[asterisk-commits] russell: tag 1.4.11 r80186 - in /tags/1.4.11: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Aug 21 14:00:34 CDT 2007
Author: russell
Date: Tue Aug 21 14:00:33 2007
New Revision: 80186
URL: http://svn.digium.com/view/asterisk?view=rev&rev=80186
Log:
importing files for 1.4.11 release
Added:
tags/1.4.11/.lastclean (with props)
tags/1.4.11/.version (with props)
tags/1.4.11/ChangeLog (with props)
Added: tags/1.4.11/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.11/.lastclean?view=auto&rev=80186
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--- tags/1.4.11/ChangeLog (added)
+++ tags/1.4.11/ChangeLog Tue Aug 21 14:00:33 2007
@@ -1,0 +1,10849 @@
+2007-08-21 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.11 released.
+
+2007-08-21 18:42 +0000 [r80183] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Don't record SIP dialog history if it's not
+ turned on. Also, put an upper limit on how many history entires
+ will be stored for each SIP dialog. It is currently set to 50,
+ but can be increased if deemed necessary. (closes issue #10421,
+ closes issue #10418, patches suggested by jmoldenhauer, patches
+ updated by me) (Security implications documented in AST-2007-020)
+
+2007-08-21 16:39 +0000 [r80166-80167] Steve Murphy <murf at digium.com>
+
+ * include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing
+ the diffs from ulaw.h and alaw.h for now; accidentally added them
+ in 80166
+
+ * main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h:
+ This patch solves problem 1 in 8126; it should not slow down the
+ alaw codec, but should prevent signal degradation via multiple
+ trips thru the codec. Fossil estimates the twice thru this codec
+ will prevent fax from working. 4-6 times thru would result
+ hearable, noticeable, voice degradation.
+
+2007-08-21 15:22 +0000 [r80132] Russell Bryant <russell at digium.com>
+
+ * channels/chan_mgcp.c: Don't try to dereference the owner channel
+ when it may not exist (issue #10507, maxper)
+
+2007-08-21 15:03 +0000 [r80130] Jason Parker <jparker at digium.com>
+
+ * configs/cdr.conf.sample: (issue #10510) Reported by: casper
+ Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few
+ errors in sample cdr config file.
+
+2007-08-20 21:57 +0000 [r80088] Russell Bryant <russell at digium.com>
+
+ * apps/app_queue.c: Fix the build of app_queue
+
+2007-08-20 21:39 +0000 [r80049-80086] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: After a discussion on #asterisk-dev, it was
+ decided that this should be in 1.4 as well. (issue #10424,
+ reported and patched by irroot)
+
+ * apps/app_queue.c: Found a pointless ternary if. member->dynamic
+ was set to 1 and has no opportunity to change between then and
+ this line, so "dynamic" will ALWAYS be output.
+
+2007-08-20 16:08 +0000 [r80047] Jason Parker <jparker at digium.com>
+
+ * configs/extensions.conf.sample: (issue #10499) Reported by:
+ casper Patches: extensions.conf.sample.diff uploaded by casper
+ (license 55) Update CLI examples in extensions.conf.sample to
+ reflect command changes.
+
+2007-08-20 15:34 +0000 [r80044] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Ukrainian language voicemail support.
+ (closes issue #10458, reported and patched by Oleh)
+
+2007-08-20 02:42 +0000 [r79998] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Missing curly braces. Oops. (Reported by
+ snuffy via IRC)
+
+2007-08-18 14:30 +0000 [r79947] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Don't allocate vmu for messagecount when we
+ could just use the stack instead (closes issue #10490) Also,
+ remove a useless (and leaky) SQLAllocHandle (closes issue #10480)
+
+2007-08-17 21:01 +0000 [r79912] Russell Bryant <russell at digium.com>
+
+ * channels/chan_zap.c: Avoid a crash in the handling of DTMF based
+ Caller ID. It is valid for ast_read to return NULL in the case
+ that the channel has been hung up. (crash reported by
+ anonymouz666 on IRC in #asterisk-dev)
+
+2007-08-17 19:14 +0000 [r79906] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Patch allows for more seamless transition
+ from file storage voicemail to ODBC storage voicemail. If a
+ retrieval of a greeting from the database fails, but the file is
+ found on the file system, then we go ahead an insert the greeting
+ into the database. The result of this is that people who switch
+ from file storage to ODBC storage do not need to rerecord their
+ voicemail greetings.
+
+2007-08-17 19:12 +0000 [r79902-79904] Jason Parker <jparker at digium.com>
+
+ * channels/chan_sip.c, main/utils.c, include/asterisk/strings.h:
+ Don't send a semicolon over the wire in sip notify messages.
+ Caused by fix for issue 9938. I basically took the code that
+ existed before 9938 was fixed, and copied it into a new function
+ - ast_unescape_semicolon There should be very few places this
+ will be needed (pbx_config does NOT need this (see issue 9938 for
+ details)) Issue 10430, patch by me, with help/ideas from murf
+ (thanks murf).
+
+ * channels/chan_local.c: Re-add the setting of callerid name and
+ number. Issue 10485, reported by and fix explained by paradise.
+
+2007-08-17 13:37 +0000 [r79857] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Fix some crashes in chan_sip. This patch
+ changes various places that add items to the scheduler to ensure
+ that they don't overwrite the ID of a previously scheduled item.
+ If there is one, it should be removed. (closes issue #10391,
+ closes issue #10256, probably others, patch by me)
+
+2007-08-17 08:22 +0000 [r79833] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c: sometimes we don't need to signal dtmf
+ tones to asterisk, we just want them to go through as inband.
+ Otherwise they might be generated by the other channel partner
+ and then there is a double tone.
+
+2007-08-16 22:32 +0000 [r79756-79792] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: Fix a little race condition that could
+ cause a crash if two channels had MOH stopped at the same time
+ that were using a class that had been marked for deletion when
+ its use count hits zero.
+
+ * res/res_musiconhold.c: This patch fixes a bug where reloading the
+ module with "module reload" did not delete classes from memory
+ that were no longer in the config. This patch fixes that problem
+ as well as another one. Previously, if you reloaded MOH using the
+ "moh reload" CLI command, which behaved differently than "module
+ reload ...", MOH had to be stopped on every channel and started
+ again immediately. However, there was no way to tell what class
+ was being used, so they would all fall back to the default class.
+ (closes issue #10139) Reported by: blitzrage Patches:
+ asterisk-10139-advanced.diff.txt uploaded by jamesgolovich
+ (license 176) Tested by: jamesgolovich
+
+ * channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were
+ introduced by making frame handling and scheduling
+ multi-threaded. Unfortunately, we have to do some expensive
+ deadlock avoidance when queueing frames on to the ast_channel
+ owner of the IAX2 pvt struct. This was already handled for
+ regular frames, but ast_queue_hangup and ast_queue_control were
+ still used directly. Making these changes introduced even more
+ places where the IAX2 pvt struct can disappear in the context of
+ a function holding its lock due to calling a function that has to
+ unlock/lock it to avoid deadlocks. I went through and fixed all
+ of these places to account for this possibility. (issue #10362,
+ patch by me)
+
+2007-08-16 21:16 +0000 [r79690-79748] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_agent.c: Fixes a problem where agents would get
+ stuck busy due to their wrapuptime being longer than the queue's
+ wrapuptime and ringinuse=no for the queue. (closes issue #10215,
+ reported by Doug, repaired by me) Special thanks to fkasumovic
+ for pointing out the source of the problem and to bweschke for
+ helping to come up with a solution!
+
+ * apps/app_voicemail.c: base_encode is not trying to open a log
+ file, so we should not call it a log file in the warning.
+ (related to issue #10452, reported by bcnit)
+
+2007-08-16 09:37 +0000 [r79665] Philippe Sultan <philippe.sultan at gmail.com>
+
+ * res/res_jabber.c: A fix for two critical problems detected while
+ working with Daniel McKeehan in issue #10184. Upon priority
+ change, the resource list is not NULL terminated when moving an
+ item to the end of the list. This makes Asterisk endlessy loop
+ whenever it needs to read the list. Jids with different resource
+ and priority values, like in Gmail's and GoogleTalk's jabber
+ clients put that problem in evidence. Upon reception of a 'from'
+ attribute with an empty resource string, Asterisk crashes when
+ trying to access the found->cap pointer if the resource list for
+ the given buddy is not empty. This situation is perfectly valid
+ and must be handled. The Gizmoproject's jabber client put that
+ problem in evidence. Also added a few comments in the code as
+ well as a handle for the capabilities from Gmail's jabber client,
+ which are stored in a caps:c tag rather than the usual c tag.
+ Closes issue #10184.
+
+2007-08-16 08:21 +0000 [r79642] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when
+ we want to send IA5 Chars.
+
+2007-08-15 14:40 +0000 [r79553] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: (closes issue #10440) Reported by: irroot (closes
+ issue #10454) Reported by: flo_turc Increase maximum timestamp
+ skew to 120. 20 was apparently far too low.
+
+2007-08-15 14:26 +0000 [r79527] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Fixed an error in the Russian language
+ voicemail intro. (issue #10458, reported and patched by Oleh)
+
+2007-08-15 14:18 +0000 [r79523] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: (closes issue #10456) Reported by: irroot
+ Patches: sip_timeout.patch uploaded by irroot (license 52) Change
+ hardcoded timer value to defined value. I'm doing this in 1.4 as
+ well so if it needs to be changed in the future this place would
+ not have been forgotten.
+
+2007-08-14 18:49 +0000 [r79436-79470] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix another spot where an iax2_peer would
+ be leaked if realtime was in use.
+
+ * channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2
+ related to the use of realtime. I found these while working on
+ iax2_peer object reference tracking.
+
+2007-08-14 15:27 +0000 [r79397] Joshua Colp <jcolp at digium.com>
+
+ * res/res_features.c: (closes issue #10415) Reported by: atis
+ Revert fix for #10327 as it causes more issues then it solves.
+
+2007-08-13 22:40 +0000 [r79363] Steve Murphy <murf at digium.com>
+
+ * pbx/pbx_ael.c: memset really, really needs to be used here.
+
+2007-08-13 21:57 +0000 [r79334] Joshua Colp <jcolp at digium.com>
+
+ * res/res_speech.c, apps/app_speech_utils.c,
+ include/asterisk/speech.h: Instead of accepting a single DTMF
+ character accept a full string.
+
+2007-08-13 20:37 +0000 [r79272-79301] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Don't call find_peer in
+ registry_authrequest with the pvt lock held to avoid a deadlock.
+
+ * channels/chan_iax2.c: Release the pvt lock before calling
+ find_peer in register_verify to avoid a deadlock. Also, remove
+ some unnecessary locking in auth_fail that was only done
+ recursively.
+
+ * channels/chan_iax2.c: Don't call find_peer within update_registry
+ with a pvt lock held. This can cause a deadlock as the code will
+ eventually call find_callno.
+
+ * channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I
+ have tracked them down to a single core issue. You can not call
+ find_callno() while holding a pvt lock as this function has to
+ lock another (every) other pvt lock. Doing so can lead to a
+ classic deadlock. So, I am tracking down all of the code paths
+ where this can happen and fixing them. The fix I committed
+ earlier today was along the same theme. This patch fixes some
+ code down the path of authenticate_reply.
+
+2007-08-13 17:49 +0000 [r79255] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael-test/ref.ael-vtest21 (added),
+ pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ael-vtest21/extensions.ael (added),
+ pbx/ael/ael-test/ael-vtest21 (added),
+ pbx/ael/ael-test/ref.ael-vtest17,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c,
+ pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch
+ fixes bug 10411. I added a new regression test, some regression
+ test cleanups
+
+2007-08-13 15:28 +0000 [r79214] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix a potential deadlock in socket_process.
+ check_provisioning can eventually call find_callno. You can't
+ hold a pvt lock while calling find_callno because it goes through
+ and locks every single one looking for a match.
+
+2007-08-13 14:51 +0000 [r79174-79207] Joshua Colp <jcolp at digium.com>
+
+ * res/res_speech.c, apps/app_speech_utils.c,
+ include/asterisk/speech.h: Add an API call to allow the engine to
+ know that DTMF was received.
+
+ * channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c,
+ channels/chan_local.c, channels/chan_misdn.c,
+ channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_h323.c, channels/chan_gtalk.c,
+ channels/chan_iax2.c: (closes issue #10437) Reported by: haklin
+ Don't set the callerid name and number a second time on a newly
+ created channel. ast_channel_alloc itself already sets it and
+ setting it twice would cause a memory leak.
+
+2007-08-11 05:23 +0000 [r79142] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * res/res_odbc.c: Ensure the connection gets marked as used at
+ allocation time (closes issue #10429, report and fix by
+ mnicholson)
+
+2007-08-10 20:53 +0000 [r79044-79099] Steve Murphy <murf at digium.com>
+
+ * main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From
+ a user complaint on #asterisk, I have forced pbx_spool to explain
+ what reason codes mean, when they are logged
+
+ * main/cdr.c: Re bug behavior mentioned in #asterisk, made this
+ tweak to code, to prevent hundreds of log messages from being
+ generated
+
+ * main/cdr.c: This will help debug; from a question asked on
+ #asterisk
+
+2007-08-10 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.10.1 released.
+
+2007-08-10 15:20 +0000 [r78995] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/lock.h: The last set of changes that I made to
+ "core show locks" made it not able to track mutexes unless they
+ were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized
+ with ast_mutex_init() were not tracked. It should work now.
+
+2007-08-10 14:15 +0000 [r78951-78955] Joshua Colp <jcolp at digium.com>
+
+ * main/file.c: Don't bother having the core pass through or emulate
+ begin DTMF frames when in an ast_waitstream. It only cares about
+ the end of DTMF.
+
+ * configs/queues.conf.sample: (closes issue #10422) Reported by:
+ bhowell Add note to sample configuration about module load order
+ and how it can cause perfectly good queue members to be marked as
+ invalid.
+
+2007-08-10 13:24 +0000 [r78936] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/ie.c,
+ channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser
+ information element. We send them now also in the disconnect
+ message.
+
+2007-08-09 23:47 +0000 [r78907] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Improved a bit of logic regarding
+ comma-separated mailboxes in has_voicemail. Also added some
+ braces to some compound if statements since unbraced if
+ statements scare me in general.
+
+2007-08-09 23:10 +0000 [r78891] Steve Murphy <murf at digium.com>
+
+ * Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty
+ output
+
+2007-08-09 22:03 +0000 [r78826-78860] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Removing some extra debug code I left in my
+ last commit
+
+ * apps/app_voicemail.c: Quite a few changes regarding IMAP storage.
+ 1. instead of using inboxcount as the core message counting
+ function, we use messagecount instead. This makes it possible to
+ count messages in folders besides just INBOX and Old. 2.
+ inboxcount and hasvoicemail now use messagecount as their means
+ of determining return values. 3. Added a copy_message function
+ for IMAP storage. Unfortunately I don't have the means to test
+ it, but it seems like a pretty straightforward function. 4.
+ Removed a #ifndef IMAP_STORAGE and matching #endif from
+ leave_voicemail for a couple of reasons. One, we want to support
+ copying mail to multiple IMAP boxes, and two, IMAP was broken
+ because a STORE macro had been moved into this section of code.
+
+ * channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I
+ put some new code in the wrong place and so I've reverted the
+ canreinvite section to how it was and put my new code where it
+ should be.
+
+2007-08-09 17:58 +0000 [r78717-78778] Russell Bryant <russell at digium.com>
+
+ * apps/app_voicemail.c: add a comment to indicate that inboxcount
+ for ODBC_STORAGE needs to be fixed to support multiple mailboxes
+
+ * apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for
+ ODBC_STORAGE. Also, leave a comment for this to be fixed for
+ IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was
+ working on this code right now for another reason. This is broken
+ even worse in trunk, but for a different reason. The fact that
+ the mailbox option supported multiple mailboxes is completely not
+ obvious from the code in the channel drivers. Anyway, I will fix
+ that in another commit ...
+
+ * apps/app_meetme.c: Fix a problem with the combination of the 'F'
+ option to pass DTMF through a conference and options that use
+ DTMF to activate various features. The problem was that the BEGIN
+ frame would be passed through, but the END frame would get
+ intercepted to activate a feature. Then, the other conference
+ members would hear DTMF for forever, which they didn't seem to
+ like very much. (closes issue #10400, reported by stevefeinstein,
+ fixed by me)
+
+2007-08-08 19:29 +0000 [r78646] Jason Parker <jparker at digium.com>
+
+ * doc/jabber.txt: Fix mogs email address.
+
+2007-08-08 18:16 +0000 [r78575-78620] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Fixed some compiler warnings so that
+ compiling with dev-mode and IMAP storage would not have any
+ errors. This section of code may get changed again shortly since
+ my change uncovers a rather silly bit of logic.
+
+ * apps/app_queue.c: Changing a bit of logic so that someone will
+ NEVER exit the queue on timeout unless they have enabled the 'n'
+ option. This commit relates to issue #10320. Thanks to
+ jfitzgibbon for detailing the idea behind this code change.
+
+2007-08-08 13:51 +0000 [r78569] Joshua Colp <jcolp at digium.com>
+
+ * configs/sip.conf.sample: (closes issue #10335) Reported by:
+ adamgundy Update sip.conf to include another scenario where
+ directrtpsetup will fail.
+
+2007-08-07 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.10 released.
+
+2007-08-07 20:57 +0000 [r78488] Russell Bryant <russell at digium.com>
+
+ * res/res_config_odbc.c: Fix the build of this module on 64-bit
+ platforms
+
+2007-08-07 19:43 +0000 [r78450] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: The logic behind inboxcount's return value
+ was reversed in has_voicemail and message_count. (closes issue
+ #10401, reported by st1710, patched by me)
+
+2007-08-07 19:34 +0000 [r78437] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * res/res_odbc.c: Don't free the environment handle when the
+ connection fails, because other connections might be depending
+ upon it
+
+2007-08-07 19:11 +0000 [r78416] Jason Parker <jparker at digium.com>
+
+ * channels/chan_sip.c: Allow chan_sip to build in devmode
+
+2007-08-07 19:09 +0000 [r78415] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c, res/res_config_odbc.c,
+ apps/app_directory.c: Reconnection doesn't happen automatically
+ when a DB goes down (fixes issue #9389)
+
+2007-08-07 18:25 +0000 [r78375] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Properly check the capabilities count to
+ avoid a segfault. (ASA-2007-019)
+
+2007-08-07 17:45 +0000 [r78371] Russell Bryant <russell at digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) |
+ 4 lines Revert patch committed for issue #9660. It broke E&M
+ trunks. (closes issue #10360) (closes issue #10364) ........
+
+2007-08-06 21:41 +0000 [r78275] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c: Add additional DTMF log messages to help when
+ debugging issues.
+
+2007-08-06 20:44 +0000 [r78184-78242] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix an issue where dynamic threads can get
+ free'd, but still exist in the dynamic thread list. (closes issue
+ #10392, patch from Mihai, with credit to his colleague, Pete)
+
+ * include/asterisk/linkedlists.h: Fix the return value of
+ AST_LIST_REMOVE(). This shouldn't be causing any problems,
+ though, because the only code that uses the return value only
+ checks to see if it is NULL. (closes issue #10390, pointed out by
+ mihai)
+
+2007-08-06 16:32 +0000 [r78182] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: It is possible for a transfer to occur
+ before the remote device has our tag in which case they send none
+ in the transfer. In this case we need to not fail the transfer
+ dialog lookup.
+
+2007-08-06 16:30 +0000 [r78180] Jason Parker <jparker at digium.com>
+
+ * main/config.c: Fix an issue with using UpdateConfig (manager
+ action) where escaped semicolons in a config would be converted
+ to just semicolons (\; to ;) Issue 9938
+
+2007-08-06 15:27 +0000 [r78166-78172] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that
+ we pass through RTP timestamp information we need to make the
+ allowed timestamp skew considerably less. There are situations
+ where a source may change and due to the timestamp difference the
+ receiver will experience an audio gap since we did not indicate
+ by setting the marker bit that the source changed.
+
+ * configure, configure.ac: (closes issue #10383) Reported by: rizzo
+ Include stdlib.h so NULL gets defined for gethostbyname_r checks.
+
+2007-08-06 13:33 +0000 [r78164] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fixed a mistake I made in realtime_peer
+ which caused it to return NULL every time. Thanks to Jon Fealy
+ for emailing me the correction.
+
+2007-08-05 14:18 +0000 [r78146] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes
+ bug #10382)
+
+2007-08-05 04:15 +0000 [r78143] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/lock.h: Fix compilation failure when
+ MALLOC_DEBUG is enabled, but DEBUG_THREADS is not
+
+2007-08-05 03:29 +0000 [r78139] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * channels/chan_sip.c: If peer is not found, the error message is
+ misleading (should be peer not found, not ACL failure)
+
+2007-08-03 20:25 +0000 [r78103] Mark Michelson <mmichelson at digium.com>
+
+ * main/config.c, channels/chan_sip.c, include/asterisk/config.h:
+ Changed the behavior of sip's realtime_peer function to match the
+ corresponding way of matching for non-realtime peers. Now matches
+ are made on both the IP address and port number, or if the
+ insecure setting is set to "port" then just match on the IP
+ address. In order to accomplish this, I also added a new API
+ call, ast_category_root, which returns the first variable of an
+ ast_category struct
+
+2007-08-03 20:14 +0000 [r78028-78101] Russell Bryant <russell at digium.com>
+
+ * apps/app_voicemail.c: (closes issue #10194) Reported by:
+ blitzrage Patches: bug0010194 uploaded by vovochka Tested by:
+ blitzrage Fix a problem when you call Voicemail() with multiple
+ mailboxes specified and ODBC_STORAGE is in use. The audio part of
+ the message was only given to the first mailbox specified.
+
+ * main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some
+ improvements to lock debugging. These changes take effect with
+ DEBUG_THREADS enabled and provide the following: * This will keep
+ track of which locks are held by which thread as well as which
+ lock a thread is waiting for in a thread-local data structure. A
+ reference to this structure is available on the stack in the
+ dummy_start() function, which is the common entry point for all
+ threads. This information can be easily retrieved using gdb if
+ you switch to the dummy_start() stack frame of any thread and
+ print the contents of the lock_info variable. * All of the
+ thread-local structures for keeping track of this lock
+ information are also stored in a list so that the information can
+ be dumped to the CLI using the "core show locks" CLI command.
+ This introduces a little bit of a performance hit as it requires
+ additional underlying locking operations inside of every
+ lock/unlock on an ast_mutex. However, the benefits of having this
+ information available at the CLI is huge, especially considering
+ this is only done in DEBUG_THREADS mode. It means that in most
+ cases where we debug deadlocks, we no longer have to request
+ access to the machine to analyze the contents of ast_mutex_t
+ structures. We can now just ask them to get the output of "core
+ show locks", which gives us all of the information we needed in
+ most cases. I also had to make some additional changes to astmm.c
+ to make this work when both MALLOC_DEBUG and DEBUG_THREADS are
+ enabled. I disabled tracking of one of the locks in astmm.c
+ because it gets used inside the replacement memory allocation
+ routines, and the lock tracking code allocates memory. This
+ caused infinite recursion.
+
+ * channels/chan_iax2.c: Only pass through HOLD and UNHOLD control
+ frames when the mohinterpret option is set to "passthrough". This
+ was pointed out by Kevin in the middle of a training session.
+
+ * channels/chan_iax2.c: Don't reuse the timespec that was set to 0
+ in the previous timedwait as it will just return immediately.
+ Also, fix some logic so the thread's lock isn't unlocked twice in
+ the weird case of dynamic threads getting acquired right after a
+ timeout. (pointed out by SteveK)
+
+2007-08-02 21:53 +0000 [r77993-77996] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we
+ actually allow 6 chars to be sent. Also make note of the "A"
+ option of date format. Issue 9779, modifications by DEA, wedhorn,
+ and myself.
+
+ * channels/chan_skinny.c: If a device disconnects, the session will
+ go away. If this happens during call setup, we need to give up.
+ Issue 10325.
+
+2007-08-02 19:25 +0000 [r77949] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix the case where a dynamic thread times
+ out waiting for something to do during the first time it runs.
+ This shouldn't ever happen, but we should account for it anyway.
+ (pointed out by pete, who works with mihai)
+
+2007-08-02 18:42 +0000 [r77947] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Make sure we clear the prompt status
+ message on a hangup. Also rearrange messages to better fit with
+ what a wireshark trace shows it should be. Issue 10299, initial
+ patch and solution by sbisker, modified by me to fit with
+ wireshark trace.
+
+2007-08-02 18:21 +0000 [r77945] Steve Murphy <murf at digium.com>
+
+ * main/fskmodem.c, /: Merged revisions 77942 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1
+ line This patch hopefully solves 10141; The user is running with
+ it, and it doesn't appear to harm asterisk's operation, and may
+ prevent a crash. I'll store it in 1.2, as we have shut down
+ support on 1.2, but since I developed the patch before support
+ finished, and it might affect 1.4 and trunk, I'm going ahead with
+ it. ........
+
+2007-08-02 18:04 +0000 [r77939-77943] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix another race condition in the handling
+ of dynamic threads. If the dynamic thread timed out waiting for
+ something to do, but was acquired to perform an action
+ immediately afterwords, then wait on the condition again to give
+ the other thread a chance to finish setting up the data for what
+ action this thread should perform. Otherwise, if it immediately
+ continues, it will perform the wrong action. (reported on IRC by
+ mihai, patch by me) (related to issue #10289)
+
+ * channels/chan_iax2.c: Add another sanity check to
+ vnak_retransmit(). This check ensures that frames that have
+ already been marked for deletion don't get retransmitted. (closes
+ issue #10361, patch from mihai)
+
+2007-08-02 15:15 +0000 [r77890-77894] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Make sure that we show the correct
+ extension if dialed from a macro "From: 5555" rather than "From:
+ s" Issue 10358, initial patch by DEA, reworked by me to use S_OR,
+ tested by sbisker
+
+ * channels/chan_skinny.c: Put in some additional debug information
+ for softkey/stimulus messages. Issue 10291, patch by DEA.
+
+2007-08-01 22:16 +0000 [r77887] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix some race conditions which have been
+ causing weird problems in chan_iax2. The most notable problem is
+ that people have been seeing storms of VNAK frames being sent due
+ to really old frames mysteriously being in the retransmission
+ queue and never getting removed. It was possible that a dynamic
+ thread got created, but did not acquire its lock before the
+ thread that created it signals it to perform an action. When this
+ happens, the thread will sleep until it hits a timeout, and then
+ get destroyed. So, the action never gets performed and in some
+ cases, means a frame doesn't get transmitted and never gets freed
+ since the scheduler never gets a chance to reschedule
+ transmission. Another less severe race condition is in the
+ handling of a timeout for a dynamic thread. It was possible for
+ it to be acquired to perform at action at the same time that it
+ hit a timeout. When this occurs, whatever action it was acquired
+ for would never get performed. (patch contributed by Mihai and
+ SteveK) (closes issue #10289) (closes issue #10248) (closes issue
+ #10232) (possibly related to issue #10359)
+
+2007-08-01 22:14 +0000 [r77886] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does
+ not compile cleanly (missing def)
+
+2007-08-01 21:08 +0000 [r77883] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Fix an issue that caused one-way audio on
+ some newer devices (specifically the 7921), due to sending
+ packets in the wrong order during hangup. Also make sure we clear
+ tones/messages on the correct line/instance. Issue 10291, patch
+ by DEA, tested by sbisker and myself.
+
+2007-08-01 18:08 +0000 [r77863-77871] Joshua Colp <jcolp at digium.com>
+
+ * main/cli.c: (closes issue #10351) Reported by: ftarz Some
+ platforms don't like it when you pass NULL to vsnprintf so pass
+ "" instead.
+
+ * include/asterisk/threadstorage.h, channels/chan_mgcp.c,
+ apps/app_voicemail.c, main/acl.c, utils/smsq.c,
+ channels/chan_iax2.c: Add some fixes for building on Solaris.
+
+ * main/utils.c: Whoops, I meant R_5 not R5.
+
+ * configure, configure.ac: And for my last trick... make sure that
+ if gethostbyname_r is exported by a library that it is used.
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/utils.c: Extend autoconf logic to determine which version of
+ gethostbyname_r is on the system.
+
+2007-08-01 14:08 +0000 [r77852-77854] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Fixes an issue I introduced to queues wherein a
+ queue with joinempty=yes would kick people out of the queue
+ because of erroneously thinking the 'n' option was in use.
+ (closes issue #10320, reported by jfitzgibbon, patched by me,
+ tested by blitzrage and me) Thank you blitzrage for all the
+ testing you've done lately with queues! It's much appreciated!
+
+ * apps/app_queue.c: If a queue uses dynamic realtime members, then
+ the member list should be updated after each attempt to call the
+ queue. This fixes an issue where if a caller calls into a queue
+ where no one is logged in, they would wait forever even if a
+ member logged in at some point. (closes issue #10346, reported by
+ and tested by blitzrage, patched by me)
+
+2007-07-31 21:09 +0000 [r77845-77846] Jim Dixon <telesistant at hotmail.com>
+
+ * apps/app_rpt.c: Much newer version, 0.70 with much additions
+
+ * main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF
+ receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in
+ logic in TONE_VERIFY/RELAX mode in chan_zap.
+
+2007-07-31 20:59 +0000 [r77844] Steve Murphy <murf at digium.com>
+
+ * /, contrib/scripts/ast_grab_core: Merged revisions 77842 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1
+ line This probably isn't super-general, but it's a first stab at
+ using kill -11 to generate a core file instead of gcore. ........
+
+2007-07-31 16:17 +0000 [r77831] Joshua Colp <jcolp at digium.com>
+
+ * res/res_speech.c, include/asterisk/speech.h: Add a flag to the
+ speech API that allows an engine to set whether it received
+ results or not.
+
+2007-07-31 15:53 +0000 [r77827] Kevin P. Fleming <kpfleming at digium.com>
+
+ * build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without
+ DEBUG_THREADS or it does nothing
+
+2007-07-31 15:21 +0000 [r77824] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: This patch makes Asterisk send 100 Trying
+ provisional responses upon receipt of re-invites. This makes it
+ so that if there are two or more Asterisk servers between
+ endpoints, the Asterisk servers will not keep retransmitting the
+ re-invites. (closes issue #10274, reported by cstadlmann, patched
+ by me with approval from file)
+
+2007-07-30 20:17 +0000 [r77795] Jason Parker <jparker at digium.com>
+
+ * main/say.c: Applications like SayAlpha() should not hang up the
+ channel if you request an "unknown" character such as a comma.
+ Instead, skip the character and move on. Issue 10083, initial
+ patch by jsmith, modified by me.
+
+2007-07-30 20:16 +0000 [r77785-77794] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix an issue that could potentially cause
+ corruption of the global iax frame queue. In the network_thread()
+ loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE
+ macro. However, to remove an element of the list within this
+ loop, it used AST_LIST_REMOVE, instead of
+ AST_LIST_REMOVE_CURRENT, which I believe could leave some of the
+ internal variables of the SAFE macro invalid. Mihai says that he
+ already made this change in his local copy and it didn't help his
+ VNAK storm issues, but I still think it's wrong. :)
+
+ * res/res_agi.c: (closes issue #10279) Reported by: seanbright
+ Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by
+ seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch
+ uploaded by seanbright (license 71) Allow the "agi_network: yes"
+ line to be printed out in the AGI debug output. Also, allow
+ partial writes to be handled when writing out this line just like
+ it is for all of the others.
+
+ * main/channel.c: file and I both committed changes for issue
+ #10301. Remove a duplicated assignment to restore the original
+ value of the previous channel.
+
+2007-07-30 18:43 +0000 [r77783] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * /, res/res_agi.c: Merged revisions 77782 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007)
+ | 2 lines Revert change in revision 71656, even though it fixed a
+ bug, because many people were depending upon the (broken)
+ behavior. ........
+
+2007-07-30 17:29 +0000 [r77780] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: (closes issue #10301) Reported by: fnordian
+ Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
+ (license 110) Additional changes by me Fix some problems in
+ channel_find_locked() which can cause an infinite loop. The
+ reference to the previous channel is set to NULL in some cases.
+ These changes ensure that the reference to the previous channel
+ gets restored before needing it again. I'm not convinced that the
+ code that is setting it to NULL is really the right thing to do.
+ However, I am making these changes to fix the obvious problem and
+ just leaving an XXX comment that it needs a better explanation
+ that what is there now.
+
+2007-07-30 17:11 +0000 [r77768-77778] Joshua Colp <jcolp at digium.com>
+
+ * res/res_features.c: (closes issue #10327) Reported by: kkiely
+ Instead of directly mucking with the extension/context/priority
[... 10052 lines stripped ...]
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