[asterisk-commits] russell: tag 1.2.16-netsec r57593 -
/tags/1.2.16-netsec/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Mar 2 17:19:06 MST 2007
Author: russell
Date: Fri Mar 2 18:19:06 2007
New Revision: 57593
URL: http://svn.digium.com/view/asterisk?view=rev&rev=57593
Log:
importing files for 1.2.16-netsec release
Added:
tags/1.2.16-netsec/.lastclean (with props)
tags/1.2.16-netsec/.version (with props)
tags/1.2.16-netsec/ChangeLog (with props)
Added: tags/1.2.16-netsec/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.2.16-netsec/.lastclean?view=auto&rev=57593
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==============================================================================
--- tags/1.2.16-netsec/ChangeLog (added)
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@@ -1,0 +1,4964 @@
+2007-03-02 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.2.16 released
+
+2007-03-02 18:32 +0000 [r57523] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/isdn_lib.c: fixed typo
+
+2007-03-02 17:02 +0000 [r57475] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: If a SIP message comes in and goes to a
+ method handler that requires additional values that may not be
+ present then send back an error.
+
+2007-03-02 16:39 +0000 [r57458] Steve Murphy <murf at digium.com>
+
+ * pbx.c: further refinement in wording of goto documentation, as
+ per 9156, goto not proceeding to next instruction
+
+2007-03-01 22:19 +0000 [r57317] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_local.c: Don't even attempt to optimize things when
+ a proxy channel is involved. It will just explode in weird and
+ unexplaineable ways. (issue #9175 reported by
+ clegall_proformatique)
+
+2007-02-28 19:12 +0000 [r57118] Steve Murphy <murf at digium.com>
+
+ * pbx.c: a small documentation update, to reflect reality in the
+ goto doc strings, as per 9156, Goto does not proceed to next prio
+ if jump fails
+
+2007-02-28 18:55 +0000 [r57092] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_agent.c: Fix a few more issues with the agent
+ logoff CLI command. (issue #9123 reported by arbrandes)
+
+2007-02-28 16:09 +0000 [r57034] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed
+ bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302,
+ #303, #304
+
+2007-02-26 20:05 +0000 [r56850] Joshua Colp <jcolp at digium.com>
+
+ * pbx/pbx_config.c: Obey the clearglobalvars option in extensions
+ reload (or dialplan reload depending on your version). (issue
+ #9146 reported by ramonpeek)
+
+2007-02-26 00:34 +0000 [r56729] Russell Bryant <russell at digium.com>
+
+ * utils.c: Ensure that lock.h is included in utils.c with
+ AST_API_MODULE defined so that the implementations will be
+ properly included when the AST_INLINE_API functions are not going
+ to be inlined. (issue #9124, festr)
+
+2007-02-25 14:38 +0000 [r56684] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * channel.c: Issue 9130 - If prev is the last item on the channel
+ list, then evaluating additional conditions (e.g. name prefix)
+ will cause a NULL dereference.
+
+2007-02-23 23:20 +0000 [r56406-56504] Russell Bryant <russell at digium.com>
+
+ * asterisk.c: Fix up a couple more signal handlers to not do bad
+ things that could cause various undesirable results. The other
+ day, I made Asterisk deadlock by hitting Control-C because of a
+ bad signal handler. Now, signal handlers just set a flag and
+ write to an alert pipe for the flag to be handled. Then, there is
+ another thread that is monitoring for these flags. If being run
+ in console mode, it is just the main thread. If Asterisk is in
+ the background, a thread is created to do it.
+
+ * channels/chan_iax2.c: Don't destroy mutexes before unregistering
+ all of the entry points from the core. Also, fix a potential
+ memory leak from not destroying the locks for all of the possible
+ call numbers (about 32k of them).
+
+2007-02-22 23:19 +0000 [r56230-56279] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_agent.c: Always defer Agent logoff if any channels
+ are up until they hang up. (issue #9123 reported by arbrandes)
+
+ * channels/chan_sip.c, channel.c: Only change the original or clone
+ channel if it's the channel behind the proxy channel, not if it's
+ just a regular bridged channel.
+
+2007-02-22 00:53 +0000 [r56010] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: If we receive a frame that is not in any of
+ the negotiated formats, then drop it. (potentially issue #8781
+ and SPD-12)
+
+2007-02-21 20:32 +0000 [r55956] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_meetme.c: Change naughty warning message to provide
+ useful information. If a write now fails on a channel in meetme
+ it will tell you the channel name instead of spitting out the
+ wrong error message.
+
+2007-02-21 14:03 +0000 [r55868] Kevin P. Fleming <kpfleming at digium.com>
+
+ * build_tools/make_svn_branch_name: use new tag version script
+
+2007-02-21 00:19 +0000 [r55750] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c, utils.c, include/asterisk/lock.h: Fix random
+ crashes when using the MeetMe application. This patch converts
+ list handling to use the linked list macros and most importantly,
+ implements reference counting on the ast_conference objects. The
+ reference counting was first backported from 1.4. However, that
+ code has some problems that caused the reference count to never
+ hit zero. Those problems are fixed in this patch and will be
+ resolved in 1.4 and trunk next, with a different patch. (issues
+ #7647, #9073, #9106, BE-115).
+
+2007-02-20 22:39 +0000 [r55669] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_agent.c: Defer clearing callback information if
+ channels are up until they are hung up. This ensures the hangup
+ process goes smoothly and no channels get hung in limbo. (issue
+ #8088 reported by kebl0155)
+
+2007-02-20 19:49 +0000 [r55588] Russell Bryant <russell at digium.com>
+
+ * apps/app_page.c: Convert a tab to spaces so that the
+ documentation is printed out properly aligned.
+
+2007-02-19 18:09 +0000 [r55434] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: forcename and forcegreetings options should
+ check to see if the recording already exists
+
+2007-02-18 12:32 +0000 [r55249-55277] Olle Johansson <oej at edvina.net>
+
+ * apps/app_record.c: Documentation update (#9053, jsmith)
+
+ * channels/chan_sip.c: Issue #9020 - SIP message retransmission
+ time too short. Backporting fix implemented in 1.4, where we have
+ a minimum level for the T1 timer.
+
+2007-02-17 03:53 +0000 [r55073-55153] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c: Answer the channel before recording privacy
+ information. (issue #8926 reported by lmamane)
+
+ * channels/chan_sip.c: Allow chan_sip to handle attended transfers
+ from a SIP phone that is sitting behind chan_agent. Yes folks,
+ all it took was one line of code. (issue #8784 reported by
+ pzieba)
+
+2007-02-16 22:48 +0000 [r55005] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c, configs/meetme.conf.sample: Revert the change
+ I did in revisions 54955, 54969, and 54970, in 1.2, 1.4, and
+ trunk. I decided that once a conference is created from
+ meetme.conf, it is acceptable behavior that the pin can not be
+ changed until the conference goes away. I also added a note in
+ meetme.conf to describe this behavior. We still have another
+ issue in 1.4 and trunk where some conferences with no users don't
+ go away. That is the real bug that needs to be addressed here.
+
+2007-02-16 22:13 +0000 [r54999] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_agent.c: Do not send indications through
+ ast_indicate in chan_agent but instead go directly to the
+ technology. This way when indications are emulated they happen on
+ the Agent channel and do not screw up formats on the channels.
+ (issue #8439 reported by punkgode)
+
+2007-02-16 20:56 +0000 [r54955] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: For conferences that are configured in
+ meetme.conf, check the configuration file every time someone
+ joins the conference instead of only when the conference is first
+ created. This is to ensure that changes to the pin numbers in the
+ config file are always honored. (issue #9073)
+
+2007-02-16 11:38 +0000 [r54771] Olle Johansson <oej at edvina.net>
+
+ * res/res_agi.c: Issue #9069 - If we open with TH we should not
+ close with /TD. (seanbright)
+
+2007-02-15 16:14 +0000 [r54622] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c: Use a separate variable to indicate execution
+ should continue instead of the return value. (issue #8842
+ reported by pluto70)
+
+2007-02-14 18:40 +0000 [r54438] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c:
+
+2007-02-14 16:25 +0000 [r54373] Matt Frederickson <creslin at digium.com>
+
+ * channels/chan_zap.c: When handling glare on a PRI, move the
+ requested channel rather than hang up the old one. Fix for 8957
+ and 9011.
+
+2007-02-13 18:35 +0000 [r54179] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Make sure that outbound calls are applied to
+ the peer. This fixes some issues with "hints not working", but
+ only in 1.2.
+
+2007-02-08 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.2.15 released
+
+2007-02-08 22:17 +0000 [r53658] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/codec_zap.c: ensure channelcount is cleared before we
+ enumerate transcoders, so 'reload' doesn't double the channel
+ count
+
+2007-02-08 13:36 +0000 [r53529] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Issue 9003 - If fullname is empty, quote()
+ passes back "\""
+
+2007-02-07 15:38 +0000 [r53357] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: Fix a few potential memory leaks with
+ realtime users and peers. (issue #8999 reported by bsmithurst)
+
+2007-02-07 15:30 +0000 [r53354] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_macro.c: Issue 7440 - Macro called from Macro from the h
+ extension exits prematurely
+
+2007-02-06 06:58 +0000 [r53245] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * manager.c: Issue 8987 - Status could return two responses
+ (mnicholson)
+
+2007-02-06 00:08 +0000 [r53224] Jason Parker <jparker at digium.com>
+
+ * configs/dnsmgr.conf.sample: Add a proper newline at the end of
+ this sample config file.
+
+2007-02-03 20:39 +0000 [r53133-53134] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c, utils.c, include/asterisk/lock.h: Revert some
+ changes that accidentally got committed as a part of another fix.
+
+ * apps/app_dial.c, apps/app_meetme.c, utils.c,
+ include/asterisk/lock.h: set the DIALSTATUS variable to contain
+ "INVALIDARGS" when the dial application exits early because of
+ invalid arguments instead of just leaving it empty. (issue #8975)
+
+2007-02-02 16:58 +0000 [r53117] Joshua Colp <jcolp at digium.com>
+
+ * config.c: Pass the glob expanded filename to process_text_line so
+ that error messages contain the actual filename, not the original
+ include one. (issue #8959 reported by tzafrir)
+
+2007-02-01 23:14 +0000 [r53107] Jason Parker <jparker at digium.com>
+
+ * apps/app_chanspy.c: Fix a small typo. Synopsis lines shouldn't
+ have a newline
+
+2007-02-01 22:21 +0000 [r53095-53103] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Copy noncodeccapability over to the joint
+ variable so that telephone-event will get transmitted in the sent
+ INVITE.
+
+ * channels/chan_sip.c: Don't negotiate RFC2833 when not configured
+ to do so. (issue #8799 reported by mdu113)
+
+2007-02-01 21:12 +0000 [r53090] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: - Make sure we release call from call
+ counter before we destroy call (maybe #7744 and more) -
+ Backported by accident from 1.4
+
+2007-02-01 21:03 +0000 [r53084] Joshua Colp <jcolp at digium.com>
+
+ * res/res_musiconhold.c: Return previous behavior of having MOH
+ pick up where it was left off. (issue #8672 reported by
+ sinistermidget)
+
+2007-02-01 20:07 +0000 [r53069-53074] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * funcs/func_strings.c: Bug 8965
+
+ * funcs/func_strings.c, pbx.c: No wonder FIELDQTY doesn't work with
+ functions... the documentation in pbx.c was wrong
+
+2007-01-31 21:25 +0000 [r53045] Russell Bryant <russell at digium.com>
+
+ * channels/chan_mgcp.c, manager.c, channels/chan_zap.c,
+ pbx/pbx_spool.c, apps/app_meetme.c, apps/app_page.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_queue.c,
+ channels/chan_iax2.c, apps/app_rpt.c, cdr.c, pbx.c: Fix a bunch
+ of places where pthread_attr_init() was called, but
+ pthread_attr_destroy() was not.
+
+2007-01-31 18:58 +0000 [r53044] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/codec_zap.c: update to match modified transcoder API
+
+2007-01-31 17:41 +0000 [r53039] Russell Bryant <russell at digium.com>
+
+ * rtp.c: Use the proper format string to print unsigned values in
+ the rtp debug output. (issue #8954, wmis)
+
+2007-01-31 17:28 +0000 [r53034] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/Makefile: allow codec_zap to build again, now that
+ transcoder support is in zaptel 1.2
+
+2007-01-30 19:41 +0000 [r52857-52954] Russell Bryant <russell at digium.com>
+
+ * channel.c: Don't print a message indicating that we don't know
+ what to do with a proceeding control frame in
+ ast_request_and_dial(). We just need to ignore it. (reported by
+ JerJer on #asterisk-dev)
+
+ * asterisk.c: The SIGHUP handler was implemented to allow admins to
+ send SIGHUP to a running Asterisk process to reload the
+ configuration. However, doing the actual reload in the signal
+ handler itself is a very bad thing to do, because the reload
+ process includes calling non-reentrant functions such as
+ malloc/calloc/etc. If Asterisk is running in the background, then
+ the reload will happen immediately. However, if running in
+ console mode, the reload doesn't work until something is typed at
+ the console. That sort of defeats the purpose, but I don't see an
+ easy way to get around it at this point.
+
+ * codecs/Makefile: Comment out the parts in the Makefile that make
+ codec_zap get built. It will not yet build against zaptel 1.2, so
+ I am disabling it to prevent further bug reports until it gets
+ merged. (issue #8940)
+
+2007-01-30 14:38 +0000 [r52843] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed some
+ possible segfaults. also fixed an very important bug which occurs
+ on high load (when calls are very fast generated)
+
+2007-01-30 00:15 +0000 [r52762] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix the extraction of the timestamp from
+ video frames. It was using the mapping for a mini-frame instead
+ of a video-frame, which caused it to get invalid data. (issue
+ #8795, mihai)
+
+2007-01-29 23:39 +0000 [r52716] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_mixmonitor.c: Now that filename is part of the structure
+ and since it comes before postprocess... we have to add it to our
+ postprocess line. (reported on asterisk-dev by Boris Bakchiev)
+
+2007-01-29 16:48 +0000 [r52503] Jason Parker <jparker at digium.com>
+
+ * codecs/codec_zap.c: Use the correct zaptel header file location.
+ Currently, this will not build - transcoder support will be added
+ to zaptel later today.
+
+2007-01-27 02:09 +0000 [r52360-52415] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_queue.c: Make COMPLETECALLER and COMPLETEAGENT output to
+ queue_log follow documentation. (issue #7677 reported by amilcar)
+
+ * channels/chan_iax2.c: Make the last context entry read in the
+ dominant one. (issue #8918 reported by pj)
+
+2007-01-25 19:15 +0000 [r52162-52264] Joshua Colp <jcolp at digium.com>
+
+ * jitterbuf.c: Allow dequeueing of frames with negative timestamp
+ by moving jitterbuffer frames check to jb_next. (issue #8546
+ reported by harmen)
+
+ * apps/app_mixmonitor.c: Add another note about audio files being
+ played back to each bridged party. (issue #8718 reported by ppyy)
+
+2007-01-25 00:39 +0000 [r52137] Russell Bryant <russell at digium.com>
+
+ * apps/app_groupcount.c: Fix a seg fault when running this
+ application with no arguments from AGI. (issue #8905, junky)
+
+2007-01-24 17:43 +0000 [r52002] Steve Murphy <murf at digium.com>
+
+ * utils/check_expr.c, utils/Makefile: updated check_expr via 8322
+ (refactoring of expression checking impl); elfring contributed a
+ nice code reorg, I contributed some time to get it working again,
+ better messages
+
+2007-01-24 10:48 +0000 [r51966] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c: fixed the busy problem (dialstatus was not
+ busy when we called a busy extension)
+
+2007-01-24 00:57 +0000 [r51828-51843] Russell Bryant <russell at digium.com>
+
+ * channel.c: Fix an issue related to synchronization of recordings
+ when using Monitor(). The bug is a miscalculation of the amount
+ to seek the stream for writing to disk when the number of samples
+ coming in and out of a channel do not match up. (issue #8298,
+ #8887, report and patch by guillecabeza, patch files created and
+ testing done by whoiswes)
+
+ * apps/app_while.c: Don't set a new value for the END_ variable on
+ the channel before using the old value. If you do, it will lead
+ to accessing a memory address that has been free()'d. (issue
+ #8895, arkadia)
+
+2007-01-23 01:41 +0000 [r51512] Joshua Colp <jcolp at digium.com>
+
+ * res/res_musiconhold.c: Yield before reading from zaptel timing
+ source under Solaris so that other threads get a chance to do
+ things. (issue #7875 reported by bob)
+
+2007-01-22 19:39 +0000 [r51410] Russell Bryant <russell at digium.com>
+
+ * codecs/Makefile, codecs/codec_zap.c (added): Merge codec_zap
+ support for the transcoder card. This is a standalone codec
+ module so it will not affect anything else.
+
+2007-01-22 19:08 +0000 [r51359-51406] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_mixmonitor.c: Move filestream creation to Mixmonitor
+ loop. This will prevent a blank file from being created if no
+ frames ever pass through to be recorded. (issue #7589 reported by
+ steve_mcneil)
+
+ * channels/chan_h323.c: Explicitly declare what codecs are
+ supported by default globally since using a bitmask for all may
+ include ones we don't need. (issue #8357 reported by
+ gknispel_proformatique)
+
+2007-01-19 16:44 +0000 [r51300] Russell Bryant <russell at digium.com>
+
+ * asterisk.c: Fix a memory leak on command line tab completion. The
+ container for the matches was freed, but the individual matches
+ themselves were not. (issue #8851, arkadia)
+
+2007-01-18 23:47 +0000 [r51271] Dwayne M. Hubbard <dhubbard at digium.com>
+
+ * channels/chan_zap.c: issue 7877: chan_zap module reload does not
+ use default/initialized values on subsequent loads. Reset
+ configuration variables to default values prior to parsing
+ configuration file.
+
+2007-01-18 23:35 +0000 [r51269] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_zap.c: support echo cancellers that can handle 64ms
+ or 128ms of echo cancellation
+
+2007-01-18 21:11 +0000 [r51235-51255] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * stdtime/localtime.c: If a timezone is not specified, assume
+ localtime (instead of gmtime) (Issue #7748)
+
+ * contrib/scripts/vmdb.sql: Document all the fields, including the
+ indication that "uniqueid" should not be renamed.
+
+2007-01-17 21:17 +0000 [r51197] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Move the check for a failure of
+ ast_channel_alloc() to before locking the pvt structure again.
+ Otherwise, on a failure, this will cause a deadlock.
+
+2007-01-17 20:52 +0000 [r51158-51194] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * utils.c: When ast_strip_quoted was called with a zero-length
+ string, it would treat a NULL as if it were the quoting character
+ (and would thus return the string in memory immediately following
+ the passed-in string).
+
+ * doc/voicemail_odbc_postgresql.txt (added): Add documentation
+ walkthrough on getting Postgres to work with voicemail (from
+ Issue 8513)
+
+ * apps/app_voicemail.c: Postgres driver doesn't like a NULL pointer
+ when retrieving the length (Bug 8513)
+
+2007-01-16 17:36 +0000 [r51085-51145] Joshua Colp <jcolp at digium.com>
+
+ * res/res_features.c: Return previous behavior. ParkedCalls will be
+ able to do DTMF based transfers again. trunk however will get an
+ option to allow this to be set on/off. (issue #8804 reported by
+ nortex)
+
+ * channels/chan_zap.c: Add none as a valid callgroup/pickupgroup
+ option. I consider it a bug that it would inherit it all the way
+ down and not have any way to reset it to nothing - so that's why
+ it is in 1.2. (issue #8296 reported by gkloepfer)
+
+2007-01-15 23:09 +0000 [r50987] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_groupcount.c: Check return value before dereferencing
+ (Bug 8822)
+
+2007-01-15 20:44 +0000 [r50946] Matt O'Gorman <mogorman at digium.com>
+
+ * apps/app_voicemail.c: Solves issue with forwarding voicemails
+ from folders other than inbox. patch by anthonyl.
+
+2007-01-14 05:01 +0000 [r50781] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * db1-ast/recno/rec_utils.c, db1-ast/hash/hash_bigkey.c,
+ db1-ast/hash/hsearch.c, db1-ast/recno/rec_open.c,
+ db1-ast/recno/rec_delete.c, db1-ast/btree/bt_page.c,
+ db1-ast/hash/hash_buf.c, db1-ast/hash/hash_page.c,
+ db1-ast/recno/rec_close.c, db1-ast/recno/rec_search.c,
+ db1-ast/btree/bt_get.c, db1-ast/hash/hash.c,
+ db1-ast/recno/rec_put.c, db1-ast/include/ndbm.h, db1-ast/db/db.c,
+ db1-ast/btree/bt_debug.c, db1-ast/mpool/mpool.c,
+ db1-ast/btree/bt_seq.c, db1-ast/recno/rec_get.c,
+ db1-ast/btree/bt_split.c, db1-ast/hash/hash_func.c,
+ db1-ast/btree/bt_utils.c, db1-ast/btree/bt_open.c,
+ db1-ast/recno/rec_seq.c, db1-ast/btree/bt_delete.c,
+ db1-ast/btree/bt_overflow.c, db1-ast/hash/hash_log2.c,
+ db1-ast/btree/bt_search.c, db1-ast/btree/bt_conv.c,
+ db1-ast/btree/bt_close.c, db1-ast/btree/bt_put.c: Bug 8814 - db
+ should look for its header using a relative path, instead of the
+ system path (Fixes FreeWRT)
+
+2007-01-12 14:34 +0000 [r50561] Kevin P. Fleming <kpfleming at digium.com>
+
+ * pbx.c: minor documentation clarification
+
+2007-01-11 18:11 +0000 [r50517] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Issue #8793 bad response for Unsupported
+ Extension (different fix).
+
+2007-01-11 14:45 +0000 [r50495-50506] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/isdn_lib.c: when we get L2 UP, the L1 is UP
+ definitely too, so we set the L1 state up as well.
+
+ * channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c,
+ channels/misdn/isdn_lib.c: * more additions to make the RESTART
+ message work * added fix for misdn_call to allow SETUPs with
+ empty extensions, replaced the strtok_r functions with strsep for
+ that (inspired by Sandro Cappellazzo, thanks)
+
+2007-01-10 09:51 +0000 [r50335] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/fac.c, channels/misdn/ie.c,
+ channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: more
+ fixes regarding warnings for gcc-4 and first additions for the
+ restart Information element, in the first step we initiate a
+ restart with a CLI command
+
+2007-01-10 04:51 +0000 [r50295] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c: Add another return value to dial_exec_full that
+ indicates execution is going to continuing at a new
+ extension/context/priority and to just let it slide. (issue #8598
+ reported by jon)
+
+2007-01-10 02:16 +0000 [r50227] Russell Bryant <russell at digium.com>
+
+ * Makefile: Make the number that represents the major version
+ number a single digit instead of 2. Using two digits makes it an
+ octal number when put into version.h, which breaks the
+ compilation of any out of tree module that checks the version for
+ any version after 1.2.7 (reported by Matteo Brancaleoni on the
+ asterisk-dev mailing list, who gave credit to vihai for pointing
+ it out)
+
+2007-01-09 13:30 +0000 [r50150] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: The advent of realtime has enabled people
+ to use commas in the fullname field. This could cause an issue
+ with sending voicemails, when the field is unquoted. (Issue 8595)
+
+2007-01-08 08:37 +0000 [r49922] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/isdn_lib.h, channels/misdn/ie.c,
+ channels/misdn/isdn_lib.c: make gcc 4 happy, remove some warnings
+
+2007-01-08 05:10 +0000 [r49889] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: Ensure we use the default refresh value of
+ 60 if the remote server does not send one. (issue #8746 reported
+ by maethor)
+
+2007-01-07 21:43 +0000 [r49833] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_dictate.c: If openstream fails, then we crash (Issue
+ 8564)
+
+2007-01-05 16:56 +0000 [r49635] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_iax2.c: ensure that threads which are supposed to
+ be detached (because we aren't going to wait on them) are created
+ properly
+
+2007-01-04 17:45 +0000 [r49354-49447] Matt O'Gorman <mogorman at digium.com>
+
+ * apps/app_voicemail.c: converted a lot of 256 to PATH_MAX and some
+ white space fixes.
+
+ * apps/app_voicemail.c: good catch russell sorry i missed that. fix
+ magic number with proper sizeof
+
+ * apps/app_voicemail.c: When using ODBC_STORAGE VoicemailMain
+ doesn't create the subdirectories for a mailbox such as the INBOX
+ directory. this patch solves that problem, was written by anthony
+ be-125
+
+2007-01-03 08:24 +0000 [r49135-49303] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: * Added check
+ for bridging in misdn_call to avoid setting echocancellation when
+ 2 mISDN channels are involved and when bridging is set. That lead
+ to a kernel panic before under different situations, because we
+ switched about 2 times between hardware bridging and
+ echocancelation * readded MISDN_URATE variable which got lost
+ before, this should make app_v110 work again * fixed typo
+
+ * channels/misdn_config.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, configs/misdn.conf.sample,
+ channels/misdn/isdn_lib.c: added check for channel ranges in the
+ set/empty channel functions. set pmp_l1_check default to no.
+ added misdn restart pid cli command. added cleaning of channel
+ when we send a RELEASE_COMPLETE.
+
+2006-12-29 00:32 +0000 [r49045] Kevin P. Fleming <kpfleming at digium.com>
+
+ * BUGS: location of the bug posting guidelines has changed
+
+2006-12-27 15:43 +0000 [r48974] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Issue 8596: Set CAN_BYE flag for 100 trying
+ too
+
+2006-12-25 05:19 +0000 [r48939-48955] Russell Bryant <russell at digium.com>
+
+ * funcs/func_math.c: Fix an error introduced by copying and pasting
+ the handling of the >= operator for the MATH function. If a
+ single equal sign was used as an operator, the function would
+ treat it is as if it were the >= operator. Now, it properly
+ handles it as an invalid operator. (issue #8665, patch by
+ tempest1)
+
+ * channels/chan_iax2.c: Check for the proper return value on an
+ error in a call to mmap(). This was reported by Andy Wang on the
+ asterisk-dev list. Thanks!
+
+ * channels/chan_sip.c: Remove a couple of misplaced dots in log
+ messages. This was reported by Andrea Spadaccini on the
+ asterisk-dev mailing list.
+
+2006-12-21 20:25 +0000 [r48782] Joshua Colp <jcolp at digium.com>
+
+ * redhat/asterisk.spec: Add new silence sound files to the spec for
+ Redhat. (issue #8652 reported by alvaro_palma_aste)
+
+2006-12-19 21:10 +0000 [r48584] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c: Free localuser structure when we fail to dial
+ (issue #8612 reported by rizzo)
+
+2006-12-19 13:08 +0000 [r48576] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/isdn_lib.c: when we reject a channel, because it's
+ in use already, we shouldn't process the setup anymore. made the
+ channel allocation a bit easier and more understandable, removed
+ a few unused lines
+
+2006-12-18 10:19 +0000 [r48552] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: when our PTP
+ Partner sends us a SETUP with a preselected channel we just
+ accept it, even when we're NT. added some checks for segfaults.
+
+2006-12-15 10:51 +0000 [r48484] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Issue #8592 - handle 504 as 503 - congestion
+
+2006-12-14 13:03 +0000 [r48467] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: removed FIXUP
+ state. added check for channel allocation conflict when we create
+ a setup while the other site creates a setup on the same channel,
+ besides the check we resolve this conflict.
+
+2006-12-14 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.2.14 released
+
+2006-12-13 04:23 +0000 [r48434] Steve Murphy <murf at digium.com>
+
+ * channel.c: This small patch fixes bug 8541, where the L option to
+ the Dial app wasn't working right. A similar bug (8386) was filed
+ and fixed earlier, but an intervening bug fix to a DTMF problem
+ broke the L() code in a different way. Hopefully, everything is
+ happy now.
+
+2006-12-12 05:11 +0000 [r48403] Kevin P. Fleming <kpfleming at digium.com>
+
+ * sounds/silence (added), sounds/silence/1.gsm (added),
+ sounds/silence/10.gsm (added), sounds/silence/2.gsm (added),
+ sounds/silence/3.gsm (added), sounds/silence/4.gsm (added),
+ sounds/silence/5.gsm (added), sounds/silence/6.gsm (added),
+ sounds/silence/7.gsm (added), sounds/silence/8.gsm (added),
+ sounds/silence/9.gsm (added): add silence files
+
+2006-12-11 23:00 +0000 [r48394-48398] Matt O'Gorman <mogorman at digium.com>
+
+ * Makefile, apps/app_externalivr.c, sounds.txt: app_externalivr
+ needs a real silence file, and additional changes to add silence
+ files into core instead of extra patch provided by bug 8177 with
+ minor additions.
+
+2006-12-11 00:33 +0000 [r48374] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_nbscat.c, apps/app_festival.c, apps/app_mp3.c,
+ res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
+ apps/app_ices.c, res/res_musiconhold.c: When doing a fork() and
+ exec(), two problems existed (Issue 8086): 1) Ignored signals
+ stayed ignored after the exec(). 2) Signals could possibly fire
+ between the fork() and exec(), causing Asterisk signal handlers
+ within the child to execute, which caused nasty race conditions.
+
+2006-12-10 02:14 +0000 [r48371] Steve Murphy <murf at digium.com>
+
+ * channels/chan_zap.c: This version applies the patch suggested by
+ stevens in bug 7836 (make inbound channel RINGING state
+ consistent with other channels).
+
+2006-12-09 15:45 +0000 [r48361] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Use locking when accessing the
+ registrations list. This list is not actually used very often, so
+ the likelihood of there being a problem is pretty small, but
+ still possible. For example, if the CLI command to list the
+ registrations was called at the same time that a reload was
+ occurring and the registrations list was getting destroyed and
+ rebuilt, a crash could occur.
+
+2006-12-07 18:14 +0000 [r48356] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: Ensure that the file position is not
+ incremented beyond the total number of files available for
+ playback. (issue #8539, ulogic)
+
+2006-12-06 16:05 +0000 [r48322] Russell Bryant <russell at digium.com>
+
+ * configs/iax.conf.sample: Fix the name of the rtignoreregexpire
+ option in the sample configuration file. (issue #8526, arkadia)
+
+2006-12-06 15:48 +0000 [r48321] Christian Richter <christian.richter at beronet.com>
+
+ * doc/README.misdn, channels/chan_misdn.c,
+ channels/misdn/isdn_msg_parser.c: added the export and import of
+ the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the
+ extension is already completely dialed or if there might come
+ additional digits by information elements. also added some docs
+ for that.
+
+2006-12-06 15:42 +0000 [r48320] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Issue #8528 - make sure we don't delete the
+ dialog too quickly after receiving a 487. Move 487 handling into
+ handle_response_invite where it really belongs and don't add an
+ ALREADYGONE flag to the dialog.
+
+2006-12-06 14:35 +0000 [r48319] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c: changed a few debugs to higher debug
+ levels
+
+2006-12-06 12:14 +0000 [r48272-48315] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Don't add Contact header on BYE, CANCEL,
+ MESSAGE requests (Bye, Cancel backported from 1.4, MESSAGE bug
+ reported to me by Gunnar at Omnitor)
+
+ * channels/chan_sip.c: Only set the ALREADYGONE flag once in
+ handle_response()
+
+2006-12-05 01:26 +0000 [r48251] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: If the recording in the database is too
+ large, it will fail to retrieve with an mmap error. Not too sure
+ why this doesn't happen when we put it in the database, also, but
+ since that doesn't seem to be broken, I'm not going to fix it (at
+ least until someone reports it). Solution is to ask for the file
+ in smaller chunks. (Bug 8385)
+
+2006-12-04 21:20 +0000 [r48236-48246] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.c: Revert change from 8016 - this breaks other
+ stuff... Needs further review. Tip: When you've reported a bug
+ about something and somebody has put up a patch for it.. It's not
+ a good idea to open a completely new bug and say that something
+ is broken because of the patch in the other bug - PLEASE mention
+ something in the bug where the patch was actually created.
+
+ * apps/app_voicemail.c: Fix an issue where a message isn't saved
+ correctly when using ODBC storage and reviewing a message. Issue
+ 8016 - patch by sokhapkin.
+
+2006-12-04 18:14 +0000 [r48233] Joshua Colp <jcolp at digium.com>
+
+ * channel.c: If the generic bridge tells us not to retry, and we
+ have a frame to spit out then break the bridge. Props to markit
+ in #asterisk-bugs for bringing this up.
+
+2006-12-01 23:30 +0000 [r48192] Kevin P. Fleming <kpfleming at digium.com>
+
+ * apps/app_dial.c: if Dial() is going to send music-on-hold to the
+ calling party, it has to send PROGRESS first to ensure that the
+ reverse audio path has been setup first (BE-106)
+
+2006-12-01 20:19 +0000 [r48183] Jason Parker <jparker at digium.com>
+
+ * configs/extensions.conf.sample: Fix a small typo - issue 8848,
+ reported by pabelanger
+
+2006-11-30 20:47 +0000 [r48165] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Issue 8319 - noriyuki - nonce-count updated
+ *after* use
+
+2006-11-30 20:27 +0000 [r48142-48161] Joshua Colp <jcolp at digium.com>
+
+ * channel.c: Don't write AST_FRAME_NULL or AST_FRAME_IAX frames out
+ to the channel driver. (issue #8390 reported by hselasky)
+
+ * channels/chan_iax2.c: Only print out debug message if bridged
+ channel is not NULL. (issue #8412 reported by jubilex)
+
+ * res/res_features.c: Do not listen for DTMF on the bridge that
+ comes into existence when ParkedCall is executed. This means
+ native bridging can now occur for this. (issue #8406 reported by
+ kebl0155)
+
+ * cdr.c: Print certain CDR messages out at the NOTICE level versus
+ WARNING since they can occur when used with the CDR applications
+ and are perfectly fine. (issue #8367 reported by dartvader)
+
+ * res/res_features.c: Remember the pointer to the allocated block
+ of memory so that we can free it and not cause a memory leak.
+ (issue #8449 reported by arkadia)
+
+ * configs/sip.conf.sample: Document 'port' for SIP peers, came up
+ because of the current mailing list thread. (issue #8450 reported
+ by blitzrage)
+
+2006-11-30 09:05 +0000 [r48127] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Do proper test and don't leave dialogs
+ hanging...
+
+2006-11-29 16:47 +0000 [r48053-48106] Joshua Colp <jcolp at digium.com>
+
+ * rtp.c: If the frame was duplicated before writing out then we
+ need to free it. (issue #8429 reported by edguy3)
+
+ * channels/chan_phone.c: According to the research I have done we
+ never needed to include compiler.h in the first place so let's
+ not! (issue #8430 reported by edguy3)
+
+ * apps/app_voicemail.c: Use the proper function to get the new
+ message count instead of always using the filesystem. (issue
+ #8421 reported by slimey)
+
+2006-11-27 17:15 +0000 [r48045] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * res/res_musiconhold.c: Random MOH wasn't really random (bug 8381)
+
+2006-11-27 15:30 +0000 [r48037] Joshua Colp <jcolp at digium.com>
+
+ * pbx/pbx_spool.c: Do not reference the freed outgoing structure in
+ the debug message. (issue #8425 reported by arkadia)
+
+2006-11-24 14:33 +0000 [r47987] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Change some logging levels. Not having hints
+ is not an ERROR, but still should be reported.
+
+2006-11-23 16:10 +0000 [r47968] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c,
+ channels/misdn/isdn_lib.c: fixed a litle bug regarding
+ HOLD/RETRIEVE. beatufied some logs, changed some loglevels.
+ changed the default value of block_on_alarm
+
[... 4088 lines stripped ...]
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