[svn-commits] russell: tag 1.4.23-rc3 r165382 - /tags/1.4.23-rc3/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Dec 17 16:07:56 CST 2008
Author: russell
Date: Wed Dec 17 16:07:55 2008
New Revision: 165382
URL: http://svn.digium.com/view/asterisk?view=rev&rev=165382
Log:
Importing files for 1.4.23-rc3 release
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tags/1.4.23-rc3/.version (with props)
tags/1.4.23-rc3/ChangeLog (with props)
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+2008-12-17 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.23-rc3 released.
+
+2008-12-17 21:14 +0000 [r165317] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_macro.c: Reverse the fix from issue #6176 and add proper
+ handling for that issue. (Closes issue #13962, closes issue
+ #13363) Fixed by myself (license 14)
+
+2008-12-17 20:51 +0000 [r164977-165255] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_meetme.c, apps/app_realtime.c, apps/app_directory.c,
+ apps/app_queue.c: Fix some memory leaks found while looking at
+ how realtime configs are handled. Also cleaned up some coding
+ guidelines violations in app_realtime.c, mostly related to
+ spacing
+
+ * channels/chan_sip.c: After looking through SIP registration code
+ most of the day, this is one of the few things I could find that
+ was just plain wrong. Even though it probably isn't possible for
+ it to happen, it seems weird to have code that checks if a
+ pointer is NULL and then immediately dereferences that pointer if
+ it was NULL.
+
+2008-12-16 21:38 +0000 [r164672-164881] Russell Bryant <russell at digium.com>
+
+ * main/utils.c: Fix an issue where DEBUG_THREADS may erroneously
+ report that a thread is exiting while holding a lock. If the last
+ lock attempt was a trylock, and it failed, it will still be in
+ the list of locks so that it can be reported. (closes issue
+ #13219) Reported by: pj
+
+ * apps/app_macro.c: Do not dereference the channel if
+ AST_PBX_KEEPALIVE has been returned. This is a bug I noticed
+ while looking at the code for app_macro. This return code means
+ that another thread has assumed ownership of the channel and it
+ can no longer be touched. (I hate this return code with a
+ passion, by the way.)
+
+ * main/manager.c: Add "restart gracefully" to the AMI blacklist of
+ CLI commands. "module unload" was already identified as a command
+ that can not be used from the AMI. "restart gracefully"
+ effectively unloads all modules, and will run in to the same
+ problems. (closes issue #13894) Reported by: kernelsensei
+
+ * include/asterisk/threadstorage.h, main/threadstorage.c: Fix
+ memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
+ One issue was that the ast_mutex_* API was being used within the
+ context of the thread local data destructors. We would go off and
+ allocate more thread local data while the pthread lib was in the
+ middle of destroying it all. This led to a memory leak. Another
+ issue was an invalid argument being provided to the the
+ object_add API call. (closes issue #13678) Reported by: ys Tested
+ by: Russell
+
+ * channels/chan_sip.c: Fix a memory leak related to the use of the
+ "setvar" configuration option. The problem was that these
+ variables were being appended to the list of vars on the sip_pvt
+ every time a re-registration or re-subscription came in. Since
+ it's just a waste of memory to put them there unless the request
+ was an INVITE, then the fix is to check the request type before
+ copying the vars. (closes issue #14037) Reported by: marvinek
+ Tested by: russell
+
+2008-12-16 15:15 +0000 [r164634] Steve Murphy <murf at digium.com>
+
+ * main/pbx.c: I added a sentence to clarify why - and ' ' are
+ ignored in patterns as per bug 14076. Leif says he'll put some
+ stuff about it in the extensions.conf sample, etc.
+
+2008-12-16 14:28 +0000 [r164605] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: Don't try to change working directory if a
+ directory was not configured. (closes issue #14089) Reported by:
+ caspy
+
+2008-12-15 19:53 +0000 [r164416-164422] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/pbx.h: Add the deadlock note to
+ ast_spawn_extension as well
+
+ * include/asterisk/channel.h, include/asterisk/pbx.h: Add notes to
+ autoservice and pbx doxygen regarding a potential deadlock
+ scenario so that it is avoided in the future
+
+2008-12-15 18:11 +0000 [r164204-164350] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Do not try to unlock a non-existant channel
+ if the transfer fails. (closes issue #13800) Reported by: dwagner
+ Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety
+ (license 608)
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/channel.h: Use autoconf logic to determine
+ whether the system has timersub or not. Do not blindly assume
+ Solaris does not. (closes issue #13838) Reported by: ano
+
+ * apps/app_dial.c: Can we try not to assign an unsigned int to -1?
+ (closes issue #14074) Reported by: wetwired
+
+2008-12-15 14:31 +0000 [r164201] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, res/res_features.c: Handle a case where a call
+ can be bridged to a channel that is still ringing. The issue that
+ was reported was about a case where a RINGING channel got
+ redirected to an extension to pick up a call from parking. Once
+ the parked call got taken out of parking, it heard silence until
+ the other side answered. Ideally, the caller that was parked
+ would get a ringing indication. This patch fixes this case so
+ that the caller receives ringback once it comes out of parking
+ until the other side answers. The fixes are: - Make sure we
+ remember that a channel was an outgoing channel when doing a
+ masquerade. This prevents an erroneous ast_answer() call on the
+ channel, which causes a bogus 200 OK to be sent in the case of
+ SIP. - Add some additional comments to explain related parts of
+ code. - Update the handling of the ast_channel visible_indication
+ field. Storing values that are not stateful is pointless. Control
+ frames that are events or commands should be ignored. - When a
+ bridge first starts, check to see if the peer channel needs to be
+ given ringing indication because the calling side is still
+ ringing. - Rework ast_indicate_data() a bit for the sake of
+ readability. (closes issue #13747) Reported by: davidw Tested by:
+ russell Review: http://reviewboard.digium.com/r/90/
+
+2008-12-13 23:22 +0000 [r164082] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_dial.c: Change the default calldurationlimit from the
+ special value 0 to -1, so we can better detect an exceptional
+ case. This follows on to the changes made in revision 156386.
+ Related to issue #13851. (closes issue #13974) Reported by:
+ paradise Patches: 20081208__bug13974.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: file, blitzrage, ZX81
+
+2008-12-12 22:20 +0000 [r163785] Russell Bryant <russell at digium.com>
+
+ * /: Set the reviewboard:url property on 1.4, as well
+
+2008-12-12 22:03 +0000 [r163761] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, main/editline/read.c: Simple fix for Ctrl-C not
+ immediately exiting Asterisk, but also add a pointer inside
+ editline to look back to asterisk.c, so others don't spend as
+ much time as I did looking (in the wrong place) for the
+ appropriate function. Reported by: ZX81, via the #asterisk-users
+ channel Fixed by: me (license 14)
+
+2008-12-12 14:40 +0000 [r163448-163511] Russell Bryant <russell at digium.com>
+
+ * pbx/pbx_dundi.c: Specify uint32_t for variables storing a CRC32
+ so that it is actually 32 bits on 64-bit machines, as well.
+ (inspired by issue #13879)
+
+ * main/channel.c, main/autoservice.c, include/asterisk/channel.h:
+ Resolve issues that could cause DTMF to be processed out of
+ order. These changes come from team/russell/issue_12658 1) Change
+ autoservice to put digits on the head of the channel's frame
+ readq instead of the tail. If there were frames on the readq that
+ autoservice had not yet read, the previous code would have
+ resulted in out of order processing. This required a new API call
+ to queue a frame to the head of the queue instead of the tail. 2)
+ Change up the processing of DTMF in ast_read(). Some of the
+ problems were the result of having two sources of pending DTMF
+ frames. There was the dtmfq and the more generic readq. Both were
+ used for pending DTMF in various scenarios. Simplifying things to
+ only use the frame readq avoids some of the problems. 3) Fix a
+ bug where a DTMF END frame could get passed through when it
+ shouldn't have. If code set END_DTMF_ONLY in the middle of digit
+ emulation, and a digit arrived before emulation was complete,
+ digits would get processed out of order. (closes issue #12658)
+ Reported by: dimas Tested by: russell, file Review:
+ http://reviewboard.digium.com/r/85/
+
+2008-12-11 23:35 +0000 [r163383] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: When a Ctrl-C or Ctrl-D ends a remote console,
+ on certain shells, the terminal is messed up. By intercepting
+ those events with a signal handler in the remote console, we can
+ avoid those issues. (closes issue #13464) Reported by: tzafrir
+ Patches: 20081110__bug13464.diff.txt uploaded by Corydon76
+ (license 14) Tested by: blitzrage
+
+2008-12-11 22:44 +0000 [r163316] Matt Nicholson <mnicholson at digium.com>
+
+ * pbx/pbx_dundi.c: Clean up the dundi cache every 5 minutes.
+ (closes issue #13819) Reported by: adomjan Patches:
+ pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
+ dundi_clearecache3.diff uploaded by mnicholson (license 96)
+ Tested by: adomjan
+
+2008-12-11 21:46 +0000 [r163092-163253] Russell Bryant <russell at digium.com>
+
+ * funcs/func_strings.c, funcs/func_cut.c: Fix some observed
+ slowdowns in dialplan processing. The change is to remove
+ autoservice usage from dialplan functions that do not need it
+ because they do not perform operations that potentially block.
+ (closes issue #13940) Reported by: tbelder
+
+ * res/res_features.c: Fix an issue that made it so you could only
+ have a single caller executing a custom feature at a time. This
+ was especially problematic when custom features ran for any
+ appreciable amount of time. The fix turned out to be quite
+ simple. The dynamic features are now stored in a read/write list
+ instead of a list using a mutex. (closes issue #13478) Reported
+ by: neutrino88 Fix suggested by file
+
+2008-12-11 16:51 +0000 [r163088] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_agi.c: Don't wait forever, if there's a specified
+ recording timeout. (closes issue #13885) Reported by: bamby
+ Patches: res_agi.c.patch uploaded by bamby (license 430)
+
+2008-12-11 16:46 +0000 [r163080-163084] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Revert this cast to long. Using time_t here
+ causes build failures on a FreeBSD 32-bit build.
+
+ * apps/app_queue.c: Fix a potential crash due to unsafe datastore
+ handling. This patch also contains a conversion from using long
+ to time_t for representing times for a queue, as well as some
+ whitespace fixes. (closes issue #14060) Reported by: nivek
+ Patches: datastore_fixup.patch.corrected uploaded by nivek
+ (license 636) with slight modification from me Tested by: nivek
+
+2008-12-10 22:52 +0000 [r162874-162926] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_musiconhold.c: Oops, inverted logic for a strcasecmp
+ check. Pointed out by mmichelson, thanks!
+
+ * res/res_musiconhold.c: (closes issue #13229) Reported by:
+ clegall_proformatique Ensure that moh_generate does not return
+ prematurely before local_ast_moh_stop is called. Also, the sleep
+ in mp3_spawn now only occurs for http locations since it seems to
+ have been added originally only for failing media streams.
+
+2008-12-10 19:01 +0000 [r162738-162804] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix subscription based MWI up a bit. We only
+ want to put sip: at the beginning of the URI if it is not already
+ there and revert code to ignore destination check if subscribing
+ for MWI. (closes issue #12560) Reported by: vsauer Patches:
+ patch001.diff uploaded by ramonpeek (license 266)
+
+ * channels/chan_sip.c: When a SIP peer unregisters set the expiry
+ time back to 0 so that the 200 OK contains an expires of 0.
+ (closes issue #13599) Reported by: hjourdain Patches:
+ chan_sip.c.diff uploaded by hjourdain (license 583)
+
+2008-12-10 16:45 +0000 [r162671] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/ael/ael.flex: (closes issue #14022)
+ Reported by: wetwired Tested by: murf I checked, and I added a
+ mod to the trunk version of Asterisk that would make it 8-bit
+ transparent on 27 Nov 2007, but I made no such updates to 1.4. My
+ best guess is that 1.4 was released, and it was not appropriate
+ to commit an enhancement. But I'm going to add the same fixes to
+ 1.4 now, for the following reasons: 1. wetwired is correct; 1.4
+ is **mostly** 8-bit transparent now. This is because the lexical
+ token forming rules use . in most 'word' state continuances. It's
+ just the beginning of a 'word' that is picky. 2. Accepting 8-bit
+ chars in some places and not others leads to bug reports like
+ this.
+
+2008-12-10 16:44 +0000 [r162659-162670] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/stringfields.h: Update to stringfield handling
+ so that side-effects on parameters are not evaluated multiple
+ times. An example where this caused a problem was in chan_sip.c,
+ with the line ast_string_field_set(p, fromdomain, ++fromdomain);
+ This patch was originally uploaded to issue #13783 by jamessan.
+ While the issue was closed for other reasons, this patch is valid
+ and fixes a separate problem, and is thus being committed.
+
+ * channels/chan_sip.c: Revert fix for issue 13570. It has caused
+ more problems than it helped to fix. (closes issue #13783)
+ Reported by: navkumar (closes issue #14025) Reported by: ffs
+
+ * doc/misdn.txt: Add missing documentation to misdn.txt (closes
+ issue #14052) Reported by: festr Patches: misdn.txt.patch
+ uploaded by festr (license 443)
+
+2008-12-10 16:05 +0000 [r162653] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Increment the sequence number on the end packets for
+ RFC2833. After reading the RFC some more and doing some testing I
+ agree with this change. (closes issue #12983) Reported by: vt
+ Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
+ 520)
+
+2008-12-09 23:08 +0000 [r162463] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Oops, should be "tz", not "zonetag".
+
+2008-12-09 22:17 +0000 [r162413] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c, include/asterisk/utils.h, main/utils.c: Remove
+ the test_for_thread_safety() function completely. The test is not
+ valid. Besides, if we actually suspected that recursive mutexes
+ were not working, we would get a ton of LOG_ERROR messages when
+ DEBUG_THREADS is turned on. (inspired by a discussion on the
+ asterisk-dev list)
+
+2008-12-09 21:53 +0000 [r162348] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: We appear to have documented tz= in the
+ [general] section of voicemail.conf, without actually having
+ implemented it. Oops. (Reported by Olivier on the -users list)
+
+2008-12-09 21:14 +0000 [r162341] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_directed_pickup.c: Add 'down' as a valid state for
+ directed call pickup. This creeps up when we receive session
+ progress when dialing a device and not ringing. (closes issue
+ #14005) Reported by: ddl
+
+2008-12-09 20:57 +0000 [r162286] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: Fix an issue where callers on an incoming call
+ on an SLA trunk would not hear ringback. We need to make sure
+ that we don't start writing audio to the trunk channel until
+ we're actually ready to answer it. Otherwise, the channel driver
+ will treat it as inband progress, even though all they are
+ getting is silence. (closes issue #12471) Reported by: mthomasslo
+
+2008-12-09 20:44 +0000 [r162273] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_festival.c: Fix double declaration of 'x' on the PPC
+ platform. (closes issue #14038) Reported by: ffloimair
+
+2008-12-09 20:28 +0000 [r162265] Mark Michelson <mmichelson at digium.com>
+
+ * main/pbx.c: If we fail to start a thread for the pbx to run in,
+ we need to be sure to decrease the number of active calls on the
+ system. This fix may relate to ABE-1713, but it is not certain
+ yet.
+
+2008-12-09 20:20 +0000 [r162264] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/ael/ael.flex: In discussion with
+ seanbright on #asterisk-dev, I have added a default rule, and an
+ option to suppress the default rule from being generated in the
+ flex output, for the sake of those OS's where they didn't tweak
+ flex's ECHO macro, and the compiler doesn't like it. The
+ regressions are OK with this.
+
+2008-12-09 19:47 +0000 [r162188-162204] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Make sure that the timestamp for DTMF is not the same
+ as the previous voice frame and do not send audio when
+ transmitting DTMF as this confuses some equipment. (closes issue
+ #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file
+ (license 11) Tested by: ip-rob, bujones
+
+ * main/rtp.c: Take video into account when early bridging RTP.
+ (closes issue #13535) Reported by: davidw
+
+2008-12-09 18:13 +0000 [r162136] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/ael/ael.flex: Previous fix used ast_malloc
+ and ast_copy_string and messed up the standalone stuff. Fixed.
+
+2008-12-09 17:07 +0000 [r162071] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_phone.c: For some reason, after a distclean, gcc
+ started returning 'value computed is not used'. Fixing (for
+ --enable-dev-mode).
+
+2008-12-09 16:46 +0000 [r162014] Russell Bryant <russell at digium.com>
+
+ * apps/app_disa.c: Allow DISA to handle extensions that start with
+ #. (closes issue #13330) Reported by: jcovert
+
+2008-12-09 16:31 +0000 [r162013] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h,
+ pbx/ael/ael.flex: (closes issue #14019) Reported by: ckjohnsonme
+ Patches: 14019.diff uploaded by murf (license 17) Tested by:
+ ckjohnsonme, murf This crash was the result of a few small errors
+ that would combine in 64-bit land to result in a crash. 32-bit
+ land might have seen these combine to mysteriously drop the args
+ to an application call, in certain circumstances. Also, in trying
+ to find this bug, I spotted a situation in the flex input, where,
+ in passing back a 'word' to the parser, it would allocate a
+ buffer larger than necessary. I changed the usage in such
+ situations, so that strdup was not used, but rather, an
+ ast_malloc, followed by ast_copy_string. I removed a field from
+ the pval struct, in u2, that was never getting used, and set in
+ one spot in the code. I believe it was an artifact of a previous
+ fix to make switch cases work invisibly with extens. And, for
+ goto's I removed a '!' from before a strcmp, that has been there
+ since the initial merging of AEL2, that might prevent the proper
+ target of a goto from being found. This was pretty harmless on
+ its own, as it would just louse up a consistency check for users.
+ Many thanks to ckjohnsonme for providing a simplified and
+ complete set of information about the bug, that helped
+ considerably in finding and fixing the problem. Now, to get
+ aelparse up and running again in trunk, and out of its "horribly
+ broken" state, so I can run the regression suite!
+
+2008-12-09 14:52 +0000 [r161948] Russell Bryant <russell at digium.com>
+
+ * main/app.c: Fix a problem with GROUP() settings on a masquerade.
+ The previous code carried over group settings from the old
+ channel to the new one. However, it did nothing with the group
+ settings that were already on the new channel. This patch removes
+ all group settings that already existed on the new channel. I
+ have a more complicated version of this patch which addresses
+ only the most blatant problem with this, which is that a channel
+ can end up with multiple group settings in the same category.
+ However, I could not think of a use case for keeping any of the
+ group settings from the old channel, so I went this route for
+ now. (closes AST-152)
+
+2008-12-08 17:52 +0000 [r161725] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Make the usereqphone option work again.
+ (closes issue #13474) Reported by: mmaguire Patches:
+ 20080912_bug13474.diff uploaded by mmaguire (license 571)
+
+2008-12-05 21:02 +0000 [r161426] Sean Bright <sean.bright at gmail.com>
+
+ * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
+ 161421 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec
+ 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned
+ int). (closes issue #14006) Reported by: alphaque Patches:
+ astobj2.h-patch uploaded by alphaque (license 259) (Slightly
+ modified by seanbright) ........
+
+2008-12-05 16:51 +0000 [r161354] Dwayne M. Hubbard <dhubbard at digium.com>
+
+ * utils/smsq.c: kill a warning
+
+2008-12-05 14:12 +0000 [r161287] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c: Fix a NULL format string warning found by buildbot.
+
+2008-12-04 18:30 +0000 [r161013] Jeff Peeler <jpeeler at digium.com>
+
+ * main/rtp.c: (closes issue #13835) Reported by: matt_b Tested by:
+ jpeeler This mirrors a check that was present in ast_rtp_read to
+ also be in ast_rtp_raw_write to not schedule sending the receiver
+ report if the remote RTCP endpoint address isn't present in the
+ RTCP structure. Closes AST-142.
+
+2008-12-04 16:44 +0000 [r160943] Mark Michelson <mmichelson at digium.com>
+
+ * main/callerid.c: Fix a callerid parsing issue. If someone
+ formatted callerid like the following: "name <number>" (including
+ the quotation marks), then the parts would be parsed as name:
+ "name number: number This is because the closing quotation mark
+ was not discovered since the number and everything after was
+ parsed out of the string earlier. Now, there is a check to see if
+ the closing quote occurs after the number, so that we can know if
+ we should strip off the opening quote on the name. Closes AST-158
+
+2008-12-03 21:54 +0000 [r160770] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Some compilers warn on null format strings;
+ some don't (caught by buildbot)
+
+2008-12-03 21:38 +0000 [r160764] Jason Parker <jparker at digium.com>
+
+ * channels/chan_agent.c: Only show this warning when we want to
+ show it. (closes issue #13982) Reported by: coolmig Patches:
+ chan_agent.c.patch uploaded by coolmig (license 621)
+
+2008-12-03 20:41 +0000 [r160703] Steve Murphy <murf at digium.com>
+
+ * funcs/func_callerid.c: (closes issue #13597) Reported by:
+ john8675309 Patches: patch.13597 uploaded by murf (license 17)
+ Tested by: murf, john8675309 This patch causes the setcid func to
+ update the CDR clid after setting the channel field.
+
+2008-12-03 17:55 +0000 [r160480-160570] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_iax2.c: During bridge code, the channel bridge may
+ return a retry code, if a transfer was initiated but not yet
+ completed. If the bridge is immediately retried, then we may send
+ a storm of TXREQ packets, even though the first set is sent
+ reliably (retransmitted). Fixes AST-137.
+
+ * pbx/pbx_spool.c: If an entry is added to the directory during a
+ scan when another entry expires, then that new entry will not be
+ processed promptly, but must wait for either a future entry to
+ start or a current entry's retry to occur. If no other entries
+ exist in the directory (other than the new entries) when a bunch
+ expire, then the new entries must wait until another new entry is
+ added to be processed. This was a rather weird race condition,
+ really. Fixes AST-147.
+
+ * pbx/pbx_spool.c: Don't start scanning the directory until all
+ modules are loaded, because some required modules (channels,
+ apps, functions) may not yet be in memory yet. Fixes AST-149.
+
+ * channels/chan_sip.c: Jon Bonilla (Manwe) pointed out on the -dev
+ list: "I guess that having only ip-phones in mind is not a good
+ approach. Since it is possible to have a sip proxy connected to
+ asterisk we could receive a 407 (unauthorized) or 483 (too many
+ hops) as response and dialog ending would not be a good
+ behavior." So modified.
+
+2008-12-02 23:58 +0000 [r160390-160411] Terry Wilson <twilson at digium.com>
+
+ * res/res_features.c: Channel is masqueraded, don't keep alive
+
+ * res/res_features.c: A situation like A calls B, A builtin_atxfers
+ B to C, C parks B would lead to a crash. Thanks to file for
+ telling me how to fix it! (closes issue #13854) Reported by: Adam
+ Lee Tested by: otherwiseguy
+
+2008-12-02 17:42 +0000 [r160297] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: When the text does not match exactly (e.g.
+ RTP/SAVP), then the %n conversion fails, and the resulting
+ integer is garbage. Thus, we must initialize the integer and
+ check it afterwards for success. (closes issue #14000) Reported
+ by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by
+ folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded
+ by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
+ uploaded by folke (license 626)
+
+2008-12-02 01:16 +0000 [r160266] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/astmm.h: make compile with dev mode and malloc
+ debug
+
+2008-12-02 00:25 +0000 [r160207] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/stringfields.h, apps/app_voicemail.c,
+ main/pbx.c, main/frame.c: Ensure that Asterisk builds with
+ --enable-dev-mode, even on the latest gcc and glibc.
+
+2008-12-01 Tilghman Lesher <tilghman at digium.com>
+
+ * Released 1.4.23-rc2
+
+2008-12-01 17:27 +0000 [r160003] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Apply some logic used in iax2_indicate() to
+ iax2_setoption(), as well, since they both have the potential to
+ send control frames in the middle of call setup. We have to wait
+ until we have received a message back from the remote end before
+ we try to send any more frames. Otherwise, the remote end will
+ consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
+
+2008-12-01 16:08 +0000 [r159976] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/manager.c: Get rid of the useless format string and argument
+ in the Bogus/ manager channelname. Noted by kpfleming and name
+ Bogus/manager suggested by eliel
+
+2008-12-01 14:52 +0000 [r159900] Russell Bryant <russell at digium.com>
+
+ * .cleancount: Force a "make clean" to avoid a bizarre build issue
+ ...
+
+2008-12-01 14:05 +0000 [r159897] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/manager.c: make manager compile on OpenBSD. The last (10th)
+ argument to ast_channel_alloc here should be a pointer and NULL
+ is not really a pointer.
+
+2008-11-29 16:58 +0000 [r159808] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/enum.c, utils/frame.c, configure, res/res_agi.c,
+ include/asterisk/module.h, main/logger.c, main/dns.c,
+ include/asterisk/threadstorage.h, include/asterisk/utils.h,
+ include/asterisk/devicestate.h, channels/chan_sip.c,
+ include/asterisk/dundi.h, main/jitterbuf.c,
+ channels/chan_agent.c, configure.ac, utils/astman.c,
+ include/asterisk/cli.h, include/asterisk/channel.h,
+ include/jitterbuf.h, include/asterisk/manager.h,
+ main/ast_expr2.c, Makefile, include/asterisk/logger.h,
+ include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c,
+ include/asterisk/linkedlists.h, include/asterisk/lock.h,
+ include/asterisk/strings.h, makeopts.in,
+ include/asterisk/stringfields.h, utils/check_expr.c,
+ channels/chan_vpb.cc, res/res_features.c, channels/chan_iax2.c:
+ update dev-mode compiler flags to match the ones used by default
+ on Ubuntu Intrepid, so all developers will see the same warnings
+ and errors since this branch already had some printf format
+ attributes, enable checking for them and tag functions that
+ didn't have them format attributes in a consistent way
+
+2008-11-26 20:21 +0000 [r159476-159571] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_oss.c, channels/busy.h (removed),
+ channels/ring_tone.h (added), channels/chan_alsa.c,
+ channels/ringtone.h (removed), channels/busy_tone.h (added),
+ channels/Makefile: rename these files so as to avoid conflicts
+ when users update their working copies and have unversioned files
+ already in place
+
+ * channels, agi/Makefile, utils/Makefile, channels/busy.h (added),
+ Makefile.moddir_rules, Makefile.rules, channels/ringtone.h
+ (added), channels/Makefile: simplify (and slightly bug-fix) the
+ recent developer-oriented COMPILE_DOUBLE mode add channels/busy.h
+ and channels/ringtone.h to the repository instead of generating
+ them repeatedtly; most users do not change the settings to build
+ them, but the Makefile rules are still there if they wish to do
+ so ensure that 'make clean' removes dependency files for .i files
+ that are created in COMPILE_DOUBLE mode
+
+2008-11-25 22:41 +0000 [r159316] Steve Murphy <murf at digium.com>
+
+ * main/cdr.c, channels/chan_iax2.c: (closes issue #12694) Reported
+ by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17)
+ Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX
+ fix. the change to cdr.c allows no-answer to percolate up into
+ CDR's, and feels like the right place to locate this fix; if BUSY
+ is done here, no-answer should be, too.
+
+2008-11-25 21:56 +0000 [r159246-159269] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_iax2.c: Don't try to send a response on a NULL pvt.
+ (closes issue #13919) Reported by: barthpbx Patches:
+ chan_iax2.c.patch uploaded by eliel (license 64) Tested by:
+ barthpbx
+
+ * /, channels/chan_iax2.c: Merged revisions 159245 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25
+ Nov 2008) | 7 lines Regression fix for last security fix. Set the
+ iseqno correctly. (closes issue #13918) Reported by: ffloimair
+ Patches: 20081119__bug13918.diff.txt uploaded by Corydon76
+ (license 14) Tested by: ffloimair ........
+
+2008-11-25 17:34 +0000 [r159158] Russell Bryant <russell at digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Add ao2_trylock() to
+ go along with ao2_lock() and ao2_unlock()
+
+2008-11-25 16:23 +0000 [r159096] Terry Wilson <twilson at digium.com>
+
+ * apps/app_festival.c: Add missing variable declaration in the PPC
+ code
+
+2008-11-25 04:50 +0000 [r159025] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_rpt.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: System call ioperm is non-portable, so check for
+ its existence in autoconf. (Closes issue #13863)
+
+2008-11-22 00:04 +0000 [r158629] Jeff Peeler <jpeeler at digium.com>
+
+ * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c: (closes
+ issue #13786) Reported by: tzafrir When compiling against Zaptel
+ dahdi_compat will now only define all the DAHDI defines if the
+ Zaptel define is present. Also, there is no such thing as
+ DAHDI_PRI.
+
+2008-11-21 23:14 +0000 [r158603] Steve Murphy <murf at digium.com>
+
+ * res/res_features.c: In reference to the fix made for 13871, I was
+ merging the fix into 1.6.0 and realized I missed the code in the
+ h-exten block, and didn't catch it because my test case had the
+ h-exten commented out. So, this corrects the code I missed, as a
+ preventative against another crash report. Tested with the
+ h-exten defined, all is well.
+
+2008-11-21 23:07 +0000 [r158600] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c: The passed extension may not be the same in the list
+ as the current entry, because we strip spaces when copying the
+ extension into the structure. Therefore, use the copied item to
+ place the item into the list. (found by lmadsen on -dev, fixed by
+ me)
+
+2008-11-21 22:05 +0000 [r158539] Russell Bryant <russell at digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: When compiling with
+ DEBUG_THREADS, report the real file/func/line for
+ ao2_lock/ao2_unlock
+
+2008-11-21 21:19 +0000 [r158483] Steve Murphy <murf at digium.com>
+
+ * res/res_features.c: (closes issue #13871) Reported by: mdu113
+ This one is totally my fault. The code doesn't even create a
+ bridge if the channel CDR has POST_DISABLED. I didn't check for
+ that at the end of the bridge. Fixed with a few small insertions.
+ Tested. Looks good. No cdr generated, no crash, no unnecc. data
+ objects created either.
+
+2008-11-21 15:24 +0000 [r158053-158306] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: This change had somehow gotten reverted due to
+ a completely unrelated commit. Thanks to Theo Belder on the
+ Asterisk-dev list for pointing this out.
+
+ * include/asterisk/file.h, main/frame.c, main/file.c,
+ include/asterisk/frame.h: There was an issue when attempting to
+ reference an embedded frame in a freed ast_filestream. This patch
+ makes use of the ao2 functions to make sure that we do not free
+ an ast_filestream structure until the embedded ast_frame has been
+ "freed" as well. (closes issue #13496) Reported by: fst-onge
+ Patches: filestream_frame_1_4.diff uploaded by putnopvut (license
+ 60) Tested by: putnopvut Closes AST-89
+
+ * channels/chan_sip.c: We don't handle 4XX responses to BYE well.
+ According to section 15 of RFC 3261, we should terminate a dialog
+ if we receive a 481 or 408 in response to our BYE. Since I am
+ aware of at least one phone manufacturer who may sometimes send a
+ 404 as well, I am being liberal and saying that any 4XX response
+ to a BYE should result in a terminated dialog. (closes issue
+ #12994) Reported by: pabelanger Patches: 12994.patch uploaded by
+ putnopvut (license 60) Closes AST-129
+
+ * apps/app_dial.c, channels/chan_sip.c: Make sure to set the hangup
+ cause on the calling channel in the case that ast_call() fails.
+ For incoming SIP channels, this was causing us to send a 603
+ instead of a 486 when the call-limit was reached on the
+ destination channel. (closes issue #13867) Reported by: still_nsk
+ Patches: 13867.diff uploaded by putnopvut (license 60) Tested by:
+ blitzrage
+
+2008-11-20 01:46 +0000 [r158010] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Merged revision 157977 from
+ https://origsvn.digium.com/svn/asterisk/team/group/issue8824
+ ........ Fixes JIRA ABE-1726 The dial extension could be empty if
+ you are using MISDN_KEYPAD to control ISDN provider features.
+
+2008-11-19 21:34 +0000 [r157859] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree,
+ channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, channels,
+ main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash,
+ codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules,
+ channels/misdn, main/db1-ast/mpool, pbx/Makefile, Makefile.rules,
+ res/snmp, res/Makefile: the gcc optimizer frequently finds broken
+ code (use of uninitalized variables, unreachable code, etc.),
+ which is good. however, developers usually compile with the
+ optimizer turned off, because if they need to debug the resulting
+ code, optimized code makes that process very difficult. this
+ means that we get code changes committed that weren't adequately
+ checked over for these sorts of problems. with this build system
+ change, if (and only if) --enable-dev-mode was used and
+ DONT_OPTIMIZE is turned on, when a source file is compiled it
+ will actually be preprocessed (into a .i or .ii file), then
+ compiled once with optimization (with the result sent to
+ /dev/null) and again without optimization (but only if the first
+ compile succeeded, of course). while making these changes, i did
+ some cleanup work in Makefile.rules to move commonly-used
+ combinations of flag variables into their own variables, to make
+ the file easier to read and maintain
+
+2008-11-18 22:47 +0000 [r157503] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Add some missing invite state changes
+ necessary in the sip_write function. Not setting the invite state
+ correctly on the call was resulting in the Record application
+ leaving empty files. I also have updated the doxygen comment next
+ to the declaration of the INV_EARLY_MEDIA constant to reflect
+ that we also use this state when we *send* a 18X response to an
+ INVITE. (closes issue #13878) Reported by: nahuelgreco Patches:
+ sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
+ (license 162) Tested by: putnopvut
+
+2008-11-18 19:13 +0000 [r157365] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_meetme.c: (closes issue #13899) Reported by: akkornel
+ This fix is the result of a bug fix in ast_app_separate_args
+ r124395. If an argument does not exist it should always be set to
+ a null string rather than a null pointer.
+
+2008-11-18 18:25 +0000 [r157305] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_dial.c, channels/chan_local.c, res/res_features.c,
[... 20855 lines stripped ...]
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