when a
+ AST_CONTROL_SRCUPDATE is received it breaks from the native
+ bridge, but since there is no code path to handle srcupdate it
+ just goes to be beginning of the loop. This was causing packet
+ storms of srcupdate frames between servers. Now srcupdate frames
+ do not break the native bridge for processing. (closes issue
+ #13749) Reported by: adiemus
+
+2009-02-16 21:10 +0000 [r176216] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/utils.c: fix a flaw in the ast_string_field_build() family
+ of API calls; these functions made no attempt to reuse the space
+ already allocated to a field, so every time the field was written
+ it would allocate new space, leading to what appeared to be a
+ memory leak.
+
+2009-02-16 15:33 +0000 [r176029] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Don't have the Via header stored as a
+ stringfield as it can change often during the lifetime of a
+ dialog. This issue crept up with subscriptions on the AA50. When
+ an outgoing NOTIFY is sent a new branch value is created and the
+ Via header is changed to reflect it. Since this was a stringfield
+ a new spot in the pool was used for the value while the old was
+ left untouched/unused. If the current pool was full a new pool
+ was created. This would cause memory usage to increase steadily.
+ (issue #AA50-2332)
+
+2009-02-15 23:37 +0000 [r175921] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/pbx.c, channels/chan_sip.c, main/devicestate.c,
+ include/asterisk/manager.h: fix mis-spelling of the word
+ registered. Reported by De_Mon on #asterisk-dev.
+
+2009-02-15 20:33 +0000 [r175777-175825] Olle Johansson <oej at edvina.net>
+
+ * formats/format_ilbc.c: format_ilbc does not depend on codec
+ libraries and can therefore always be made. My mistake. Ursäkta!
+
+ * formats/format_ilbc.c: Disable format_ilbc.so by default, like
+ codec_ilbc.so
+
+ * channels/chan_sip.c: Make sure that the debug line is not printed
+ on debug level 0
+
+2009-02-13 21:53 +0000 [r175698] Jason Parker <jparker at digium.com>
+
+ * include/asterisk/dahdi_compat.h: Zaptel is not DAHDI. Rather,
+ Zaptel is actually Zaptel. (in case you're confused, DAHDI is
+ still DAHDI)
+
+2009-02-13 19:47 +0000 [r175407-175590] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Fix a potential crash situation when using
+ IMAP voicemail If calling into VoiceMailMain when using IMAP
+ storage, it was possible to crash Asterisk by hanging up the
+ phone when prompted for a voicemail mailbox. This patch fixes the
+ issue. While it may appear that this patch is superficial, it
+ allows code execution to continue to the failure case just below
+ the IMAP_STORAGE code block where this patch has been applied
+ (closes issue #14473) Reported by: dwpaul Patches:
+ voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license
+ 689)
+
+ * main/file.c: Fix a place where filestreams were not refcounted
+ properly This section was already present in trunk and other
+ branches, but did not exist in 1.4. (closes issue #14395)
+ Reported by: ZX81 Patches: 14395.patch uploaded by putnopvut
+ (license 60) Tested by: ZX81
+
+2009-02-12 21:19 +0000 [r175311] Tilghman Lesher <tlesher at digium.com>
+
+ * main/udptl.c: Fix crashes when receiving certain T.38 packets.
+ Also, increase the maximum size of T.38 packets and warn users
+ when they try to set the limits above those maximums. (closes
+ issue #13050) Reported by: schern Patches:
+ 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: schern
+
+2009-02-12 20:34 +0000 [r175187-175294] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_features.c: Fix ParkedCall event information for From
+ field in the case of a blind transfer If the parker information
+ can not be obtained from the peer, try and see if the
+ BLINDTRANSFER channel variable has been set. Previously, a blind
+ transfer to the ParkAndAnnounce app would return nothing for the
+ From. Closes AST-189
+
+ * res/res_features.c: Fix crash in event of failed attempt to
+ transfer to parking The peer may not necessarily exist, such as
+ in the case of a transfer to ParkAndAnnounce. In this case don't
+ try to play a sound to it.
+
+2009-02-12 16:51 +0000 [r175124] Russell Bryant <russell at digium.com>
+
+ * main/rtp.c: Don't send DTMF for infinite time if we do not
+ receive an END event. I thought that this was going to end up
+ being a pretty gnarly fix, but it turns out that there was
+ actually already a configuration option in rtp.conf, dtmftimeout,
+ that was intended to handle this situation. However, in between
+ Asterisk 1.2 and Asterisk 1.4, the code that processed the option
+ got lost. So, this commit brings it back to life. The default
+ timeout is 3 seconds. However, it is worth noting that having
+ this be configurable at all is not really the recommended
+ behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the
+ time period of extending the tone is necessary to avoid that a
+ tone "gets stuck". Regardless of the algorithm used, the tone
+ SHOULD NOT be extended by more than three packet interarrival
+ times. A slight extension of tone durations and shortening of
+ pauses is generally harmless. Three seconds will pretty much
+ _always_ be far more than three packet interarrival times.
+ However, that behavior is not required, so I'm going to leave it
+ with our legacy behavior for now. Code from
+ svn/asterisk/team/russell/issue_14460 (closes issue #14460)
+ Reported by: moliveras
+
+2009-02-12 10:16 +0000 [r175029] Philippe Sultan <philippe.sultan at gmail.com>
+
+ * channels/chan_gtalk.c: Set the initiator attribute to lowercase
+ in our replies when receiving calls. This attribute contains a
+ JID that identifies the initiator of the GoogleTalk voice
+ session. The GoogleTalk client discards Asterisk's replies if the
+ initiator attribute contains uppercase characters. (closes issue
+ #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded
+ by jcovert (license 551) Tested by: jcovert
+
+2009-02-12 00:19 +0000 [r174997] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Revert RTP changes for continuation of DTMF. Proxy
+ commit by russell via SMS.
+
+2009-02-12 00:01 +0000 [r174985-174986] Russell Bryant <russell at digium.com>
+
+ * main/rtp.c: Clear out the current event after forcing the end of
+ a digit
+
+ * main/rtp.c: Fixify infinite DTMF in the case that no RFC2833 END
+ event is ever received
+
+2009-02-11 20:54 +0000 [r174885] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c, apps/app_macro.c: Restore a behavior that was
+ recently changed, when we fixed issue #13962 and issue #13363
+ (related to issue #6176). When a hangup occurs during a Macro
+ execution in earlier 1.4, the h extension would execute within
+ the Macro context, whereas it was always supposed to execute only
+ within the main context (where Macro was called). So this fix
+ checks for an "h" extension in the deepest macro context where a
+ hangup occurred; if it exists, that "h" extension executes,
+ otherwise the main context "h" is executed. (closes issue #14122)
+ Reported by: wetwired Patches: 20090210__bug14122.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: andrew
+
+2009-02-10 18:50 +0000 [r174644] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Go off hold when we get an empty reinvite
+ telling us to. (closes issue #14448) Reported by: frawd Patches:
+ hold_invite_nosdp.patch uploaded by frawd (license 610)
+
+2009-02-10 17:52 +0000 [r174583] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/jitterbuf.c: Improve behavior of jitterbuffer when
+ maxjitterbuffer is set. This change improves the way the
+ jitterbuffer handles maxjitterbuffer and dramatically reduces the
+ number of frames dropped when maxjitterbuffer is exceeded. In the
+ previous jitterbuffer, when maxjitterbuffer was exceeded, all new
+ frames were dropped until the jitterbuffer is empty. This change
+ modifies the code to only drop frames until maxjitterbuffer is no
+ longer exceeded. Also, previously when maxjitterbuffer was
+ exceeded, dropped frames were not tracked causing stats for
+ dropped frames to be incorrect, this change also addresses that
+ problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
+ by mnicholson (license 96) Tested by: mnicholson Review:
+ http://reviewboard.digium.com/r/144/
+
+2009-02-10 02:27 +0000 [r174369] Steve Murphy <murf at digium.com>
+
+ * apps/app_rpt.c: This patch solves some compiler complaints in
+ both 32 and 64-bit environments.
+
+2009-02-09 17:11 +0000 [r174282] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Don't do an SRV lookup if a port is
+ specified RFC 3263 says to do A record lookups on a hostname if a
+ port has been specified, so that's what we're going to do. See
+ section 4.2. (closes issue #14419) Reported by: klaus3000
+ Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by
+ klaus3000 (license 65)
+
+2009-02-09 14:48 +0000 [r174218] Joshua Colp <jcolp at digium.com>
+
+ * res/res_musiconhold.c: Don't overwrite our pointer to the music
[... 22508 lines stripped ...]
More information about the svn-commits
mailing list