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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-06-30 16:17:41 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-06-30 16:17:41 +0000
commit9f549d191ee90c848a3676c999814b8968366bd2 (patch)
tree634c7d399de9244365fdf4068cb460f95355cfa5
parenta65b6c05c866c36c220563224918825c11f8a3dd (diff)
remove version files so the mkrelease script can create them
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.21.1@126579 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--.version1
-rw-r--r--ChangeLog18100
2 files changed, 0 insertions, 18101 deletions
diff --git a/.version b/.version
deleted file mode 100644
index 93f7be0b7..000000000
--- a/.version
+++ /dev/null
@@ -1 +0,0 @@
-1.4.21.1
diff --git a/ChangeLog b/ChangeLog
deleted file mode 100644
index 6689b1f08..000000000
--- a/ChangeLog
+++ /dev/null
@@ -1,18100 +0,0 @@
-2008-06-30 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.21.1 released.
-
-2008-06-30 Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: Fix a typo that introduced a number of
- deadlocks, most commonly in chan_iax2, when not running with
- DEBUG_THREADS enabled.
-
-2008-06-12 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.21 released.
-
-2008-06-06 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.21-rc2 released.
-
-2008-06-05 18:03 +0000 [r120731-120735] Russell Bryant <russell@digium.com>
-
- * UPGRADE-1.2.txt: fix filename
-
- * UPGRADE-1.2.txt (added), UPGRADE.txt: Add the UPGRADE.txt file
- from Asterisk 1.2, for handy reference.
-
-2008-06-05 16:56 +0000 [r120675] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Ignore appended resource when comparing JIDs.
-
-2008-06-05 16:38 +0000 [r120671] Russell Bryant <russell@digium.com>
-
- * doc/smdi.txt, res/res_smdi.c: It turns out that searching on the
- forwarding station isn't very useful for most people, so pull in
- the changes that allow searching for SMDI messages based on other
- components of the SMDI message. Also, update the SMDI
- documentation.
-
-2008-06-04 22:05 +0000 [r120513] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Make sure that the string we set will survive
- the unref of the queue member. Thanks to Russell, who pointed
- this out.
-
-2008-06-04 18:35 +0000 [r120425] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: If we fail to setup the PRI request channel,
- don't continue, exit with an error. (closes issue #11989)
- Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt
- uploaded by Corydon76 (license 14)
-
-2008-06-04 16:26 +0000 [r120371] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_config.c: Make the "dialplan remove include" CLI command
- actually work. Also, tweak some formatting, and make the success
- message a little bit more clear. (closes AST-52)
-
-2008-06-04 14:11 +0000 [r120285] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Tab completion when removing a member should
- give the member's interface, not the name, since the interface is
- what is expected for the command. (closes issue #12783) Reported
- by: davevg
-
-2008-06-04 13:31 +0000 [r120282] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, pbx/pbx_config.c: Fix a log message and add a message
- for when the dialplan is done reloading. (closes issue #12716)
- Reported by: chappell Patches: dialplan_reload_2.diff uploaded by
- chappell (license 8)
-
-2008-06-03 22:41 +0000 [r120226] Tilghman Lesher <tlesher@digium.com>
-
- * pbx/pbx_loopback.c: Due to incorrect use of the
- AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform
- any translation on the extension number before searching for it
- in the target context. (closes issue #12473) Reported by:
- chappell Patches: pbx_loopback.c.diff uploaded by chappell
- (license 8)
-
-2008-06-03 22:15 +0000 [r120173] Jeff Peeler <jpeeler@digium.com>
-
- * main/config.c: (closes issue #11594) Reported by: yem Tested by:
- yem This change decreases the buffer size allocated on the stack
- substantially in config_text_file_load when LOW_MEMORY is turned
- on. This change combined with the fix from revision 117462
- (making mkintf not copy the zt_chan_conf structure) was enough to
- prevent the crash.
-
-2008-06-03 21:34 +0000 [r120168] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix another place where peer->callno could
- change at a very bad time, and also fix a place where a peer was
- used after the reference was released. (inspired by rev 120001)
-
-2008-06-03 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.21-rc1 released.
-
-2008-06-03 18:23 +0000 [r120001-120061] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c: When listing the manager users, managers in
- users.conf are not shown, even though they are allowed to
- connect. (closes issue #12594) Reported by: bkruse Patches:
- 12594-managerusers-2.diff uploaded by qwell (license 4) Tested
- by: bkruse
-
- * channels/chan_iax2.c: Save the callno when we're poking, because
- our peer structure could change during deadlock avoidance (and
- thus we unlock the wrong callno, causing a cascade failure).
- (closes issue #12717) Reported by: gewfie Patches:
- 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
- Tested by: gewfie
-
-2008-06-03 15:26 +0000 [r119929-119966] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-vtest13,
- 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-test3, pbx/ael/ael-test/ref.ael-test5,
- pbx/ael/ael-test/ref.ael-test15: Updated the regressions on AEL.
- Hadn't updated this for the changes I made to preserve ${EXTEN}
- in switches, which affected several tests because it adds extra
- priorities, and at least one needed to be updated because of the
- removal of the empty extension warning message.
-
- * pbx/pbx_ael.c: as per
- http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
- which is a message from Philipp Kempgen, requesting that the
- WARNING that an extension is empty be reduced to a NOTICE or
- less, as empty extensions are syntactically possible, and no big
- deal. With which I agree, and have removed that WARNING message
- entirely. I think it is not necessary to see this message. It
- didn't state that a NoOp() was inserted automatically on your
- behalf, and really, as users, who cares? Why freak out dialplan
- writers with unnecessary warnings? The details of the
- machinations a compiler goes thru to produce working assembly
- code is of little interest to most programmers-- we will follow
- the unix principal of doing our work silently.
-
-2008-06-03 14:46 +0000 [r119926] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Treat ECONNREFUSED as an error that will
- stop further retransmissions. (issue #AST-58, patch from
- Switchvox)
-
-2008-06-02 20:08 +0000 [r119742-119838] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Revert a change made for issue #12479. This
- change caused a regression such that a dial string such as
- (IAX2/foo) did not automatically fall back to dialing the 's'
- extension anymore. (closes issue #12770) Reported by: dagmoller
-
- * main/manager.c: Improve CLI command blacklist checking for the
- command manager action. Previously, it did not handle case or
- whitespace properly. This made it possible for blacklisted
- commands to get executed anyway. (closes issue #12765)
-
-2008-06-02 14:32 +0000 [r119740] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c, res/res_jabber.c: Do not link the guest
- account with any configured XMPP client (in jabber.conf). The
- actual connection is made when a call comes in Asterisk. Fix the
- ast_aji_get_client function that was not able to retrieve an XMPP
- client from its JID. (closes issue #12085) Reported by: junky
- Tested by: phsultan
-
-2008-06-02 12:30 +0000 [r119687] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Even of the first PING or LAGRQ doesn't get
- sent because it comes up too soon, make sure to reschedule so it
- gets sent later.
-
-2008-06-02 09:29 +0000 [r119585-119636] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c: fixed compile issue when dev-mode is
- enabled
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Added
- counter for unhandled_bmsg Print, this prevents the logs to be
- flooded to fast and save CPU in this error scenario. Added
- 'last_used' element to bc structure, when a bchannel changes from
- used to free this exact time will be marked in last_used. When a
- new channel is requested the find_free_chan function will check
- if the new empty channel was used within the last second, if yes
- it will search for the next channel, if no it will return this
- channel. This simple mechanism has prooven to prevent race
- conditions where the NT and TE tried to allocate the exact same
- channel at the same time (RELEASE cause: 44).
-
-2008-06-02 01:06 +0000 [r119530-119533] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Change a debug message to an actual debug
- message
-
- * apps/app_dial.c: Fix another typo in documentation
-
-2008-06-01 20:47 +0000 [r119478] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_dial.c: small typo fix 'retires' => 'retries'
-
-2008-05-30 21:17 +0000 [r119404] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c: When joinempty=strict, it only failed on join
- if there were busy members. If all members were logged out OR
- paused, then it (incorrectly) let callers join the queue. (closes
- issue #12451) Reported by: davidw
-
-2008-05-30 19:46 +0000 [r119354] Joshua Colp <jcolp@digium.com>
-
- * main/autoservice.c: Fix a bug I found while testing for another
- issue.
-
-2008-05-30 16:44 +0000 [r119301] Michiel van Baak <michiel@vanbaak.info>
-
- * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
- contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.slackware.asterisk: dont use a bashism way to
- check the $VERSION variable. The rc/init.d scripts, and
- safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2
- (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski)
- (closes issue #12687) Reported by: loloski Patches:
- 20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by
- mvanbaak (license 7) Tested by: loloski, mvanbaak
-
-2008-05-30 12:55 +0000 [r119076-119238] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 119237 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30
- May 2008) | 7 lines - Instead of only enforcing destination call
- number checking on an ACK, check all full frames except for PING
- and LAGRQ, which may be sent by older versions too quickly to
- contain the destination call number. (As suggested by Tim Panton
- on the asterisk-dev list) - Merge changes from
- team/russell/iax2-frame-race, which prevents PING and LAGRQ from
- being sent before the destination call number is known. ........
-
- * main/autoservice.c: Fix a race condition in channel autoservice.
- There was still a small window of opportunity for a DTMF frame,
- or some other deferred frame type, to come in and get dropped.
- (closes issue #12656) (closes issue #12656) Reported by: dimas
- Patches: v3-12656.patch uploaded by dimas (license 88) -- with
- some modifications by me
-
- * include/asterisk/audiohook.h: Oddly enough, all of the contents
- of audiohook.h were in there twice. I have removed the second
- copy.
-
-2008-05-29 20:24 +0000 [r119071] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: Call waiting tone occurs too often, because
- it's getting serviced by both subchannels. (closes issue #11354)
- Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded
- by Corydon76 (license 14)
-
-2008-05-29 19:04 +0000 [r118956-119012] Russell Bryant <russell@digium.com>
-
- * apps/app_milliwatt.c: - Fix a typo in the argument to Playtones -
- use ast_safe_sleep() instead of calling the wait application
- (thanks to tilghman for pointing these out!)
-
- * /, channels/chan_iax2.c: Merged revisions 119008 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29
- May 2008) | 7 lines Merge changes from
- team/russell/iax2-another-fix-to-the-fix As described in the
- following post to the asterisk-dev mailing list, only enforce
- destination call numbers when processing an ACK.
- http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
- (closes issue #12631) ........
-
- * apps/app_milliwatt.c: - Mark app_milliwatt dependent on
- res_indications (thanks to jsmith) - fix a typo in a log message
- (thanks to qwell)
-
- * apps/app_milliwatt.c: Change milliwatt to use the proper tone by
- default (1004 Hz) instead of 1000 Hz. An option is there to use
- 1000 Hz for anyone that might want it.
-
-2008-05-29 17:33 +0000 [r118953-118954] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Define also when not DEBUG_THREADS
-
- * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
- channels/chan_agent.c, channels/chan_alsa.c, main/utils.c,
- include/asterisk/lock.h, channels/chan_iax2.c: Add some debugging
- code that ensures that when we do deadlock avoidance, we don't
- lose the information about how a lock was originally acquired.
-
-2008-05-29 00:25 +0000 [r118858] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, apps/app_forkcdr.c: (closes issue #10668) (closes
- issue #11721) (closes issue #12726) Reported by: arkadia Tested
- by: murf These changes: 1. revert the changes made via bug 10668;
- I should have known that such changes, even tho they made sense
- at the time, seemed like an omission, etc, were actually integral
- to the CDR system via forkCDR. It makes sense to me now that
- forkCDR didn't natively end any CDR's, but rather depended on
- natively closing them all at hangup time via traversing and
- closing them all, whether locked or not. I still don't completely
- understand the benefits of setvar and answer operating on locked
- cdrs, but I've seen enough to revert those changes also, and stop
- messing up users who depended on that behavior. bug 12726 found
- reverting the changes fixed his changes, and after a long review
- and working on forkCDR, I can see why. 2. Apply the suggested
- enhancements proposed in 10668, but in a completely compatible
- way. ForkCDR will behave exactly as before, but now has new
- options that will allow some actions to be taken that will
- slightly modify the outcome and side-effects of forkCDR. Based on
- conversations I've had with various people, these small tweaks
- will allow some users to get the behavior they need. For
- instance, users executing forkCDR in an AGI script will find the
- answer time set, and DISPOSITION set, a situation not covered
- when the routines were first written. 3. A small problem in the
- cdr serializer would output answer and end times even when they
- were not set. This is now fixed.
-
-2008-05-28 16:10 +0000 [r118716] Brett Bryant <bbryant@digium.com>
-
- * channels/chan_iax2.c: merge revision 118702 from trunk to 1.4 --
- Fixes a bug in chan_iax that uses send_command to poke a peer
- while a channel is unlocked in some cases, and because it can
- cause seemingly random failures could be related to some bugs in
- the tracker...
-
-2008-05-28 14:23 +0000 [r118558-118646] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add an
- option to use the source IP address of RTP as the destination IP
- address of UDPTL when a specific option is enabled. If the remote
- side is properly configured (ports forwarded) then UDPTL will
- flow. (closes issue #10417) Reported by: cstadlmann
-
- * channels/chan_sip.c: Fix an issue where codec preferences were
- not set on dialogs that were not authenticated via a user or peer
- and allow framing to work without rtpmap in the SDP. (closes
- issue #12501) Reported by: slimey
-
-2008-05-27 19:15 +0000 [r118551] Tilghman Lesher <tlesher@digium.com>
-
- * main/cli.c: When showing an error message for a command, don't
- shorten the command output, as it tends to confuse the user (it's
- fine for suggesting other commands, however). Reported by:
- seanbright (on #asterisk-dev) Fixed by: me
-
-2008-05-27 19:07 +0000 [r118509] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Russell noted to me that in the case that
- separate threads use their own addressing system, the fix I made
- for issue 12376 does not guarantee uniqueness to the datastores'
- uids. Though I know of no system that works this way, I am going
- to change this right now to prevent trying to track down some
- future bug that may occur and cause untold hours of debugging
- time to track down. The change involves using a global counter
- which increases with each new chanspy_ds which is created. This
- guarantees uniqueness.
-
-2008-05-27 18:58 +0000 [r118465] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: NULL character should terminate only commands
- back to the core, not log messages to the console. (closes issue
- #12731) Reported by: seanbright Patches:
- 20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
- Tested by: seanbright
-
-2008-05-27 17:17 +0000 [r118416] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_voicemail.c: small update to the g() option of
- app_voicemail to note that gain changes only work on zap channels
- right now. issue #12578 shows it's not clear right now.
-
-2008-05-27 16:38 +0000 [r118365] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Add a unique id to the datastore allocated in
- app_chanspy since it is possible that multiple spies may be
- listening to the same channel. (closes issue #12376) Reported by:
- DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut
- (license 60) Tested by: destiny6628 (closes issue #12243)
- Reported by: atis
-
-2008-05-27 15:45 +0000 [r118358] Tilghman Lesher <tlesher@digium.com>
-
- * configs/queues.conf.sample: Add a note that pbx_config.so is
- needed for Local channels. (Closes issue #12671)
-
-2008-05-25 16:02 +0000 [r118251] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Realtime flag affects construction in
- multiple ways, so consulting whether rtcachefriends was set was
- done too soon (needed to be done inside build_peer, not just as a
- flag to build_peer). Also, fullcontact needed to be
- reconstructed, because realtime separates the embedded ';' into
- multiple fields. (closes issue #12722) Reported by: barthpbx
- Patches: 20080525__bug12722.diff.txt uploaded by Corydon76
- (license 14) Tested by: barthpbx (Much of the discussion happened
- on #asterisk-dev for diagnosing this issue)
-
-2008-05-23 21:21 +0000 [r118163] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: Fix a few things I missed to ensure
- zt_chan_conf structure is not modified in mkintf
-
-2008-05-23 13:18 +0000 [r118052-118055] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/utils.h: Add format type checking for recently
- de-inlined function
-
- * doc/cli.txt (added), doc/00README.1st: Add information on using
- the Asterisk console, including tab command line completion.
- (Closes issue #12681)
-
-2008-05-23 12:30 +0000 [r118048] Russell Bryant <russell@digium.com>
-
- * include/asterisk/utils.h, main/utils.c: Don't declare a function
- that takes variable arguments as inline, because it's not valid,
- and on some compilers, will emit a warning.
- http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes
- issue #12289) Reported by: francesco_r Patches by Tilghman, final
- patch by me
-
-2008-05-22 18:53 +0000 [r117809-117899] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: Also remove preamble from asynchronous events
- (reported by jsmith on #asterisk-dev)
-
- * funcs/func_realtime.c: Take into account the length of delimiters
- when calculating result string length. (closes issue #12696)
- Reported by: adomjan Patches: func_realtime.c-longdelimiter.patch
- uploaded by adomjan (license 487)
-
-2008-05-21 20:11 +0000 [r117582] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: Ensure that passed in zt_chan_conf structure
- is not modified in mkintf.
-
-2008-05-21 19:38 +0000 [r117574] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Apply the autoframing setting to dialogs
- that do not get matched against a user or peer.
-
-2008-05-21 18:44 +0000 [r117519-117523] Tilghman Lesher <tlesher@digium.com>
-
- * pbx/pbx_spool.c: Revert accidental commit of the last change
-
- * main/asterisk.c, pbx/pbx_spool.c: Strip the preamble from the
- output also when -rx is not being used (Related to issue #12702)
-
-2008-05-21 18:28 +0000 [r117479-117514] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Don't filter the magic character in the network
- verboser. It gets filtered once it reaches the client. (related
- to issue #12702, pointed out by tilghman)
-
- * main/asterisk.c, pbx/pbx_gtkconsole.c: 1) Don't print the verbose
- marker in front of every message from ast_verbose() being sent to
- remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose
- marker. (related to issue #12702)
-
- * main/asterisk.c: Don't display the verbose marker for calls to
- ast_verbose() that do not include a VERBOSE_PREFIX in front of
- the message. (closes issue #12702) Reported by: johnlange Patched
- by me
-
-2008-05-21 16:58 +0000 [r117462] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: Pass a pointer for the conf parameter to the
- function mkintf rather than the whole zt_chan_conf structure.
-
-2008-05-20 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.20 released.
-
-2008-05-14 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.20-rc3 released.
-
-2008-05-14 12:51 +0000 [r116230] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Accept text messages even with Content-Type:
- text/plain;charset=Södermanländska
-
-2008-05-13 23:47 +0000 [r116088] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, include/asterisk/lock.h: A change to the way
- channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
- After debugging a deadlock, it was noticed that when
- DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin
- of channel locks is obscured by the fact that all channel locks
- appear to happen in the function ast_channel_lock(). This code
- change redefines ast_channel_lock to be a macro which maps to
- __ast_channel_lock(), which then relays the proper file name,
- line number, and function name information to the core lock
- functions so that this information will be displayed in the case
- that there is some sort of locking error or core show locks is
- issued.
-
-2008-05-13 21:17 +0000 [r115990-116038] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c: Fix a deadlock involving channel
- autoservice and chan_local that was debugged and fixed by
- mmichelson and me. We observed a system that had a bunch of
- threads stuck in ast_autoservice_stop(). The reason these threads
- were waiting around is because this function waits to ensure that
- the channel list in the autoservice thread gets rebuilt before
- the stop() function returns. However, the autoservice thread was
- also locked, so the autoservice channel list was never getting
- rebuilt. The autoservice thread was stuck waiting for the channel
- lock on a local channel. However, the local channel was locked by
- a thread that was stuck in the autoservice stop function. It
- turned out that the issue came down to the local_queue_frame()
- function in chan_local. This function assumed that one of the
- channels passed in as an argument was locked when called.
- However, that was not always the case. There were multiple cases
- in which this channel was not locked when the function was
- called. We fixed up chan_local to indicate to this function
- whether this channel was locked or not. The previous assumption
- had caused local_queue_frame() to improperly return with the
- channel locked, where it would then never get unlocked. (closes
- issue #12584) (related to issue #12603)
-
- * main/autoservice.c: Fix an issue that I noticed in autoservice
- while mmichelson and I were debugging a different problem. I
- noticed that it was theoretically possible for two threads to
- attempt to start the autoservice thread at the same time. This
- change makes the process of starting the autoservice thread,
- thread-safe.
-
-2008-05-13 20:28 +0000 [r115944] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_alsa.c: Use the right flag to open the audio in
- non-blocking. (closes issue #12616) Reported by:
- nicklewisdigiumuser
-
-2008-05-13 18:36 +0000 [r115884] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: If the socket dies (read returns 0=EOF), return
- immediately. (Closes issue #12637)
-
-2008-05-12 17:51 +0000 [r115735] Mark Michelson <mmichelson@digium.com>
-
- * main/utils.c: If a thread holds no locks, do not print any
- information on the thread when issuing a core show locks command.
- This will help to de-clutter output somewhat. Russell said it
- would be fine to place this improvement in the 1.4 branch, so
- that's why it's going here too.
-
-2008-05-09 16:34 +0000 [r115579] Joshua Colp <jcolp@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- Improve res_ninit and res_ndestroy autoconf logic on the Darwin
- platform.
-
-2008-05-08 19:19 +0000 [r115545-115568] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Remove debug output.
-
- * /, channels/chan_iax2.c: Merged revisions 115564 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08
- May 2008) | 25 lines Fix a race condition that bbryant just found
- while doing some IAX2 testing. He was running Asterisk trunk
- running IAX2 calls through a few Asterisk boxes, however, the
- audio was extremely choppy. We looked at a packet trace and saw a
- storm of INVAL and VNAK frames being sent from one box to
- another. It turned out that what had happened was that one box
- tried to send a CONTROL frame before the 3 way handshake had
- completed. So, that frame did not include the destination call
- number, because it didn't have it yet. Part of our recent work
- for security issues included an additional check to ensure that
- frames that are supposed to include the destination call number
- have the correct one. This caused the frame to be rejected with
- an INVAL. The frame would get retransmitted for forever, rejected
- every time ... This race condition exists in all versions that
- got the security changes, in theory. However, it is really only
- likely that this would cause a problem in Asterisk trunk. There
- was a control frame being sent (SRCUPDATE) at the _very_
- beginning of the call, which does not exist in 1.2 or 1.4.
- However, I am fixing all versions that could potentially be
- affected by the introduced race condition. These changes are what
- bbryant and I came up with to fix the issue. Instead of simply
- dropping control frames that get sent before the handshake is
- complete, the code attempts to wait a little while, since in most
- cases, the handshake will complete very quickly. If it doesn't
- complete after yielding for a little while, then the frame gets
- dropped. ........
-
- * channels/chan_sip.c: Don't give up on attempting an outbound
- registration if we receive a 408 Timeout. (closes issue #12323)
-
- * contrib/scripts/postgres_cdr.sql (removed): remove
- postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as
- well (closes issue #9676)
-
- * contrib/init.d/rc.debian.asterisk: Don't exit the script if
- Asterisk is not running. (closes issue #12611)
-
- * main/pbx.c: Don't use a channel before checking for channel
- allocation failure. (closes issue #12609) Reported by: edantie
-
- * contrib/init.d/rc.debian.asterisk: Use the same method for
- executing Asterisk as the rest of the script. (closes issue
- #12611) Reported by: b_plessis
-
-2008-05-07 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.20-rc2 released.
-
-2008-05-07 18:17 +0000 [r115512-115517] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Track peer references when stored in the
- sip_pvt struct as the peer related to a qualify ping or a
- subscription. This fixes some realtime related crashes. (closes
- issue #12588) (closes issue #12555)
-
-2008-05-06 19:55 +0000 [r115418-115422] Jason Parker <jparker@digium.com>
-
- * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
- 7 lines read requires an argument on some non-bash shells (closes
- issue #12593) Reported by: bkruse Patches:
- getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
- ........
-
- * res/res_musiconhold.c: Switch to using ast_random() rather than
- just rand(). This does not fix the bug reported, but I believe it
- is correct. (from issue #12446) Patches: bug_12446.diff uploaded
- by snuffy (license 35)
-
-2008-05-06 19:31 +0000 [r115415] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: Don't print the terminating NUL. (Closes issue
- #12589)
-
-2008-05-06 13:54 +0000 [r115341] Joshua Colp <jcolp@digium.com>
-
- * configure, configure.ac: Add in missing argument.
-
-2008-05-05 22:50 +0000 [r115333] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, main/logger.c: Separate verbose output from CLI
- output, by using a preamble. (closes issue #12402) Reported by:
- Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt
- uploaded by Corydon76 (license 14)
- 20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by
- Corydon76 (license 14)
-
-2008-05-05 22:10 +0000 [r115327] Joshua Colp <jcolp@digium.com>
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
- configure.ac: Make sure that either the main speex library
- contains preprocess functions or that speexdsp does. If both fail
- then speex stuff can not be built.
-
-2008-05-05 21:41 +0000 [r115320] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Don't consider a caller "handled" until the
- caller is bridged with a queue member. There was too much of an
- opportunity for the member to hang up (either during a delay,
- announcement, or overly long agi) between the time that he
- answered the phone and the time when he actually was bridged with
- the caller. The consequence of this was that if the member hung
- up in that interval, then proper abandonment details would not be
- noted in the queue log if the caller were to hang up at any point
- after the member hangup. (closes issue #12561) Reported by:
- ablackthorn
-
-2008-05-05 20:17 +0000 [r115308-115312] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile: Reverse order, such that user configs override default
- selections
-
- * include/asterisk/res_odbc.h: Err, the documentation on the return
- value of ast_odbc_backslash_is_escape is exactly backwards.
-
-2008-05-05 19:49 +0000 [r115297-115304] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Avoid putting opaque="" in Digest
- authentication. This patch came from switchvox. It fixes
- authentication with Primus in Canada, and has been in use for a
- very long time without causing problems with any other providers.
- (closes issue AST-36)
-
-2008-05-05 03:22 +0000 [r115285] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
- contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.slackware.asterisk: When starting Asterisk, bug
- out if Asterisk is already running. (closes issue #12525)
- Reported by: explidous Patches: 20080428__bug12525.diff.txt
- uploaded by Corydon76 (license 14) Tested by: mvanbaak
-
-2008-05-04 02:09 +0000 [r115276-115282] Joshua Colp <jcolp@digium.com>
-
- * configure, acinclude.m4: Expand the test function for GCC
- attributes so that more complex attributes are properly
- recognized.
-
- * include/asterisk/compiler.h: For my next trick I will make these
- work with what our autoconf header file gives us.
-
- * configure, acinclude.m4: Treat warnings as errors when checking
- if a GCC attribute exists. We have to do this as GCC will just
- ignore the attribute and pop up a warning, it won't actually fail
- to compile.
-
-2008-05-02 20:25 +0000 [r115257] Brett Bryant <bbryant@digium.com>
-
- * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
- configure.ac, CHANGES: Add new "pri show version" command to show
- the libpri version for support reasons.
-
-2008-05-02 14:28 +0000 [r115196] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/sched.h: Clarify a comment that was, well, just
- wrong. It turns out that ignoring the way that macros expand.
- Instead, I have clarified in the comment why the macro will work
- even if the scheduler id for the task to be deleted changes
- during the execution of the macro.
-
-2008-05-01 23:20 +0000 [r115017-115102] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h: Change the comment of deprecated to
- an actual compiler deprecation
-
- * main/utils.c: '#' is another reserved character for URIs that
- also needs to be escaped. (closes issue #10543) Reported by:
- blitzrage Patches: 20080418__bug10543.diff.txt uploaded by
- Corydon76 (license 14)
-
-2008-05-01 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.20-rc1 released.
-
-2008-04-30 16:30 +0000 [r114891] Russell Bryant <russell@digium.com>
-
- * include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
- Merge changes from team/russell/iax2_find_callno and
- iax2_find_callno_1.4 These changes address a critical performance
- issue introduced in the latest release. The fix for the latest
- security issue included a change that made Asterisk randomly
- choose call numbers to make them more difficult to guess by
- attackers. However, due to some inefficient (this is by far, an
- understatement) code, when Asterisk chose high call numbers,
- chan_iax2 became unusable after just a small number of calls. On
- a small embedded platform, it would not be able to handle a
- single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
- more than about 16 IAX2 channels. Ouch. These changes address
- some performance issues of the find_callno() function that have
- bothered me for a very long time. On every incoming media frame,
- it iterated through every possible call number trying to find a
- matching active call. This involved a mutex lock and unlock for
- each call number checked. So, if the random call number chosen
- was 20000, then every media frame would cause 20000 locks and
- unlocks. Previously, this problem was not as obvious since
- Asterisk always chose the lowest call number it could. A second
- container for IAX2 pvt structs has been added. It is an astobj2
- hash table. When we know the remote side's call number, the pvt
- goes into the hash table with a hash value of the remote side's
- call number. Then, lookups for incoming media frames are a very
- fast hash lookup instead of an absolutely insane array traversal.
- In a quick test, I was able to get more than 3600% more IAX2
- channels on my machine with these changes.
-
-2008-04-30 16:23 +0000 [r114890] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't crash on bad SIP replys. Fix created
- in Huntsville together with Mark M (putnopvut) (closes issue
- #12363) Reported by: jvandal Tested by: putnopvut, oej
-
-2008-04-30 14:46 +0000 [r114875-114880] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro
- for indexing through the iaxs/iaxsl arrays so that the size of
- the arrays can be adjusted in one place, and change the size of
- the arrays from 32768 calls to 2048 calls when LOW_MEMORY is
- defined
-
- * Makefile.rules: pay attention to *all* header files for
- dependency tracking, not just the local ones (inspired by r578 of
- asterisk-addons by tilghman)
-
-2008-04-29 19:40 +0000 [r114848] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel
- variables instead of the channel's macrocontext and macroexten
- fields. This is needed because if macros are daisy-chained, the
- incorrect context and extension are placed on the new channel. I
- also added locking to the channel prior to accessing these
- variables as noted in trunk's janitor project file. (closes issue
- #12549) Reported by: darren1713 Patches:
- app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
- (with modifications from me) Tested by: putnopvut
-
-2008-04-29 17:08 +0000 [r114829] Jason Parker <jparker@digium.com>
-
- * res/res_config_pgsql.c: Change warning message to debug, since
- there are cases where 0 results is perfectly fine.
-
-2008-04-29 12:53 +0000 [r114823] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
- 2008) | 2 lines stop script from appending source code if run
- multiple times ........
-
-2008-04-28 04:47 +0000 [r114708] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, channels/chan_gtalk.c: When modules are
- embedded, they take on a different name, without the ".so"
- extension. Specifically check for this name, when we're checking
- if a module is loaded. (Closes issue #12534)
-
-2008-04-27 01:26 +0000 [r114695] Sean Bright <sean.bright@gmail.com>
-
- * configure, configure.ac: When we don't explicitly pass a path to
- the --with-tds configure option, we may end up finding tds.h in
- /usr/local/include instead of /usr/include. If this happens, the
- grep that looks for the version (from tdsver.h) will fail and
- we'll have some problems during the build.
-
-2008-04-26 13:15 +0000 [r114689] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/vmail.cgi: Clicking forward without selecting a
- message leaves an errant .lock file. (closes issue #12528)
- Reported by: pukepail Patches: patch.diff uploaded by pukepail
- (license 431)
-
-2008-04-25 21:54 +0000 [r114673] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Use consistent logic for checking to see if
- a call number has been chosen yet. Also, remove some redundant
- logic I recently added in a fix.
-
-2008-04-25 19:32 +0000 [r114662] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Move the unlock of the spyee channel to
- outside the start_spying() function so that the channel is not
- unlocked twice when using whisper mode.
-
-2008-04-25 15:53 +0000 [r114649] Tilghman Lesher <tlesher@digium.com>
-
- * configs/zapata.conf.sample, configs/iax.conf.sample,
- configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference
- documentation files that actually exist. (closes issue #12516)
- Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded
- by linuxmaniac (license 472)
-
-2008-04-24 21:35 +0000 [r114624-114632] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Re-invite RTP during a masquerade so that,
- for instance, an AMI redirect of two channels which are natively
- bridged will preserve audio on both channels. This prevents a
- problem with Asterisk not re-inviting due to one of the channels
- having being a zombie. (closes issue #12513) Reported by:
- mneuhauser Patches:
- asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
- mneuhauser (license 425)
-
- * apps/app_queue.c: Output of channel variables when
- eventwhencalled=vars was set was being truncated two characters.
- This patch corrects the problem. (closes issue #12493) Reported
- by: davidw
-
- * channels/chan_local.c: Resolve a deadlock in chan_local by
- releasing the channel lock temporarily. (closes issue #11712)
- Reported by: callguy Patches: 11712.patch uploaded by putnopvut
- (license 60) Tested by: acunningham
-
-2008-04-24 19:53 +0000 [r114621] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c: Ensure that when we set the accountcode,
- it actually shows up in the CDR. (Fix for AMI Originate) (Closes
- issue #12007)
-
-2008-04-24 15:55 +0000 [r114608] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a silly mistake in a change I made
- yesterday that caused chan_iax2 to blow up very quickly. (issue
- #12515)
-
-2008-04-24 14:55 +0000 [r114603] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Only have one max-forwards header in
- outbound REFERs. Discovered in the Asterisk SIP Masterclass in
- Orlando. Thanks Joe!
-
-2008-04-23 22:18 +0000 [r114597-114600] Russell Bryant <russell@digium.com>
-
- * main/http.c: Improve some broken cookie parsing code. Previously,
- manager login over HTTP would only work if the mansession_id
- cookie was first. Now, the code builds a list of all of the
- cookies in the Cookie header. This fixes a problem observed by
- users of the Asterisk GUI. (closes AST-20)
-
- * apps/app_chanspy.c, main/http.c: Fix an issue that caused getting
- the correct next channel to not always work. Also, remove setting
- the amount of time to wait for a digit from 5 seconds back down
- to 1/10 of a second. I believe this was so the beep didn't get
- played over and over really fast, but a while back I put in
- another fix for that issue. (closes issue #12498) Reported by:
- jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by
- jsmith (license 15)
-
-2008-04-23 18:28 +0000 [r114594] Jason Parker <jparker@digium.com>
-
- * res/res_musiconhold.c: Fix reload/unload for res_musiconhold
- module. (closes issue #11575) Reported by: sunder Patches:
- M11575_14_rev3.diff uploaded by junky (license 177)
- bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)
-
-2008-04-23 17:55 +0000 [r114587-114591] Russell Bryant <russell@digium.com>
-
- * main/manager.c, include/asterisk/manager.h: Store the manager
- session ID explicitly as 4 byte ID instead of a ulong. The
- mansession_id cookie is coded to be limited to 8 characters of
- hex, and this could break logins from 64-bit machines in some
- cases. (inspired by AST-20)
-
- * channels/chan_iax2.c: Fix find_callno_locked() to actually return
- the callno locked in some more cases.
-
-2008-04-23 16:51 +0000 [r114584] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Add 502 support for both directions, not
- only one... (see r114571)
-
-2008-04-23 14:54 +0000 [r114579] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: Instead of stopping dialplan execution when SayNumber
- attempts to say a large number that it can not print out a
- message informing the user and continue on. (closes issue #12502)
- Reported by: bcnit
-
-2008-04-22 23:51 +0000 [r114571] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Treat a 502 just like a 503, when it comes
- to processing a response code
-
-2008-04-22 22:15 +0000 [r114522-114558] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: When we receive a full frame that is
- supposed to contain our call number, ensure that it has the
- correct one. (closes issue #10078) (AST-2008-006)
-
- * main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I
- thought I was going to be able to leave 1.4 alone, but that was
- not the case. I ran into some problems with G.722 in 1.4, so I
- have merged in all of the fixes in this area that I have made in
- trunk/1.6.0, and things are happy again.
-
- * res/res_musiconhold.c: Trivial change to read the number of
- samples from a frame before calling ast_write()
-
- * res/res_features.c: After a parked call times out, allow the call
- back to the parker to time out. (closes issue #10890)
-
- * channels/chan_iax2.c: If the dial string passed to the call
- channel callback does not indicate an extension, then consider
- the extension on the channel before falling back to the default.
- (closes issue #12479) Reported by: darren1713 Patches:
- exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
- 116)
-
- * channels/chan_sip.c, include/asterisk/sched.h: Merge changes from
- team/russell/issue_9520 These changes make sure that the
- reference count for sip_peer objects properly reflects the fact
- that the peer is sitting in the scheduler for a scheduled
- callback for qualifying peers or for expiring registrations.
- Without this, it was possible for these callbacks to happen at
- the same time that the peer was being destroyed. This was
- especially likely to happen with realtime peers, and for people
- making use of the realtime prune CLI command. (closes issue
- #9520) Reported by: kryptolus Committed patch by me
-
-2008-04-21 14:39 +0000 [r114322] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Only drop audio if we receive it without a
- progress indication. We allow other frames through such as DTMF
- because they may be needed to complete the call. (closes issue
- #12440) Reported by: aragon
-
-2008-04-19 13:57 +0000 [r114297-114299] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_playback.c: Ensure that help text terminates with a
- newline
-
- * res/res_musiconhold.c: MOH usage information needs a terminating
- newline, or else "asterisk -rx 'help moh reload'" will hang.
- Reported via -dev list, fixed by me.
-
-2008-04-18 21:48 +0000 [r114275-114284] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Don't destroy a manager session if poll() returns
- an error of EAGAIN.
-
- * Makefile: ensure directories are created before we try to install
- stuff into them
-
- * Makefile: SUBDIRS_INSTALL is already listed as a subtarget for
- bininstall
-
-2008-04-18 17:44 +0000 [r114257] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c, main/callerid.c: Clearing up error messages
- so they make a bit more sense. Also removing a redundant error
- message. Issue AST-15
-
-2008-04-18 15:24 +0000 [r114248] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL)
- (closes issue #12476) Reported by: davidw Patch by me
-
-2008-04-18 13:33 +0000 [r114245] Sean Bright <sean.bright@gmail.com>
-
- * channels/chan_sip.c: Only complete the SIP channel name once for
- 'sip show channel <channel>'
-
-2008-04-18 06:49 +0000 [r114242] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_setcallerid.c: For consistency sake, ensure that the
- values that ${CALLINGPRES} returns are valid as an input to
- SetCallingPres. (Closes issue #12472)
-
-2008-04-17 22:15 +0000 [r114230] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c: Remove redundant safety net. The check for
- the autoservice channel list state accomplishes the same goal in
- a better way. (issue #12470) Reported By: atis
-
-2008-04-17 21:03 +0000 [r114207-114226] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Declaration of the peer channel in this scope
- was making it so the peer variable defined in the outer scope was
- never set properly, therefore making iterating through the
- channel list always restart from the beginning. This bug would
- have affected anyone who called chanspy without specifying a
- first argument. (closes issue #12461) Reported by: stever28
-
- * main/frame.c, include/asterisk/dsp.h: Add prototype for
- ast_dsp_frame_freed. I'm not sure how this was compiling
- before...
-
- * main/dsp.c, main/frame.c, include/asterisk/frame.h: It was
- possible for a reference to a frame which was part of a freed DSP
- to still be referenced, leading to memory corruption and eventual
- crashes. This code change ensures that the dsp is freed when we
- are finished with the frame. This change is very similar to a
- change Russell made with translators back a month or so ago.
- (closes issue #11999) Reported by: destiny6628 Patches:
- 11999.patch uploaded by putnopvut (license 60) Tested by:
- destiny6628, victoryure
-
-2008-04-17 16:23 +0000 [r114204] Russell Bryant <russell@digium.com>
-
- * Makefile: Fix the bininstall target to install from subdirs, as
- well. (closes issue AST-8, patch from bmd at switchvox)
-
-2008-04-17 13:42 +0000 [r114198] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Use keepalives effectively in order diagnose
- bug #12432.
-
-2008-04-17 12:56 +0000 [r114195] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Add special case for when the agi cannot be
- executed, to comply with the documentation that we return failure
- in that case. (closes issue #12462) Reported by: fmueller
- Patches: 20080416__bug12462.diff.txt uploaded by Corydon76
- (license 14) Tested by: fmueller
-
-2008-04-17 10:51 +0000 [r114191] Sean Bright <sean.bright@gmail.com>
-
- * apps/app_chanspy.c: Make sure we have enough room for the
- recording's filename.
-
-2008-04-16 20:46 +0000 [r114184] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by
- initializing the structure to all zeroes in case it contains
- fields that we don't write values into (which it does as of
- Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian
-
-2008-04-16 19:59 +0000 [r114180] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_vpb.cc: Backport revisions for latest vpb drivers
- to 1.4 (Closes issue #12457)
-
-2008-04-16 17:30 +0000 [r114173] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Fix "fallthrough" behavior here, so config
- options in a previously configured user don't override settings
- in general. (closes issue #12458) Reported by: tzafrir Patches:
- chanzap_users_sections.diff uploaded by tzafrir (license 46)
-
-2008-04-16 14:10 +0000 [r114167] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Include the proper headers for using mkdir on
- FreeBSD. (closes issue #12430) Reported by: ys Patches:
- app_meetme.c.diff uploaded by ys (license 281)
-
-2008-04-15 20:26 +0000 [r114148] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Handle subscribe queues in all situations...
- Thanks to festr_ on irc for telling me about this bug.
-
-2008-04-15 17:17 +0000 [r114120-114138] Jason Parker <jparker@digium.com>
-
- * contrib/scripts/autosupport: Update Digium autosupport script,
- for more useful information. (closes issue #12452) Reported by:
- angler Patches: autosupport.diff uploaded by angler (license 106)
-
- * apps/app_queue.c: Allow autofill to work in the general section
- of queues.conf. Additionally, don't try to (re)set options when
- they have empty values in realtime (all unset columns would have
- an empty value). (closes issue #12445) Reported by: atis Patches:
- 12445-autofill.diff uploaded by qwell (license 4)
-
- * channels/chan_h323.c: The call_token on the pvt can occasionally
- be NULL, causing a crash. If it is NULL, we can skip this
- channel, since it can't the one we're looking for. (closes issue
- #9299) Reported by: vazir
-
-2008-04-14 17:41 +0000 [r114106-114117] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: Increase the retry count when attempting to show
- channels. This apparently cleared an issue someone was seeing
- when attempting to show channels when the load was high. (closes
- issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
- by russell (license 2) Tested by: falves11
-
- * apps/app_dial.c, apps/app_queue.c: If the datastore has been
- moved to another channel due to a masquerade, then freeing the
- datastore here causes an eventual double free when the new
- channel hangs up. We should only free the datastore if we were
- able to successfully remove it from the channel we are
- referencing (i.e. the datastore was not moved). (closes issue
- #12359) Reported by: pguido
-
- * main/channel.c: Save a local copy of the generate callback prior
- to unlocking the channel in case the generate callback goes NULL
- on us after the channel is unlocked. Thanks to Russell for
- pointing this need out to me.
-
-2008-04-14 14:52 +0000 [r114100-114103] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: It is possible for the remote side to say
- they want T38 but not give any capabilities. (closes issue
- #12414) Reported by: MVF
-
- * main/rtp.c: Don't change the SSRC when a new source comes into
- play, this might happen quite often and depending on the remote
- side... they might not like this. (closes issue #12353) Reported
- by: dimas
-
-2008-04-11 22:32 +0000 [r114083] Terry Wilson <twilson@digium.com>
-
- * channels/chan_iax2.c: Several places in the code called
- find_callno() (which releases the lock on the pvt structure) and
- then immediately locked the call and did things with it.
- Unfortunately, the call can disappear between the find_callno and
- the lock, causing Bad Stuff(tm) to happen. Added
- find_callno_locked() function to return the callno withtout
- unlocking for instances that it is needed. (issue #12400)
- Reported by: ztel
-
-2008-04-11 21:35 +0000 [r114072] Jason Parker <jparker@digium.com>
-
- * main/pbx.c: It's possible that a channel can have an async goto
- on the successful execution of an application as well. Closes
- issue #12172.
-
-2008-04-11 15:44 +0000 [r114045-114063] Mark Michelson <mmichelson@digium.com>
-
- * res/res_features.c: Fix a race condition that may happen between
- a sip hangup and a "core show channel" command. This patch adds
- locking to prevent the resulting crash. (closes issue #12155)
- Reported by: tsearle Patches: show_channels_crash2.patch uploaded
- by tsearle (license 373) Tested by: tsearle
-
- * main/utils.c, include/asterisk/lock.h: Fix 1.4 build when
- LOW_MEMORY is enabled.
-
- * channels/chan_sip.c: Be sure that we're not about to set
- bridgepvt NULL prior to dereferencing it. (closes issue #11775)
- Reported by: fujin
-
-2008-04-10 17:26 +0000 [r114035] Jason Parker <jparker@digium.com>
-
- * main/file.c: Only try to prefix language if we are not using an
- absolute path (suffix it otherwise).
- en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
- issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
- uploaded by qwell (license 4) Tested by: kuj, qwell
-
-2008-04-10 15:58 +0000 [r114021-114032] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Forgot the 1.4 branch for russian language
- fix. (closes issue #12404) Reported by: IgorG Patches:
- voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)
-
- * apps/app_meetme.c: Create the directory where name recordings
- will go if it does not exist. (closes issue #12311) Reported by:
- rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4)
-
- * channels/chan_sip.c: Don't add custom URI options if they don't
- exist OR they are empty. (closes issue #12407) Reported by:
- homesick Patches: uri_options-1.4.diff uploaded by homesick
- (license 91)
-
-2008-04-09 20:54 +0000 [r113927] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: We need to set the persistant_route [sic]
- parameter for the sip_pvt during the initial INVITE, no matter if
- we're building the route set from an INVITE request or response.
- (closes issue #12391) Reported by: benjaminbohlmann Tested by:
- benjaminbohlmann
-
-2008-04-09 18:57 +0000 [r113874] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does
- not exist in cdr.conf, then an unload/load sequence is needed to
- correct the problem. Track whether the load succeeded with a
- variable, so we can fix this with a simple reload event, instead.
-
-2008-04-09 16:50 +0000 [r113784] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: If we receive an AUTHREQ from the remote
- server and we are unable to reply (for example they have a secret
- configured, but we do not) then queue a hangup frame on the
- Asterisk channel. This will cause the channel to hangup and a
- HANGUP to be sent via IAX2 to the remote side which is the proper
- thing to do in this scenario. (closes issue #12385) Reported by:
- viraptor
-
-2008-04-09 14:40 +0000 [r113681] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not
- a reinvite), then we should not send a BYE. (closes issue #12392)
- Reported by: fnordian Patches: chan_sip.patch uploaded by
- fnordian (license 110) with small modification from me
-
-2008-04-09 01:34 +0000 [r113596] Terry Wilson <twilson@digium.com>
-
- * channels/chan_iax2.c: Initialize fr->cacheable to make valgrind
- happy
-
-2008-04-08 19:07 +0000 [r113507] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_parkandannounce.c: Fix potential buffer overflow that
- could happen if more than 100 announce files were specified when
- calling ParkAndAnnounce. This overflow is not exploitable
- remotely and so there is no need for a security advisory. (closes
- issue #12386) Reported by: davidw
-
-2008-04-08 18:48 +0000 [r113402-113504] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Add a little more that is required for
- previously added devices.
-
- * channels/chan_skinny.c: Add support for several new(ish) devices
- - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver
- for providing me the required information.
-
- * main/asterisk.c: Work around some silliness caused by
- sys/capability.h - this should fix compile errors a number of
- users have been experiencing.
-
-2008-04-08 16:51 +0000 [r113348-113399] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/astgenkey.8: Add security note on astgenkey's
- manpage. (closes issue #12373) Reported by: lmamane Patches:
- 20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)
-
- * channels/chan_sip.c: Move check for still-bridged channels out a
- little further, to avoid possible deadlocks. (Closes issue
- #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt
- uploaded by Corydon76 (license 14) Tested by: callguy
-
-2008-04-08 15:03 +0000 [r113296] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/slinfactory.h, main/slinfactory.c,
- main/audiohook.c: If audio suddenly gets fed into one side of a
- channel after a lapse of frames flush the other factory so that
- old audio does not remain in the factory causing the sync code to
- not execute. (closes issue #12296) Reported by: jvandal
-
-2008-04-07 21:34 +0000 [r113240] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_sip.c: (closes issue #12362) [redo of 113012] This
- fixes a for loop (in realtime_peer) to check all the
- ast_variables the loop was intending to test rather than just the
- first one. The change exposed the problem of calling memcpy on a
- NULL pointer, in this case the passed in sockaddr_in struct which
- is now checked.
-
-2008-04-07 18:00 +0000 [r113118] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c, configs/skinny.conf.sample: Allow
- playback with noanswer (and add earlyrtp option). (closes issue
- #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn
- (license 30) Tested by: pj, qwell, DEA, wedhorn
-
-2008-04-07 17:51 +0000 [r113117] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_strings.c: Force ast_mktime() to check for DST, since
- strptime(3) does not. (Closes issue #12374)
-
-2008-04-07 16:08 +0000 [r113065] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: This fix prevents a deadlock that was experienced
- in chan_local. There was deadlock prevention in place in
- chan_local, but it would not work in a specific case because the
- channel was recursively locked. By unlocking the channel prior to
- calling the generator's generate callback in
- ast_read_generator_actions(), we prevent the recursive locking,
- and therefore the deadlock. (closes issue #12307) Reported by:
- callguy Patches: 12307.patch uploaded by putnopvut (license 60)
- Tested by: callguy
-
-2008-04-07 15:16 +0000 [r113012] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_sip.c: (closes issue #12362) (closes issue #12372)
- Reported by: vinsik Tested by: tecnoxarxa This one line change
- makes an if inside a for loop (in realtime_peer) check all the
- ast_variables the loop was intending to test rather than just the
- first one.
-
-2008-04-04 19:26 +0000 [r112766-112820] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Free newly allocated channel before
- returning
-
- * channels/chan_gtalk.c: Prevent call connections when codecs don't
- match. (closes issue #10604) Reported by: keepitcool Patches:
- branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
- by: phsultan
-
-2008-04-04 00:52 +0000 [r112709-112711] Joshua Colp <jcolp@digium.com>
-
- * main/Makefile: Pass in the path to Zaptel for systems that
- install Zaptel headers in a separate location.
-
- * main/asterisk.c: One thing at a time... let's get 1.4 building.
-
-2008-04-03 23:57 +0000 [r112689] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/asterisk.c: add a Zaptel timer check to verify the timer is
- responding when Zaptel support is compiled into Asterisk and
- Zaptel drivers are loaded. This will help people not waste their
- valuable time debugging side effects.
-
-2008-04-03 14:32 +0000 [r112393-112599] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c: Fix the testing of the "res" variable so
- that it is more logically correct and makes the correct warning
- and debug messages print. (closes issue #12361) Reported by:
- one47 Patches: chan_zap_deferred_digit.patch uploaded by one47
- (license 23)
-
- * main/manager.c: Fix a race condition in the manager. It is
- possible that a new manager event could be appended during a
- brief time when the manager is not waiting for input. If an event
- comes during this period, we need to set an indicator that there
- is an event pending so that the manager doesn't attempt to wait
- forever for an event that already happened. (closes issue #12354)
- Reported by: bamby Patches: manager_race_condition.diff uploaded
- by bamby (license 430) (comments added by me)
-
- * apps/app_queue.c: Ensure that there is no timeout if none is
- specified. (closes issue #12349) Reported by: johnlange
-
-2008-04-01 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.19 released.
-
-2008-03-28 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.19-rc4 released.
-
-2008-03-28 16:19 +0000 [r111658] Jason Parker <jparker@digium.com>
-
- * formats/format_wav_gsm.c: The file size of WAV49 does not need to
- be an even number. (closes issue #12128) Reported by: mdu113
- Patches: 12128-noevenlength.diff uploaded by qwell (license 4)
- Tested by: qwell, mdu113
-
-2008-03-28 14:35 +0000 [r111442-111605] Tilghman Lesher <tlesher@digium.com>
-
- * doc/valgrind.txt: Update debugging text, since Valgrind
- eliminated the --log-file-exactly option. (Closes issue #12320)
-
- * main/acl.c: For FreeBSD, at least, the ifa_addr element could be
- NULL. (closes issue #12300) Reported by: festr Patches:
- acl.c.patch uploaded by festr (license 443)
-
-2008-03-27 13:03 +0000 [r111341-111391] Steve Murphy <murf@digium.com>
-
- * apps/app_playback.c, main/pbx.c: These small documentation
- updates made in response to a query in asterisk-users, where a
- user was using Playback, but needed the features of Background,
- and had no idea that Background existed, or that it might provide
- the features he needed. I thought the best way to avert these
- kinds of queries was to provide "See Also" references in all
- three of "Background", "Playback", "WaitExten". Perhaps a project
- to do this with all related apps is in order.
-
- * pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue
- #12302) Reported by: pj Tested by: murf These changes will set a
- channel variable ~~EXTEN~~ just before generating code for a
- switch, with the value of ${EXTEN}. The exten is marked as having
- a switch, and ever after that, till the end of the exten, we
- substitute any ${EXTEN} with ${~~EXTEN~~} instead in application
- arguments; (and the ${EXTEN: also). The reason for this, is that
- because switches are coded using separate extensions to provide
- pattern matching, and jumping to/from these switch extensions
- messes up the ${EXTEN} value, which blows the minds of users.
-
-2008-03-27 00:25 +0000 [r111245-111280] Jason Parker <jparker@digium.com>
-
- * main/frame.c: Put this flag back so we don't change the API.
-
- * main/frame.c: Remove excessive smoother optimization that was
- causing audio glitches (small "pops") after (about 200ms later)
- an "incorrectly" sized frame was received. While it would be very
- nice to keep this as optimized as possible, it makes no sense for
- the smoother to be dropping random bits of audio like this. Isn't
- that the whole point of a smoother? Closes issue #12093.
-
-2008-03-26 19:55 +0000 [r111129] Joshua Colp <jcolp@digium.com>
-
- * contrib/scripts/autosupport: Update autosupport script. (closes
- issue #12310) Reported by: angler Patches: autosupport.diff
- uploaded by angler (license 106)
-
-2008-03-26 19:51 +0000 [r111126] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 111125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
- 2008) | 2 lines update UPGRADE notes to document usage of the
- script ........
-
-2008-03-26 19:37 +0000 [r111049-111121] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: This code change is made just for
- clarification. It does exactly the same thing as before. It just
- doesn't look as wrong.
-
- * apps/app_voicemail.c: Add a lock to the vm_state structure and
- use the lock around mail_open calls to prevent concurrent access
- of the same mailstream. This, along with trunk's ability to
- configure TCP timeouts for IMAP storage will help to prevent
- crashes and hangs when using voicemail with IMAP storage. (closes
- issue #10487) Reported by: ewilhelmsen
-
-2008-03-26 19:06 +0000 [r111024] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
- Merged revisions 111019 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
- 2008) | 2 lines add a script to make getting the iLBC source code
- simple for end users ........
-
-2008-03-26 19:04 +0000 [r111014-111020] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If we are requested to authenticate a
- reinvite make sure that it contains T38 SDP if need be. (closes
- issue #11995) Reported by: fall
-
- * channels/chan_iax2.c: Make sure that full video frames are sent
- whenever the 15 bit timestamp rolls over. (closes issue #11923)
- Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by
- mihai (license 94)
-
-2008-03-26 17:43 +0000 [r110880-110962] Kevin P. Fleming <kpfleming@digium.com>
-
- * UPGRADE.txt: add note that the user will need to enable
- codec_ilbc to get it to build
-
- * codecs/ilbc/StateConstructW.h (removed),
- codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h
- (removed), codecs/ilbc/getCBvec.c (removed),
- codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
- (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
- (removed), codecs/ilbc/getCBvec.h (removed),
- codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h
- (removed), codecs/ilbc/FrameClassify.c (removed),
- codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed),
- codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
- (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
- (removed), codecs/ilbc/anaFilter.c (removed),
- codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
- (removed), codecs/ilbc/doCPLC.h (removed),
- codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
- codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
- (removed), codecs/ilbc/createCB.h (removed), CHANGES,
- codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h
- (removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed),
- codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
- codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
- (removed), codecs/ilbc/iCBSearch.h (removed),
- codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
- codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
- (removed), codecs/ilbc/hpOutput.h (removed),
- codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
- codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
- (removed), codecs/ilbc/iCBConstruct.c (removed),
- codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
- (removed), codecs/ilbc/syntFilter.h (removed),
- codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
- (removed): Merged revisions 110869 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
- 2008) | 2 lines due to licensing restrictions, we cannot
- distribute the source code for iLBC encoding and decoding... so
- remove it, and add instructions on how the user can obtain it
- themselves ........
-
-2008-03-25 22:51 +0000 [r110779] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_custom.c: Make file access in cdr_custom similar to
- cdr_csv. Fixes issue #12268. Patch borrowed from r82344
-
-2008-03-25 20:03 +0000 [r110727] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_sip.c: This one line change makes an if inside a
- for loop (in realtime_peer) check all the ast_variables the loop
- was intending to test rather than just the first one.
-
-2008-03-25 15:40 +0000 [r110635] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: When reverting a commit, I accidentally left
- in this bit which was an experiment to see what would happen. It
- passed the compile test, and I didn't notice I had left this
- change in too. So this is a revert of a revert...sort of.
-
-2008-03-25 14:37 +0000 [r110628] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/options.h, main/asterisk.c, Makefile,
- main/app.c: Add an option (transmit_silence) which transmits
- silence during both Record() and DTMF generation. The reason this
- is an option is that in order to transmit silence we have to
- setup a translation path. This may not be needed/wanted in all
- cases. (closes issue #10058) Reported by: tracinet
-
-2008-03-24 19:17 +0000 [r110618] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: This is a revert for revision 108288. The
- reason is that that revision was not for an actual bug fix per
- se, and so it really should not have been in 1.4 in the first
- place. Plus, people who compile with DO_CRASH are more likely to
- encounter a crash due to this change. While I think the usage of
- DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
- beyond the scope of 1.4 and should be done instead in a developer
- branch based on trunk so that all scheduler functions are fixed
- at once. I also am reverting the change to trunk and 1.6 since
- they also suffer from the DO_CRASH potential. (closes issue
- #12272) Reported by: qq12345
-
-2008-03-24 17:34 +0000 [r110614] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Turn a NOTICE into a DEBUG message.
-
-2008-03-21 14:32 +0000 [r110474] Jason Parker <jparker@digium.com>
-
- * codecs/gsm/Makefile: Don't attempt to do optimizations of gsm on
- mips platforms either. (closes issue #12270) Reported by:
- zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt
- (license 33)
-
-2008-03-20 23:13 +0000 [r110163-110395] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c: Shorten the ast_waitfor() timeout from 500 ms
- to 50 ms in the autoservice thread. This really should not make a
- difference except in very rare cases. That case would be that all
- of the channels in autoservice are not generating any frames. In
- that case, this change reduces the potential amount of time that
- a thread waits in ast_autoservice_stop() for the autoservice
- thread to wrap back around to the beginning of its loop. (closes
- issue #12266, reported by dimas)
-
- * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
- 110335 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
- | 6 lines Fix some very broken code that was introduced in 1.2.26
- as a part of the security fix. The dnsmgr is not appropriate
- here. The dnsmgr takes a pointer to an address structure that a
- background thread continuously updates. However, in these cases,
- a stack variable was passed. That means that the dnsmgr thread
- would be continuously writing to bogus memory. ........
-
- * apps/app_meetme.c: Fix a bug where when calls on the trunk side
- hang up while on hold, the state is not properly reflected.
- (closes issue #11990, reported by anakaoka, patched by me)
-
-2008-03-19 20:33 +0000 [r110083] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Add a missing unlock in the case that memory
- allocation fails in app_chanspy. Thanks to Russell for confirming
- that this was an issue.
-
-2008-03-19 19:11 +0000 [r110019-110035] Joshua Colp <jcolp@digium.com>
-
- * res/res_musiconhold.c: Add sanity checking for position resuming.
- We *have* to make sure that the position does not exceed the
- total number of files present, and we have to make sure that the
- position's filename is the same as previous. These values can
- change if a music class is reloaded and give unpredictable
- behavior. (closes issue #11663) Reported by: junky
-
- * main/rtp.c: Make sure that the mark bit does not incorrectly
- cause video frame timestamps to be calculated as if they are
- audio frames. (closes issue #11429) Reported by: sperreault
- Patches: 11429-frametype.diff uploaded by qwell (license 4)
-
-2008-03-19 17:12 +0000 [r109973] Jason Parker <jparker@digium.com>
-
- * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
- (added): People report bugs about Asterisk crashing with DO_CRASH
- enabled was getting a little silly... Now we only show certain
- cflags when you run configure with --enable-dev-mode
- (corresponding menuselect change to follow)
-
-2008-03-19 15:41 +0000 [r109908] Steve Murphy <murf@digium.com>
-
- * main/config.c: (closes issue #11442) Reported by: tzafrir
- Patches: 11442.patch uploaded by murf (license 17) Tested by:
- murf I didn't give tzafrir very much time to test this, but if he
- does still have remaining issues, he is welcome to re-open this
- bug, and we'll do what is called for. I reproduced the problem,
- and tested the fix, so I hope I am not jumping by just going
- ahead and committing the fix. The problem was with what file_save
- does with templates; firstly, it tended to print out multiple
- options: [my_category](!)(templateref) instead of
- [my_category](!,templateref) which is fixed by this patch.
- Nextly, the code to suppress output of duplicate declarations
- that would occur because the reader copies inherited declarations
- down the hierarchy, was not working. Thus: [master-template](!)
- mastervar = bar [template](!,master-template) tvar = value
- [cat](template) catvar = val would be rewritten as: ;! ;!
- Automatically generated configuration file ;! Filename:
- experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
- Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
- [master-template](!) mastervar = bar
- [template](!,master-template) mastervar = bar tvar = value
- [cat](template) mastervar = bar tvar = value catvar = val This
- has been fixed. Since the config reader 'explodes' inherited vars
- into the category, users may, in certain circumstances, see
- output different from what they originally entered, but it should
- be both correct and equivalent.
-
-2008-03-19 04:06 +0000 [r109763-109838] Russell Bryant <russell@digium.com>
-
- * main/utils.c: Tweak spacing in a recent change because I'm very
- picky.
-
- * apps/app_chanspy.c: Fix one place where the chanspy datastore
- isn't removed from a channel. (issue #12243, reported by atis,
- patch by me)
-
-2008-03-18 20:52 +0000 [r109713] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This patch makes it so that all queue member
- status changes are handled through device state code. This
- removes several problems people were seeing where their queue
- members would get into an "unknown" state. Huge props go to atis
- on this one since he was the one who found the code section that
- was causing the problem and proposed the solution. I just wrote
- what he suggested :) (closes issue #12127) Reported by: atis
- Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested
- by: atis, jvandal
-
-2008-03-18 19:23 +0000 [r109648] Jason Parker <jparker@digium.com>
-
- * codecs/log2comp.h: Allow codecs that use log2comp (g726) to
- compile correctly on x86 with gcc4 optimizations. (closes issue
- #12253) Reported by: fossil Patches: log2comp.patch uploaded by
- fossil (license 140)
-
-2008-03-18 17:58 +0000 [r109575] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Make sure an agent doesn't try to send
- dtmf to a NULL channel closes issue #12242 Reported by Yourname
-
-2008-03-18 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.19-rc3 released.
-
-2008-03-18 16:25 +0000 [r109482] Terry Wilson <twilson@digium.com>
-
- * include/asterisk/astobj.h: Fix character string being treated ad
- format string
-
-2008-03-18 15:10 +0000 [r109393] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) |
- 3 lines Do not return with a successful authentication if the
- From header ends up empty. (AST-2008-003) ........
-
-2008-03-18 14:58 +0000 [r109386] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, channels/chan_sip.c: Put a maximum limit on the
- number of payloads accepted, and also make sure a given payload
- does not exceed our maximum value. (AST-2008-002)
-
-2008-03-18 06:37 +0000 [r109309] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ael-ntest23 (added),
- pbx/ael/ael-test/ael-ntest23/t1/a.ael (added),
- pbx/ael/ael-test/ael-ntest23/t1/b.ael (added),
- pbx/ael/ael-test/ael-ntest23/t1/c.ael (added),
- pbx/ael/ael-test/ael-ntest23/t2/d.ael (added),
- pbx/ael/ael-test/ael-ntest23/t2/e.ael (added),
- pbx/ael/ael-test/ael-ntest23/t2/f.ael (added),
- pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael_lex.c,
- pbx/ael/ael-test/ael-ntest23/t3/g.ael (added),
- pbx/ael/ael-test/ael-ntest23/t3/h.ael (added),
- pbx/ael/ael-test/ael-ntest23/t3/i.ael (added), pbx/ael/ael.flex,
- pbx/ael/ael-test/ael-ntest23/t3/j.ael (added),
- pbx/ael/ael-test/ael-ntest23/qq.ael (added),
- pbx/ael/ael-test/ael-ntest23/t1 (added),
- pbx/ael/ael-test/ael-ntest23/t2 (added),
- pbx/ael/ael-test/ael-ntest23/t3 (added),
- pbx/ael/ael-test/ael-ntest23/extensions.ael (added): (closes
- issue #11903) Reported by: atis Many thanks to atis for spotting
- this problem and reporting it. The fix was to straighten out how
- items are placed on and removed from the file stack. Regressions
- as well as the provided test case helped to straighten out all
- code paths. valgrind was used to make sure all memory allocated
- was freed. Sorry for not solving this earlier. I got distracted.
- Added the ntest23 regression test, which is mainly a copy of
- ntest22, but with a few juicy errors thrown in, to replicate the
- kind of error that atis spotted.
-
-2008-03-17 22:05 +0000 [r109226] Mark Michelson <mmichelson@digium.com>
-
- * main/utils.c: Fix a logic flaw in the code that stores lock info
- which is displayed via the "core show locks" command. The idea
- behind this section of code was to remove the previous lock from
- the list if it was a trylock that had failed. Unfortunately,
- instead of checking the status of the previous lock, we were
- referencing the index immediately following the previous lock in
- the lock_info->locks array. The result of this problem, under the
- right circumstances, was that the lock which we currently in the
- process of attempting to acquire could "overwrite" the previous
- lock which was acquired. While this does not in any way affect
- typical operation, it *could* lead to misleading "core show
- locks" output.
-
-2008-03-17 17:55 +0000 [r109171] Michiel van Baak <michiel@vanbaak.info>
-
- * channels/chan_skinny.c: Update the directory of placed calls on
- skinny phones when dialing a channel that does not provide
- progress (analog ZAP lines) The phone does handle the double
- update on calls to channels that do provide progress and wont
- insert duplicate items (closes issue #12239) Reported by: DEA
- Patches: chan_skinny-call-log.txt uploaded by DEA (license 3)
-
-2008-03-17 16:24 +0000 [r109107] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: 200 OKs in response to a reinvite need to be
- sent reliably. If the remote side does not receive one the dialog
- will be torn down. (closes issue #12208) Reported by: atrash
-
-2008-03-17 15:15 +0000 [r109057] Jason Parker <jparker@digium.com>
-
- * main/file.c: Backport revision 106439 from trunk. I didn't
- realize this was broken in 1.4 as well. Closes issue #12222.
-
-2008-03-17 14:18 +0000 [r109012] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Make sure that we release the lock on the
- spyee channel if the spyee or spy has hung up (closes issue
- #12232) Reported by: atis
-
-2008-03-16 21:47 +0000 [r108961] Michiel van Baak <michiel@vanbaak.info>
-
- * main/dial.c: add missing break to case AST_CONTROL_SRCUPDATE
- (closes issue #12228) Reported by: andrew Patches: SRC.patch
- uploaded by andrew (license 240)
-
-2008-03-14 20:09 +0000 [r108792-108796] Russell Bryant <russell@digium.com>
-
- * channels/chan_oss.c: Fix a channel name issue. chan_oss registers
- the "Console" channel type, but it created channels with an "OSS"
- prefix. (closes issue #12194, reported by davidw, patched by me)
-
- * contrib/init.d/rc.suse.asterisk: Update the SuSE init script to
- start networking before asterisk, as well. (closes issue #12200,
- reported by and change suggested by reinerotto)
-
-2008-03-14 16:44 +0000 [r108737] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fix a race condition in the SIP packet
- scheduler which could cause a crash. chan_sip uses the scheduler
- API in order to schedule retransmission of reliable packets (such
- as INVITES). If a retransmission of a packet is occurring, then
- the packet is removed from the scheduler and retrans_pkt is
- called. Meanwhile, if a response is received from the packet as
- previously transmitted, then when we ACK the response, we will
- remove the packet from the scheduler and free the packet. The
- problem is that both the ACK function and retrans_pkt attempt to
- acquire the same lock at the beginning of the function call. This
- means that if the ACK function acquires the lock first, then it
- will free the packet which retrans_pkt is about to read from and
- write to. The result is a crash. The solution: 1. If the ACK
- function fails to remove the packet from the scheduler and the
- retransmit id of the packet is not -1 (meaning that we have not
- reached the maximum number of retransmissions) then release the
- lock and yield so that retrans_pkt may acquire the lock and
- operate. 2. Make absolutely certain that the ACK function does
- not recursively lock the lock in question. If it does, then
- releasing the lock will do no good, since retrans_pkt will still
- be unable to acquire the lock. (closes issue #12098) Reported by:
- wegbert (closes issue #12089) Reported by: PTorres Patches:
- 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested
- by: jvandal
-
-2008-03-14 14:29 +0000 [r108682] Jason Parker <jparker@digium.com>
-
- * res/res_musiconhold.c: Fix a potential segfault if chan (or
- chan->music_state) is NULL. Closes issue #12210, credit to
- edantie for pointing this out.
-
-2008-03-13 21:38 +0000 [r108469-108583] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, main/channel.c, include/asterisk/channel.h:
- Fix another issue that was causing crashes in chanspy. This
- introduces a new datastore callback, called chan_fixup(). The
- concept is exactly like the fixup callback that is used in the
- channel technology interface. This callback gets called when the
- owning channel changes due to a masquerade. Before this was
- introduced, if a masquerade happened on a channel being spyed on,
- the channel pointer in the datastore became invalid. (closes
- issue #12187) (reported by, and lots of testing from atis) (props
- to file for the help with ideas)
-
- * channels/chan_sip.c: Make a tweak that gets the LEDs on polycom
- phones to blink when an extension that has been subscribed to
- goes on hold. Otherwise, they just stay on like it does when an
- extension is in use. (closes issue #11263) Reported by: russell
- Patches: notify_hold.rev1.txt uploaded by russell (license 2)
- Tested by: russell
-
- * apps/app_followme.c: Fix a couple uses of sprintf. The second one
- could actually cause an overflow of a stack buffer. It's not a
- security issue though, it only depends on your configuration.
-
-2008-03-12 21:53 +0000 [r108227-108288] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Change AST_SCHED_DEL use to ast_sched_del
- for autocongestion in chan_sip. The scheduler callback will
- always return 0. This means that this id is never rescheduled, so
- it makes no sense to loop trying to delete the id from the
- scheduler queue. If we fail to remove the item from the queue
- once, it will fail every single time. (Yes I realize that in this
- case, the macro would exit early because the id is set to -1 in
- the callback, but it still makes no sense to use that macro in
- favor of calling ast_sched_del once and being done with it) This
- is the first of potentially several such fixes.
-
- * include/asterisk/sched.h: Added a large comment before the
- AST_SCHED_DEL macro to explain its purpose as well as when it is
- appropriate and when it is not appropriate to use it. I also
- removed the part of the debug message that mentions that this is
- probably a bug because there are some perfectly legitimate places
- where ast_sched_del may fail to delete an entry (e.g. when the
- scheduler callback manually reschedules with a new id instead of
- returning non-zero to tell the scheduler to reschedule with the
- same idea). I also raised the debug level of the debug message in
- AST_SCHED_DEL since it seems like it could come up quite
- frequently since the macro is probably being used in several
- places where it shouldn't be. Also removed the redundant line,
- file, and function information since that is provided by ast_log.
-
-2008-03-12 19:57 +0000 [r108135] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, main/channel.c: (closes issue #12187,
- reported by atis, fixed by me after some brainstorming on the
- issue with mmichelson) - Update copyright info on app_chanspy. -
- Fix a race condition that caused app_chanspy to crash. The issue
- was that the chanspy datastore magic that was used to ensure that
- spyee channels did not disappear out from under the code did not
- completely solve the problem. It was actually possible for
- chanspy to acquire a channel reference out of its datastore to a
- channel that was in the middle of being destroyed. That was
- because datastore destruction in ast_channel_free() was done near
- the end. So, this left the code in app_chanspy accessing a
- channel that was partially, or completely invalid because it was
- in the process of being free'd by another thread. The following
- sort of shows the code path where the race occurred:
- =============================================================================
- Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
- --------------------------------------||-------------------------------------
- ast_channel_free() || - remove channel from channel list || -
- lock/unlock the channel to ensure || that no references retrieved
- from || the channel list exist. ||
- --------------------------------------||-------------------------------------
- || channel_spy() - destroy some channel data || - Lock chanspy
- datastore || - Retrieve reference to channel || - lock channel ||
- - Unlock chanspy datastore
- --------------------------------------||-------------------------------------
- - destroy channel datastores || - call chanspy datastore d'tor ||
- which NULL's out the ds' || - Operate on the channel ...
- reference to the channel || || - free the channel || || || -
- unlock the channel
- --------------------------------------||-------------------------------------
- =============================================================================
-
-2008-03-12 19:16 +0000 [r108086] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: if we receive an INVITE with a
- Content-Length that is not a valid number, or is zero, then don't
- process the rest of the message body looking for an SDP closes
- issue #11475 Reported by: andrebarbosa
-
-2008-03-12 18:26 +0000 [r108083] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, include/asterisk/audiohook.h,
- main/audiohook.c: Add a trigger mode that triggers on both read
- and write. The actual function that returns the combined audio
- frame though will wait until both sides have fed in audio, or
- until one side stops (such as the case when you call Wait).
- (closes issue #11945) Reported by: xheliox
-
-2008-03-12 16:59 +0000 [r108031] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Destroy the channel lock after the channel
- datastores. (inspired by issue #12187)
-
-2008-03-12 01:52 +0000 [r107877] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/iax-friends.sql, contrib/scripts/sip-friends.sql:
- Document all of the possible realtime fields
-
-2008-03-11 23:37 +0000 [r107714-107826] Jason Parker <jparker@digium.com>
-
- * doc/voicemail_odbc_postgresql.txt: Update documentation for pgsql
- ODBC voicemail. (closes issue #12186) Reported by: jsmith
- Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license
- 15)
-
- * channels/chan_gtalk.c: Copy voicemail dependency logic for
- res_adsi to chan_gtalk (for jabber). (closes issue #12014)
- Reported by: junky
-
-2008-03-11 20:48 +0000 [r107713] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile.rules, channels/Makefile: get chan_vpb to build properly
- in dev mode
-
-2008-03-11 20:47 +0000 [r107712] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Add a newline on a log
-
-2008-03-11 19:20 +0000 [r107582-107646] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: Make sure the visible indication is on the
- right channel so when the masquerade happens the proper
- indication is enacted. (closes issue #11707) Reported by: iam
-
- * apps/app_meetme.c: Add an additional check for setting conference
- parameter when using the marked user options. It was possible for
- it to return to a no listen/no talk state if a masquerade
- happened. (closes issue #12136) Reported by: aragon
-
- * apps/app_exec.c: Fix a minor spelling error. (closes issue
- #12183) Reported by: darrylc
-
-2008-03-11 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.19-rc2 released.
-
-2008-03-11 15:18 +0000 [r107352-107472] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_rpt.c: backport a fix from trunk
-
- * channels/misdn/isdn_lib.c, codecs/Makefile,
- channels/chan_misdn.c: fix various other problems found by gcc
- 4.3
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- apps/app_sms.c: stop checking for mktime() in the configure
- script... we don't use it, and the test is buggy under gcc 4.3
-
- * configure, main/Makefile, configure.ac, makeopts.in: check for
- compiler support for -fno-strict-overflow before using it (tested
- with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179)
- Reported by: Netview
-
- * configure, configure.ac: fix small bug in IMAP toolkit testing
-
- * main/udptl.c, utils/Makefile, main/Makefile,
- main/editline/readline.c, pbx/Makefile: fix up various compiler
- warnings found with gcc-4.3: - the output of flex includes a
- static function called 'input' that is not used, so for the
- moment we'll stop having the compiler tell us about unused
- variables in the flex source files (a better fix would be to
- improve our flex post-processing to remove the unused function) -
- main/stdtime/localtime.c makes assumptions about signed integer
- overflow, and gcc-4.3's improved optimizer tries to take
- advantage of handling potential overflow conditions at compile
- time; for now, suppress these optimizations until we can fiure
- out if the code needs improvement - main/udptl.c has some
- references to uninitialized variables; in one case there was no
- bug, but in the other it was certainly possibly for unexpected
- behavior to occur - main/editline/readline.c had an unused
- variable
-
-2008-03-11 00:59 +0000 [r107290] Terry Wilson <twilson@digium.com>
-
- * channels/chan_sip.c: If we fail to alloc a channel, we should
- re-lock the pvt structure before returning.
-
-2008-03-10 21:32 +0000 [r107230] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: Use non-global storage for eswitch
-
-2008-03-10 20:27 +0000 [r107173] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Make sure to reenable echo can after a
- "failed" (canceled, etc) three-way call. (closes issue #11335)
- Reported by: rebuild
-
-2008-03-10 20:17 +0000 [r107099-107161] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Fix another bug specifically related to asynchronous
- call origination. Once the PBX is started on the channel using
- ast_pbx_start(), then the ownership of the channel has been
- passed on to another thread. We can no longer access it in this
- code. If the channel gets hung up very quickly, it is possible
- that we could access a channel that has been free'd. (inspired by
- BE-386)
-
- * main/pbx.c: Fix some bugs related to originating calls. If the
- code failed to start a PBX on the channel (such as if you set a
- call limit based on the system's load average), then there were
- cases where a channel that has already been free'd using
- ast_hangup() got accessed. This caused weird memory corruption
- and crashes to occur. (fixes issue BE-386) (much debugging credit
- goes to twilson, final patch written by me)
-
- * main/channel.c: Resolve a compiler warning.
-
- * main/channel.c: Fix a race condition where the generator can go
- away (closes issue #12175, reported by edantie, patched by me)
-
-2008-03-10 14:33 +0000 [r107016] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, main/cdr.c, include/asterisk/cdr.h: Move where
- unanswered CDRs are dropped to the CDR core, not everything uses
- app_dial. (closes issue #11516) Reported by: ys Patches:
- branch_1.4_cdr.diff uploaded by ys (license 281) Tested by:
- anest, jcapp, dartvader
-
-2008-03-08 15:59 +0000 [r106945] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: don't generate D-Channel "up" and "down"
- messages unless the channel state is actually changing; also,
- generate the "up" message when an implicit "up" occurs due to
- reception of a normal event when we thought the channel was
- "down"
-
-2008-03-07 22:51 +0000 [r106895] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Only start the SLA thread if SLA has actually
- been configured.
-
-2008-03-07 22:14 +0000 [r106842] Jason Parker <jparker@digium.com>
-
- * main/editline/Makefile.in: Fix hardcoded grep in editline, were
- GNU grep is required. (closes issue #12124) Reported by: dmartin
-
-2008-03-07 19:32 +0000 [r106788] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Ignore source update control frame. (closes issue
- #12168) Reported by: plack
-
-2008-03-07 17:16 +0000 [r106704] Russell Bryant <russell@digium.com>
-
- * include/asterisk/sched.h: Change a warning message to a debug
- message. This is happening quite frequently, and it is not worth
- spamming users with these messages unless we are pretty confident
- that it should never happen. As it stands today, it _will_ and
- _does_ happen and until that gets cleaned up a reasonable amount
- on the development side, let's not spam the logs of everyone
- else. (closes issue #12154)
-
-2008-03-07 16:22 +0000 [r106552-106635] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Warn the user when a temporary greeting
- exists (Closes issue #11409)
-
- * main/rtp.c: Properly initialize rtp->schedid (Closes issue
- #12154)
-
- * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c,
- apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c,
- funcs/func_enum.c, channels/chan_misdn.c, main/frame.c,
- main/manager.c: Safely use the strncat() function. (closes issue
- #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt
- uploaded by Corydon76 (license 14)
-
-2008-03-06 22:10 +0000 [r106437] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Quell an annoying message that is likely to print
- every single time that ast_pbx_outgoing_app is called. The reason
- is that __ast_request_and_dial allocates the cdr for the channel,
- so it should be expected that the channel will have a cdr on it.
- Thanks to joetester on IRC for pointing this out
-
-2008-03-06 04:40 +0000 [r106328] Tilghman Lesher <tlesher@digium.com>
-
- * sounds/Makefile: Upgrade to the next release of sounds
-
-2008-03-05 22:37 +0000 [r106237] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a potential deadlock and a few
- different potential crashes. (closes issue #12145, reported by
- thiagarcia, patched by me)
-
-2008-03-05 22:32 +0000 [r106235] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_oss.c, main/rtp.c, channels/chan_mgcp.c,
- apps/app_dial.c, main/channel.c, channels/chan_phone.c,
- main/dial.c, channels/chan_zap.c, channels/chan_sip.c,
- channels/chan_skinny.c, channels/chan_h323.c, main/file.c,
- channels/chan_alsa.c, apps/app_followme.c,
- include/asterisk/frame.h: Add a control frame to indicate the
- source of media has changed. Depending on the underlying
- technology it may need to change some things. (closes issue
- #12148) Reported by: jcomellas
-
-2008-03-05 21:12 +0000 [r106178] Michiel van Baak <michiel@vanbaak.info>
-
- * doc/realtime.txt: document var_metric so no bugreports will come
- in when it's actually a configuration issue. (issue #12151)
- Reported and patched by: caio1982 1.4 patch by me
-
-2008-03-05 15:32 +0000 [r106038] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: when a PRI call must be moved to a different
- B channel at the request of the other endpoint, ensure that any
- DSP active on the original channel is moved to the new one
- (closes issue #11917) Reported by: mavetju Tested by: mavetju
-
-2008-03-05 15:17 +0000 [r106015] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c, include/asterisk/sched.h: Correctly
- initialize retransid in SIP, and ensure that the warning when
- failing to delete a schedule entry can actually hit the log.
- (closes issue #12140) Reported by: slavon Patches: sch2.patch
- uploaded by slavon (license 288) (Patch slightly modified by me)
-
-2008-03-05 01:52 +0000 [r105932] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, main/translate.c, include/asterisk/frame.h: Fix a bug
- that I just noticed in the RTP code. The calculation for setting
- the len field in an ast_frame of audio was wrong when G.722 is in
- use. The len field represents the number of ms of audio that the
- frame contains. It would have set the value to be twice what it
- should be.
-
-2008-03-04 18:10 +0000 [r105674-105676] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: In addition to setting the marker bit let's change
- our ssrc so they know for sure it is a different source.
-
- * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: When a
- new source of audio comes in (such as music on hold) make sure
- the marker bit gets set. (closes issue #10355) Reported by:
- wdecarne Patches: 10355.diff uploaded by file (license 11)
- (closes issue #11491) Reported by: kanderson
-
-2008-03-04 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.19-rc1 released.
-
-2008-03-04 04:31 +0000 [r105591] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Backport a minor bug fix from trunk that I found
- while doing random code cleanup. Properly break out of the loop
- when a context isn't found when verify that includes are valid.
-
-2008-03-03 18:06 +0000 [r105572] Jason Parker <jparker@digium.com>
-
- * res/snmp/agent.c: Fix type for astNumChannels. (closes issue
- #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg
- (license 192)
-
-2008-03-03 17:16 +0000 [r105563-105570] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c: In the case of an ast_channel allocation
- failure, take the local_pvt out of the pvt list before destroying
- it.
-
- * channels/chan_local.c: Fix a potential memory leak of the
- local_pvt struct when ast_channel allocation fails. Also, in
- passing, centralize the code necessary to destroy a local_pvt.
-
- * main/autoservice.c: Update the copyright information for
- autoservice. Most of the code in this file now is stuff that I
- have written recently ...
-
- * main/asterisk.c, main/channel.c, include/asterisk.h,
- main/autoservice.c: Merge in some changes from
- team/russell/autoservice-nochans-1.4 These changes fix up some
- dubious code that I came across while auditing what happens in
- the autoservice thread when there are no channels currently in
- autoservice. 1) Change it so that autoservice thread doesn't keep
- looping around calling ast_waitfor_n() on 0 channels twice a
- second. Instead, use a thread condition so that the thread
- properly goes to sleep and does not wake up until a channel is
- put into autoservice. This actually fixes an interesting bug, as
- well. If the autoservice thread is already running (almost always
- is the case), then when the thread goes from having 0 channels to
- have 1 channel to autoservice, that channel would have to wait
- for up to 1/2 of a second to have the first frame read from it.
- 2) Fix up the code in ast_waitfor_nandfds() for when it gets
- called with no channels and no fds to poll() on, such as was the
- case with the previous code for the autoservice thread. In this
- case, the code would call alloca(0), and pass the result as the
- first argument to poll(). In this case, the 2nd argument to
- poll() specified that there were no fds, so this invalid pointer
- shouldn't actually get dereferenced, but, this code makes it
- explicit and ensures the pointers are NULL unless we have valid
- data to put there. (related to issue #12116)
-
-2008-03-03 15:28 +0000 [r105557-105560] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: It is possible for no audio to pass between the
- current digit and next digit so expand logic that clears
- emulation to AST_FRAME_NULL. (closes issue #11911) Reported by:
- edgreenberg Patches: v1-11911.patch uploaded by dimas (license
- 88) Tested by: tbsky
-
- * channels/chan_sip.c: Add a comment to describe some logic.
- (closes issue #12120) Reported by: flefoll Patches:
- chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
- 244)
-
-2008-02-29 23:34 +0000 [r105409] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c: Fix a major bug in autoservice. There was a
- race condition in the handling of the list of channels in
- autoservice. The problem was that it was possible for a channel
- to get removed from autoservice and destroyed, while the
- autoservice thread was still messing with the channel. This led
- to memory corruption, and caused crashes. This explains multiple
- backtraces I have seen that have references to autoservice, but
- do to the nature of the issue (memory corruption), could cause
- crashes in a number of areas. (fixes the crash in BE-386) (closes
- issue #11694) (closes issue #11940) The following issues could be
- related. If you are the reporter of one of these, please update
- to include this fix and try again. (potentially fixes issue
- #11189) (potentially fixes issue #12107) (potentially fixes issue
- #11573) (potentially fixes issue #12008) (potentially fixes issue
- #11189) (potentially fixes issue #11993) (potentially fixes issue
- #11791)
-
-2008-02-29 14:47 +0000 [r105326] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Fix a potential memory leak
-
-2008-02-29 14:34 +0000 [r105296] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: If the message file does not exist, just
- return harmlessly, instead of crashing. (Closes issue #12108)
-
-2008-02-29 13:48 +0000 [r105261] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Bump up the size of the uniqueid variable.
- (closes issue #12107) Reported by: asgaroth
-
-2008-02-29 13:05 +0000 [r105209] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Automatically create new buddy upon reception
- of a presence stanza of type subscribed. (closes issue #12066)
- Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
- phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
- (license 73) Tested by: ffadaie, phsultan
-
-2008-02-28 22:23 +0000 [r105116] Russell Bryant <russell@digium.com>
-
- * main/utils.c, include/asterisk/lock.h: Fix a bug in the lock
- tracking code that was discovered by mmichelson. The issue is
- that if the lock history array was full, then the functions to
- mark a lock as acquired or not would adjust the stats for
- whatever lock is at the end of the array, which may not be
- itself. So, do a sanity check to make sure that we're updating
- lock info for the proper lock. (This explains the bizarre stats
- on lock #63 in BE-396, thanks Mark!)
-
-2008-02-28 21:56 +0000 [r105113] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.debian.asterisk: Update init script for LSB
- compat (closes issue #9843) Reported by: ibc Patches:
- rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by:
- paravoid
-
-2008-02-28 20:11 +0000 [r105059] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: When using autofill, members who are in use
- should be counted towards the number of available members to call
- if ringinuse is set to yes. Thanks to jmls who brought this issue
- up on IRC
-
-2008-02-28 19:20 +0000 [r104920-105005] Jason Parker <jparker@digium.com>
-
- * main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into
- applications, if we get NULL. This protects against possible
- segfaults in applications that may try to use data before
- checking length (ast_strdupa'ing it, for example) (closes issue
- #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
- uploaded by qwell (license 4)
-
- * channels/chan_skinny.c: According to a video at www.cisco.com,
- the 7921G supports 6 line appearances.
-
-2008-02-28 00:05 +0000 [r104868] Tilghman Lesher <tlesher@digium.com>
-
- * main/Makefile, build_tools/strip_nonapi: Compatibility fix for
- PPC64 (closes issue #12081) Reported by: jcollie Patches:
- asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
- Tested by: jcollie, Corydon76
-
-2008-02-27 21:49 +0000 [r104841] Mark Michelson <mmichelson@digium.com>
-
- * main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a
- lockable list. This is because in some cases, the ast_dial may
- exist in multiple threads due to asynchronous execution of its
- application, and I found some cases where race conditions could
- exist. 2. Check in ast_dial_join to be sure that the channel
- still exists before attempting to lock it, since it could have
- gotten hung up but the is_running_app flag on the
- ast_dial_channel may not have been cleared yet. (closes issue
- #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
- putnopvut (license 60) Tested by: jvandal
-
-2008-02-27 20:56 +0000 [r104787] Joshua Colp <jcolp@digium.com>
-
- * apps/app_chanspy.c: Don't loop around infinitely trying to spy on
- our own channel, and don't forget to free/detach the datastore
- upon hangup of the spy.
-
-2008-02-27 20:36 +0000 [r104783] Mark Michelson <mmichelson@digium.com>
-
- * main/file.c: Bump a couple of more buffers up by 2 so that
- annoying warnings aren't generated like crazy on every
- fileexists_core call.
-
-2008-02-27 18:15 +0000 [r104704] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c: Ensure the session ID can't be 0.
-
-2008-02-27 17:41 +0000 [r104665] Joshua Colp <jcolp@digium.com>
-
- * main/file.c: Bump up the buffer by 2.
-
-2008-02-27 17:33 +0000 [r104625] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c: Fix a problem in ChanSpy where it could get
- stuck in an infinite loop without being able to detect that the
- calling channel hung up. (closes issue #12076, reported by junky,
- patched by me)
-
-2008-02-27 17:26 +0000 [r104598] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Inherit language from the transfering channel
- on a blind transfer. (closes issue #11682) Reported by: caio1982
- Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982
- (license 22) Tested by: caio1982, victoryure
-
-2008-02-27 17:07 +0000 [r104596] Joshua Colp <jcolp@digium.com>
-
- * main/loader.c: Use the lock (which already existed, it just
- wasn't used) on the updaters list to protect the contents instead
- of the overall module list lock. (closes issue #12080) Reported
- by: ChaseVenters
-
-2008-02-27 16:53 +0000 [r104593] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/file.c: fallback to standard English prompts properly when
- using new prompt directory layout (closes issue #11831) Reported
- by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license
- 20) (modified by me to improve code and conform rest of function
- to coding guidelines)
-
-2008-02-27 16:45 +0000 [r104591] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: When we receive a known alarm, make sure
- that the unknown alarm flag is not still set to make sure that
- when we come back out of alarm, it gets reported in the log and
- manager interface (after discussion with tzafrir on the -dev
- list)
-
-2008-02-27 15:52 +0000 [r104536] Joshua Colp <jcolp@digium.com>
-
- * res/res_smdi.c: Only stop the MWI monitor thread if it was
- actually started. (closes issue #12086) Reported by: francesco_r
-
-2008-02-27 01:15 +0000 [r104332-104334] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c: Avoid some recursion in the cleanup code for
- the chanspy datastore (closes issue #12076, reported by junky,
- patched by me)
-
- * channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as
- an alarm. However, Asterisk 1.4 does not know what to do with
- these alarms. Only Asterisk 1.6 cares about it. So, if we get an
- unknown alarm in chan_zap, don't generate confusing log messages
- about it.
-
-2008-02-26 18:26 +0000 [r104132-104141] Jason Parker <jparker@digium.com>
-
- * Makefile: Add badshell to .PHONY target (thanks Kevin)
-
- * Makefile: Since all shells aren't as awesome as bash, we have to
- fail if somebody tries to use a literal "~" in DESTDIR.
-
- * sounds/Makefile: Revert previous abspath change. ...abspath is
- new in GNU make 3.81. I feel so...defeated. Must find new fix!
-
- * sounds/Makefile: Fix a very bizarre issue we were seeing with our
- buildbot when using a DESTDIR that wasn't an absolute path (such
- as DESTDIR=~/asterisk-1.4). Apparently what was happening, was
- that some of the targets were being expanded to the full path, so
- $@ ended up being /root/asterisk-1.4/[...]/ rather than
- ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
- in GNU make. (*cough*
- http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
-
-2008-02-26 00:25 +0000 [r104119] Russell Bryant <russell@digium.com>
-
- * include/asterisk/smdi.h, apps/app_voicemail.c,
- channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample:
- Merge changes from team/russell/smdi-1.4 This commit brings in a
- significant set of changes to the SMDI support in Asterisk. There
- were a number of bugs in the current implementation, most notably
- being that it was very likely on busy systems to pop off the
- wrong message from the SMDI message queue. So, this set of
- changes fixes the issues discovered as well as introducing some
- new ways to use the SMDI support which are required to avoid the
- bugs with grabbing the wrong message off of the queue. This code
- introduces a new interface to SMDI, with two dialplan functions.
- First, you get an SMDI message in the dialplan using
- SMDI_MSG_RETRIEVE() and then you access details in the message
- using the SMDI_MSG() function. A side benefit of this is that it
- now supports more than just chan_zap. For example, with this
- implementation, you can have some FXO lines being terminated on a
- SIP gateway, but the SMDI link in Asterisk. Another issue with
- the current implementation is that it is quite common that the
- station ID that comes in on the SMDI link is not necessarily the
- same as the Asterisk voicemail box. There are now additional
- directives in the smdi.conf configuration file which let you map
- SMDI station IDs to Asterisk voicemail boxes. Yet another issue
- with the current SMDI support was related to MWI reporting over
- the SMDI link. The current code could only report a MWI change
- when the change was made by someone calling into voicemail. If
- the change was made by some other entity (such as with IMAP
- storage, or with a web interface of some kind), then the MWI
- change would never be sent. The SMDI module can now poll for MWI
- changes if configured to do so. This work was inspired by and
- primarily done for the University of Pennsylvania. (also related
- to issue #9260)
-
-2008-02-26 00:03 +0000 [r104111] Jason Parker <jparker@digium.com>
-
- * channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris.
- (closes issue #12050) Reported by: asgaroth Patches: 12050.patch
- uploaded by putnopvut (license 60)
-
-2008-02-25 23:42 +0000 [r104102-104106] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c: This patch fixes some pretty significant
- problems with how app_chanspy handles pointers to channels that
- are being spied upon. It was very likely that a crash would occur
- if the channel being spied upon hung up. This was because the
- current ast_channel handling _requires_ that the object is locked
- or else it could disappear at any time (except in the owning
- channel thread). So, this patch uses some channel datastore magic
- on the spied upon channel to be able to detect if and when the
- channel goes away. (closes issue #11877) (patch written by me,
- but thanks to kpfleming for the idea, and to file for review)
-
- * main/utils.c: Improve the lock tracking code a bit so that a
- bunch of old locks that threads failed to lock don't sit around
- in the history. When a lock is first locked, this checks to see
- if the last lock in the list was one that was failed to be
- locked. If it is, then that was a lock that we're no longer
- sitting in a trylock loop trying to lock, so just remove it.
- (inspired by issue #11712)
-
-2008-02-25 21:37 +0000 [r104095] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make it so a users.conf user creates both a
- SIP peer and a SIP user. The user will be used for inbound
- authentication for the device, and peer will be used for placing
- calls to the device. (closes issue #9044) Reported by: queuetue
- Patches: sip-gui-friend.diff uploaded by qwell (license 4)
-
-2008-02-25 21:31 +0000 [r104094] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: If the destination folder is full, don't
- delete a message when exiting. (closes issue #12065) Reported by:
- selsky Patch by: (myself)
-
-2008-02-25 20:49 +0000 [r104092] Jason Parker <jparker@digium.com>
-
- * main/config.c: Allow the use of #include and #exec in situations
- where the max include depth was only 1. Specifically, this fixes
- using #include and #exec in extconfig.conf. This was basically
- caused because the config file itself raises the include level to
- 1. I opted not to raise the include limit, because recursion here
- could cause very bizarre behavior. Pointed out, and tested by
- jmls (closes issue #12064)
-
-2008-02-25 18:38 +0000 [r104086] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c: Ensure that the channel doesn't disappear
- in agent_logoff(). If it does, it could cause a crash. (fixes the
- crash reported in BE-396)
-
-2008-02-25 16:16 +0000 [r104082-104084] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If a resubscription comes in for a dialog we
- no longer know about tell the remote side that the dialog does
- not exist so they subscribe again using a new dialog. (closes
- issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff
- uploaded by file (license 11)
-
- * channels/chan_sip.c: Due to recent changes tag will no longer be
- NULL if not present so we have to use ast_strlen_zero to see if
- it's actually blank. (closes issue #12061) Reported by: flefoll
- Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by
- flefoll (license 244)
-
-2008-02-22 22:45 +0000 [r104037] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Backwards debug message. (closes issue
- #12052) Reported by: flefoll Patches:
- chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license
- 244)
-
-2008-02-21 21:05 +0000 [r104026-104027] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c: And as a followup to revision 104026,
- completely remove event-related calls from a section of code
- where we know there was no event to handle or get.
-
- * channels/chan_zap.c: Remove an incorrect debug message. It
- reported that it had received a specific event and tried to
- report which event was received. What actually was happening was
- that it was reporting the number of bytes returned from a call to
- read(). Thanks to Jared Smith for bringing the issue up on IRC
-
-2008-02-21 14:33 +0000 [r104015] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c: reduce the likelihood that HTTP Manager session
- ids will consist of primarily '1' bits
-
-2008-02-20 22:32 +0000 [r103956] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Clear up confusion when viewing the
- QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
- user's perspective, the queue does exist, we shouldn't tell them
- we couldn't find the queue. Instead since it is a dead queue,
- report a 0 waiting count This issue was brought up on IRC by jmls
-
-2008-02-20 22:06 +0000 [r103953] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c: Don't wait for additional digits when
- overlap dialing is enabled if the setup message contains the
- sending_complete information element. (closes issue #11785)
- Reported by: klaus3000 Patches:
- sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by
- klaus3000 (license 65)
-
-2008-02-20 21:40 +0000 [r103904] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c: Fix a crash if the channel becomes NULL
- while attempting to lock it. (closes issue #12039) Reported by:
- danpwi
-
-2008-02-20 17:53 +0000 [r103845] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/localtime.c: Compat fix for Solaris (closes issue
- #12022) Reported by: asgaroth Patches:
- 20080219__bug12022.diff.txt uploaded by Corydon76 (license 14)
- Tested by: asgaroth
-
-2008-02-19 20:28 +0000 [r103823] Joshua Colp <jcolp@digium.com>
-
- * channels/h323/ast_h323.cxx: Send CallerID Name in setup message.
- (closes issue #11241) Reported by: tusar Patches:
- h323id_as_callerid_name.patch uploaded by tusar (license 344)
-
-2008-02-19 20:02 +0000 [r103821] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c: Account for the fact that the "other"
- channel can disappear while the local pvt is not locked. (fixes a
- problem introduced in rev 100581) (closes issue #12012) Reported
- by: stevedavies Patch by me
-
-2008-02-19 17:31 +0000 [r103807-103812] Joshua Colp <jcolp@digium.com>
-
- * configure, configure.ac: Don't look for launchd when cross
- compiling. (closes issue #12029) Reported by: ovi
-
- * channels/chan_sip.c: Fix building of chan_sip.
-
-2008-02-19 10:27 +0000 [r103806] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Make sure we send error replies correctly by
- checking the via header.
-
-2008-02-18 23:56 +0000 [r103801] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Ensure that emulated DTMFs do not get interrupted
- by another begin frame. (closes issue #11740) Reported by: gserra
- Patches: v1-11740.patch uploaded by dimas (license 88) (closes
- issue #11955) Reported by: tsearle (closes issue #10530) Reported
- by: xmarksthespot
-
-2008-02-18 22:28 +0000 [r103790-103795] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Fix previous commit so that we actually
- disable echocanbridged if echocancel is off.
-
- * channels/chan_zap.c: Correct a message when echocancelwhenbridged
- is on, but echocancel is not. Issue #12019
-
-2008-02-18 20:52 +0000 [r103786] Mark Michelson <mmichelson@digium.com>
-
- * main/app.c: There was an invalid assumption when calculating the
- duration of a file that the filestream in question was created
- properly. Unfortunately this led to a segfault in the situation
- where an unknown format was specified in voicemail.conf and a
- voicemail was recorded. Now, we first check to be sure that the
- stream was written correctly or else assume a zero duration.
- (closes issue #12021) Reported by: jakep Tested by: putnopvut
-
-2008-02-18 17:31 +0000 [r103780] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c, channels/chan_sip.c: When a SIP channel is being
- auto-destroyed, it's possible for it to still be in bridge code.
- When that happens, we crash. Delay the RTP destruction until the
- bridge is ended. (closes issue #11960) Reported by: norman
- Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76
- (license 14) Tested by: norman
-
-2008-02-18 16:37 +0000 [r103770] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c: Fix a linked list corruption that under the
- right circumstances could lead to a looped list, meaning it will
- traverse forever. (closes issue #11818) Reported by: michael-fig
- Patches: 11818.patch uploaded by putnopvut (license 60) Tested
- by: michael-fig
-
-2008-02-18 16:11 +0000 [r103763-103768] Joshua Colp <jcolp@digium.com>
-
- * main/asterisk.c: Backport fix from issue #9325. (closes issue
- #11980) Reported by: rbrunka
-
- * channels/chan_sip.c: Don't care if the extension given doesn't
- exist for subscription based MWI.
-
-2008-02-15 23:31 +0000 [r103726-103741] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race
- condition (closes issue #11780) Reported by: guillecabeza
- Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
- 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
- 380)
-
- * main/loader.c: In the case that you try to directly reload a
- module has returned AST_MODULE_LOAD_DECLINE, log a message
- indicating that the module is not fully initialized and must be
- initialized using "module load".
-
- * main/loader.c: Don't attempt to execute the reload callback for a
- module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
- that was reported against chan_console in trunk. (closes issue
- #11953, reported by junky, fixed by me)
-
-2008-02-15 17:26 +0000 [r103688-103722] Mark Michelson <mmichelson@digium.com>
-
- * doc/imapstorage.txt, configure, configure.ac: Final round of
- changes for configure script logic for IMAP Now if a directory is
- specified, then we will search that directory for a source
- installation of the IMAP toolkit. If none is found, then we will
- use that directory as the basis for detecting a package
- installation of the IMAP c-client. If that check fails, then
- configure will fail.
-
- * configure, configure.ac: Fix a bit of wrong logic in the
- configure script that caused problems when trying to configure
- without IMAP. Patch suggestion from phsultan, but I modified it
- slightly. (closes issue #12003) Reported by: pj Tested by:
- putnopvut
-
- * doc/imapstorage.txt, configure, configure.ac: I apparently
- misunderstood one of the requirements of this configure change.
- Now, if a source directory is specified with the --with-imap
- option, and a valid source installation is not detected there,
- then configure will fail and will not check for a package
- installation.
-
- * doc/imapstorage.txt: Make a small clarification in the
- documentation
-
- * doc/imapstorage.txt: Update documentation regarding configuration
- of IMAP
-
- * apps/app_voicemail.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Change to the
- configure logic regarding IMAP. Prior to this commit, if you
- wished to configure Asterisk with IMAP support, you would use the
- --with-imap configure switch in one of the following two ways:
- --with-imap=/some/directory would look in the directory specified
- for a UW IMAP source installation --with-imap would assume that
- you had imap-2004g installed in .. relative to the Asterisk
- source With this set of changes the two above options still work
- the same, but there are two new behaviors, too.
- --with-imap=system will assume that you have -libc-client.so
- where you store your shared objects and will attempt to find
- c-client headers in your include path either in the imap or
- c-client directory. If either of the two original methods of
- specifying the imap option should fail, then the check for
- --with-imap =system will be performed in addition. It is only
- after this "system" check that failure can happen.
-
- * apps/app_voicemail.c: Fix build for non-IMAP builds
-
- * apps/app_voicemail.c: Fix the new message count if delete=yes
- when using IMAP storage. (closes issue #11406) Reported by:
- jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
- 50) Tested by: jaroth
-
-2008-02-14 19:51 +0000 [r103683-103684] Jason Parker <jparker@digium.com>
-
- * funcs/func_cdr.c: swap location for this..
-
- * funcs/func_cdr.c: Document the 'l' option to the CDR() function.
- (Thanks voipgate for pointing out the option, and Leif for
- providing text for it.) Closes issue #11695.
-
-2008-02-13 06:25 +0000 [r103556-103607] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_agent.c: We aren't talking to ourselves; we're
- talking to someone else. (closes issue #11771) Reported by:
- msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982
- (license 22) Tested by: caio1982, msetim
-
- * apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is
- not loaded (which is a symbol dependency) (closes issue #11760)
- Reported by: non-poster Patches: 20080114__bug11760.diff.txt
- uploaded by Corydon76 (license 14) Tested by: Corydon76,
- non-poster, jamesgolovich
-
-2008-02-12 22:24 +0000 [r103503-103504] Jason Parker <jparker@digium.com>
-
- * main/asterisk.c: revert accidental change from last commit. oops
-
- * contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition
- that was impossible.
-
-2008-02-12 15:09 +0000 [r103324-103385] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Even if no CallerID name or number has been
- provided by the remote party still use the configured sip.conf
- ones. (closes issue #11977) Reported by: pj
-
- * apps/app_meetme.c: If entering a conference with the 'w' option
- ensure that we can't listen or speak until the marked user
- appears. (closes issue #11835) Reported by: alanmcmillan
-
-2008-02-11 17:05 +0000 [r103315] Kevin P. Fleming <kpfleming@digium.com>
-
- * configs/zapata.conf.sample: improve 2BCT documentation a bit
- (thanks Jared)
-
-2008-02-09 06:23 +0000 [r103197] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Commit fix for being unable to send
- voicemail from VoiceMailMain Reported by: William F Acker (via
- the -users mailing list) Patch by: Corydon76 (license 14)
-
-2008-02-08 18:48 +0000 [r103070-103120] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Prevent a potential three-thread deadlock. Also
- added a comment block to explicitly state the locking order
- necessary inside app_queue. (closes issue #11862) Reported by:
- flujan Patches: 11862.patch uploaded by putnopvut (license 60)
- Tested by: flujan
-
- * channels/chan_iax2.c: Yield the thread and return -1 if the ioctl
- fails for Zaptel timing device. (closes issue #11891) Reported
- by: tzafrir
-
-2008-02-08 15:08 +0000 [r102968] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Make sure the presence of dbsecret is
- factored into user scoring. (closes issue #11952) Reported by:
- bbhoss
-
-2008-02-07 19:53 +0000 [r102858] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Specify which digit string was matched in
- debug message. (closes issue #11949) Reported by: dimas Patches:
- v1-feature-debug.patch uploaded by dimas (license 88)
-
-2008-02-07 16:41 +0000 [r102807] Kevin P. Fleming <kpfleming@digium.com>
-
- * configs/zapata.conf.sample: document usage of 'transfer'
- configuration option for ISDN PRI switch-side transfers
-
-2008-02-06 17:59 +0000 [r102653-102725] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Only consider a T.38-only INVITE compatible
- if we have both a joint capability between us and them and if
- they provided T.38.
-
- * main/global_datastores.c: Add missing header file and
- ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by:
- snuffy
-
-2008-02-06 15:19 +0000 [r102651] Russell Bryant <russell@digium.com>
-
- * configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so
- that it gets inherited by outbound channels. (due to a discussion
- between me and a user via email)
-
-2008-02-06 11:48 +0000 [r102627] Kevin P. Fleming <kpfleming@digium.com>
-
- * pbx/Makefile, res/Makefile: ensure that all remaining
- multi-object modules are built using their proper CFLAGS and
- include directory paths
-
-2008-02-06 00:26 +0000 [r102576] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Move around some defines to unbreak ODBC
- storage. (closes issue #11932) Reported by: snuffy
-
-2008-02-05 20:02 +0000 [r102453] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes
- issue #11919) Reported by: eferro Patches:
- mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
- Tested by: eferro
-
-2008-02-05 19:52 +0000 [r102450] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If a REGISTER attempt comes in that is a
- retransmission of a previous REGISTER do not create a new nonce
- value. (issue #BE-381)
-
-2008-02-05 17:15 +0000 [r102425] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/Makefile: ensure that components of chan_misdn.so are
- built using any special build options that the configure script
- generated (reported by Philipp Kempgen on asterisk-dev)
-
-2008-02-05 15:09 +0000 [r102378] Joshua Colp <jcolp@digium.com>
-
- * res/res_clioriginate.c: Perform dialing asynchronously when using
- the originate CLI command so the CLI does not appear to block.
- (closes issue #11927) Reported by: bbhoss
-
-2008-02-04 21:06 +0000 [r102214-102323] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, utils/muted.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Cross-platform
- fix: OS X now deprecates the use of the daemon(3) API. (closes
- issue #11908) Reported by: oej Patches:
- 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76
-
- * funcs/func_strings.c: Missing braces. (closes issue #11912)
- Reported by: dimas Patches: sprintf.patch uploaded by dimas
- (license 88)
-
-2008-02-03 16:38 +0000 [r102090-102142] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE
- (according to RFC 3261) (closes issue #9492) Reported by:
- kryptolus Patches: bug9492.txt uploaded by oej (license 306)
- Tested by: oej
-
- * channels/chan_sip.c: Handle ACK and CANCEL in an invite
- transaction - even if we get INFO transactions during the actual
- call setup. (closes issue #10567) Reported by: jacksch Tested by:
- oej Patch by: oej inspired by suggestions from neutrino88 in the
- bug tracker
-
-2008-02-01 23:06 +0000 [r101989] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns
- out that even though G.722 is 16 kHz, it is supposed to specified
- as 8 kHz in the RTP, and RTP timestamps are supposed to be
- calculated based on 8 kHz. (Apparently this is due to a bug in a
- spec, but people follow it anyway, because it's the spec ...)
-
-2008-02-01 21:54 +0000 [r101894-101942] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Fix the VM_DUR variable for forwarded
- voicemail, and fixed several other bugs while I'm in the area.
- (closes issue #11615) Reported by: jamessan Patches:
- 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76, jamessan
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4: Change detection of getifaddrs to use
- AST_C_COMPILE_CHECK, backported from trunk (as suggested by
- kpfleming)
-
-2008-02-01 17:41 +0000 [r101822] Jason Parker <jparker@digium.com>
-
- * apps/app_authenticate.c: Remove a needless (and incorrect) call
- to feof() after fgets(). This would have exited the loop early if
- you had an authentication file with no newline at the end.
-
-2008-02-01 17:27 +0000 [r101818-101820] Russell Bryant <russell@digium.com>
-
- * apps/app_authenticate.c: off by one error
-
- * apps/app_authenticate.c: Don't overwrite the last character of a
- line if it's not a newline. This would happen if the last line in
- the file doesn't have a newline. (pointed out by Qwell)
-
-2008-02-01 15:55 +0000 [r101772] Tilghman Lesher <tlesher@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/acl.c: Compatibility fix for OpenWRT (reported by Brian
- Capouch via the mailing list)
-
-2008-02-01 00:32 +0000 [r101693] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Add some more sanity checking on IAX2 dial
- strings for the case that no peer or hostname was provided, which
- is the one part of the dial string that is absolutely required.
- If it's not there, bail out. (closes issue #11897) Reported by
- sokhapkin Patch by me
-
-2008-02-01 00:06 +0000 [r101649] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle
- periods of no channel activity" (closes issue #9256) Reported by:
- cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
- uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
- rjain
-
-2008-01-31 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.18 released.
-
-2008-01-31 23:10 +0000 [r101601] Russell Bryant <russell@digium.com>
-
- * main/translate.c, main/file.c: Fix a couple of places where
- ast_frfree() was not called on a frame that came from a
- translator. This showed itself by g729 decoders not getting
- released. Since the flag inside the translator frame never got
- unset by freeing the frame to indicate it was no longer in use,
- the translators never got destroyed, and thus the g729 licenses
- were not released. (closes issue #11892) Reported by: xrg
- Patches: 11892.diff uploaded by russell (license 2) Tested by:
- xrg, russell
-
-2008-01-31 21:00 +0000 [r101531] Mark Michelson <mmichelson@digium.com>
-
- * res/res_monitor.c: 1. Prevent the addition of an extra '/' to the
- beginning of an absolute pathname. 2. If ast_monitor_change_fname
- is called and the new filename is the same as the old, then exit
- early and don't set the filename_changed field in the monitor
- structure. Setting it in this case was causing ast_monitor_stop
- to erroneously delete them. (closes issue #11741) Reported by:
- garlew Tested by: putnopvut
-
-2008-01-31 19:52 +0000 [r101482] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes
- for struct in_addr funkiness. Issue #11885, patch by snuffy.
-
-2008-01-31 19:30 +0000 [r101480] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: closes issue #11845; that's the one where there's a
- 1004 byte cdr leak with every AMI Redirect to a zap channel
-
-2008-01-31 19:17 +0000 [r101413-101433] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c: Add more missing locking of the agents
- list ...
-
- * channels/chan_agent.c: Move the locking from find_agent() into
- the agent dialplan function handler to ensure that the agent
- doesn't disappear while we're looking at it.
-
- * channels/chan_agent.c: Add missing locking to the find_agent()
- function.
-
-2008-01-30 15:41 +0000 [r101222] Joshua Colp <jcolp@digium.com>
-
- * main/slinfactory.c: Fix an issue where if a frame of higher
- sample size preceeded a frame of lower sample size and
- ast_slinfactory_read was called with a sample size of the
- combined values or higher a crash would happen. (closes issue
- #11878) Reported by: stuarth
-
-2008-01-30 15:34 +0000 [r101219] Jason Parker <jparker@digium.com>
-
- * configs/extensions.conf.sample: Change default config to use
- descending channel order of groups, rather than ascending. Fixes
- a potential source of confusion in glare-type situations. Issue
- 11875, reported by JimVanM.
-
-2008-01-30 15:23 +0000 [r101216] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fix a logic error with regards to autofill.
- Prior to this change, it was possible for a caller to go out of
- turn if autofill were enabled and callers ahead in the queue were
- attempting to call a member. This change fixes this.
-
-2008-01-30 11:20 +0000 [r101152] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Stop musiconhold on attended transfer.
- (closes issue #11872) Reported by: gareth Patches:
- svn-101018.patch uploaded by gareth (license 208)
-
-2008-01-29 23:50 +0000 [r101080] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * build_tools/make_version: updated build_tools to handle the
- autotag directory structure changes; changes related to BE-353.
- Patch by The Russell and reviewed by The Me.
-
-2008-01-29 23:02 +0000 [r100973-101035] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Remove a memory leak from updating realtime
- queues
-
- * apps/app_queue.c: Fixing an erroneous return value returned when
- attempting to pause or unpause a queue member fails. Fixes
- BE-366, thanks to John Bigelow for writing the patch.
-
-2008-01-29 17:57 +0000 [r100934] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c: Don't forget to record the channel so we
- know whether it is bridged or not later. (closes issue #11811)
- Reported by: slavon
-
-2008-01-29 17:43 +0000 [r100932] Russell Bryant <russell@digium.com>
-
- * main/Makefile: Fix the last couple of issues related to building
- from a path that contains spaces. (closes issue #11834)
-
-2008-01-29 17:41 +0000 [r100930] Jason Parker <jparker@digium.com>
-
- * channels/misdn_config.c: Initialize an array to 0s if config
- option not specified. (closes issue #11860) Patches:
- misdn_get_config.v1.diff uploaded by IgorG (license 20)
-
-2008-01-29 17:21 +0000 [r100882-100922] Russell Bryant <russell@digium.com>
-
- * Makefile: Use GNU make magic instead of shell magic to escape
- spaces in the working directory. (related to issue #11834)
-
- * Makefile: Fix building Asterisk when the working path has spaces
- in it. (closes issue #11834) Reported by: spendergrass Patched
- by: me
-
-2008-01-29 16:10 +0000 [r100835] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Allow zap groups above 30 to work properly.
- (closes issue #11590) Reported by: tbsky
-
-2008-01-29 10:36 +0000 [r100793] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed potential segfault in misdn show
- channels CLI command
-
-2008-01-29 08:26 +0000 [r100740] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: (closes issue #11736) Reported by: MVF
- Patches: bug11736-2.diff uploaded by oej (license 306) Tested by:
- oej, MVF, revolution (russellb: This was the showstopper for the
- release.)
-
-2008-01-28 21:02 +0000 [r100675] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly
- (went to 't' instead of 'T')
-
-2008-01-28 20:55 +0000 [r100673] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of
- chan_vpb. It is alive and well.
-
-2008-01-28 20:42 +0000 [r100672] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put
- greeting files into the database like we do with the others.
- Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded
- by dimas (license 88)
-
-2008-01-28 18:34 +0000 [r100626-100629] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: For some reason, the use of this strdupa()
- is leading to memory corruption on freebsd sparc64. This trivial
- workaround fixes it. (closes issue #10300, closes issue #11857,
- reported by mattias04 and Home-of-the-Brave)
-
- * res/res_features.c: Fix a crash in ast_masq_park_call() (issue
- #11342) Reported by: DEA Patches: res_features-park.txt uploaded
- by DEA (license 3)
-
-2008-01-28 18:23 +0000 [r100624] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Correct a comment which made little/no
- sense.
-
-2008-01-28 17:15 +0000 [r100581] Russell Bryant <russell@digium.com>
-
- * main/channel.c, channels/chan_local.c,
- include/asterisk/channel.h: Make some deadlock related fixes.
- These bugs were discovered and reported internally at Digium by
- Steve Pitts. - Fix up chan_local to ensure that the channel lock
- is held before the local pvt lock. - Don't hold the channel lock
- when executing the timing function, as it can cause a deadlock
- when using chan_local. This actually changes the code back to be
- how it was before the change for issue #10765. But, I added some
- other locking that I think will prevent the problem reported
- there, as well.
-
-2008-01-27 21:59 +0000 [r100465] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c, channels/chan_mgcp.c, main/cdr.c,
- channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c,
- channels/chan_h323.c, include/asterisk/sched.h, main/file.c,
- pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from
- the scheduler, ignoring the return value could possibly cause
- memory to be accessed after it is freed, which causes all sorts
- of random memory corruption. Instead, if a deletion fails, wait a
- bit and try again (noting that another thread could change our
- taskid value). (closes issue #11386) Reported by: flujan Patches:
- 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76, flujan, stuarth`
-
-2008-01-25 22:32 +0000 [r100418] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is
- now preferred that users of Voicetronix products use chan_zap in
- combination with their zaptel drivers.
-
-2008-01-25 21:24 +0000 [r100378] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: This would have never been true, since we're
- passing (sizeof(req.data) - 1) as the len to recvfrom().
-
-2008-01-24 21:57 +0000 [r100264] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/app.h: make these macros not assume that the
- only other field in the structure is 'argc'... this is true when
- someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
- to define your own structure as long as it has the right fields
-
-2008-01-24 17:22 +0000 [r100164] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Update main Asterisk copyright info to 2008
-
-2008-01-24 16:41 +0000 [r100138] Jason Parker <jparker@digium.com>
-
- * main/acl.c: Fix compilation on Solaris. (closes issue #11832)
- Patches: bug-11832.diff uploaded by snuffy (license 35)
-
-2008-01-23 21:07 +0000 [r99977-99978] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Second attempt. Don't change invitestate
- when receiving 18x messages in CANCEL state. (issue #11736)
- Reported by: MVF Patch by oej.
-
- * channels/chan_sip.c: Make sure we don't cancel destruction on
- calls in CANCEL state, even if we get 183 while waiting for
- answer on our CANCEL. (issue #11736) Reported by: MVF Patches:
- bug11736.txt uploaded by oej (license 306) Tested by: MVF
-
-2008-01-23 20:25 +0000 [r99975] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_externalivr.c: Fixing a typo.
-
-2008-01-23 17:46 +0000 [r99923] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c: ChanSpy issues a beep when it starts at the
- beginning of a list of channels to potentially spy on. However,
- if there were no matching channels, it would beep at you over and
- over, which is pretty annoying. Now, it will only beep once in
- the case that there are no channels to spy on, but it will still
- beep again once it reaches the beginning of the channel list
- again. (closes issue #11738, patched by me)
-
-2008-01-23 16:18 +0000 [r99878] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: These flag tests were illogical. They were
- testing sip_peer flags on a sip_pvt. Thanks to Russell for
- helping to get this odd problem figured out.
-
-2008-01-23 04:31 +0000 [r99718-99777] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: When we reset the password via an external
- command, we should also reset the password stored in the
- in-memory list, too (otherwise it doesn't really take effect).
- (closes issue #11809) Reported by: davetroy Patches:
- fix_externpass.diff uploaded by davetroy (license 384)
-
- * res/res_odbc.c: Oops, should have checked for a NULL obj, here,
- too
-
- * main/acl.c: Just confirmed that all current platforms need this
- header file
-
-2008-01-22 20:56 +0000 [r99652] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Thanks to Russell's education I realize that
- BUFSIZ has changed since I learned the C language over 20 years
- ago... Resetting chan_sip to the size of BUFSIZ that I expected
- in my old head to avoid to heavy memory allocations on some
- systems.
-
-2008-01-22 20:34 +0000 [r99643] Tilghman Lesher <tlesher@digium.com>
-
- * main/acl.c: Fix the defines for OS X (and Solaris, too)
-
-2008-01-22 17:41 +0000 [r99592-99594] Olle Johansson <oej@edvina.net>
-
- * channels/chan_local.c, res/res_features.c, channels/chan_agent.c,
- apps/app_followme.c: Add more dependencies on chan_local and add
- a note to the description of chan_local so that people don't
- disable it in menuselect just to clean up.
-
- * apps/app_dial.c: Add dependency on chan_local to app_dial. Dial
- still runs without chan_local, but will be missing forwarding
- functionality.
-
-2008-01-22 16:54 +0000 [r99540] Tilghman Lesher <tlesher@digium.com>
-
- * main/acl.c: Ensure that we can get an address even when we don't
- have a default route. (closes issue #9225) Reported by: junky
- Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
- (license 14) Tested by: oej, loloski, sergee
-
-2008-01-22 15:08 +0000 [r99501] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Cleaning up some documentation that led to
- confusion in a bug report
-
-2008-01-21 23:55 +0000 [r99426] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c: Fixing an issue wherein monitoring local
- channels was not possible. During a channel masquerade, the
- monitors on the two channels involved are swapped. In 99% of the
- cases this results in the desired effect. However, if monitoring
- a local channel, this caused the monitor which was on the local
- channel to get moved onto a channel which is immediately hung up
- after the masquerade has completed. By swapping the monitors
- prior to the masquerade, we avoid the problem by tricking the
- masquerade into placing the monitor back onto the channel where
- we want it. During the investigation of the issue, the channel's
- monitor was the only thing that was swapped in such a manner
- which did not make sense to have done. All other variable
- swapping made sense.
-
-2008-01-21 18:11 +0000 [r99341] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c, configs/res_odbc.conf.sample,
- include/asterisk/res_odbc.h: Permit the user to specify number of
- seconds that a connection may remain idle, which fixes a crash on
- reconnect with the MyODBC driver. (closes issue #11798) Reported
- by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
- uploaded by Corydon76 (license 14) Tested by: mvanbaak
-
-2008-01-21 16:01 +0000 [r99301] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Bump the buffer size for Via headers up to
- 512. There are some exceptionally large Via headers out there.
- (closes issue #11783) Reported by: ofirroval
-
-2008-01-19 10:05 +0000 [r99187] Russell Bryant <russell@digium.com>
-
- * main/slinfactory.c: Fix a couple of memory leaks with frame
- handling. Specifically, ast_frame_free() needed to be called on
- the frame that came from the translator to signed linear.
-
-2008-01-18 22:57 +0000 [r99127] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/channel.h: Remove the __ in front of the unused
- variable. This causes some compilers to freak out.
-
-2008-01-18 21:37 +0000 [r99079-99081] Russell Bryant <russell@digium.com>
-
- * include/asterisk/translate.h, main/frame.c: Revert adding the
- packed attribute, as it really doesn't make sense why that would
- do any good. Fix the real bug, which is to do the check to see if
- the frame came from a translator at the beginning of
- ast_frame_free(), instead of at the end. This ensures that it
- always gets checked, even if none of the parts of the frame are
- malloc'd, and also ensures that we aren't looking at free'd
- memory in the case that it is a malloc'd frame. (closes issue
- #11792, reported by explidous, patched by me)
-
- * include/asterisk/translate.h: Since we're relying on the offset
- between the frame and the beginning of the translator pvt struct,
- set the packed attribute to make sure we get to the right place.
- (potential fix for issue #11792)
-
-2008-01-18 17:13 +0000 [r99032] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: This should at least temporarily fix a
- problem where the 't' Dial option is incorrectly passed to the
- transferee when built-in attended transfers are used. There is
- still a problem with 'T', but better to fix some problems than no
- problems while we work on it. (closes issue #7904) Reported by:
- k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee
- (license 138) Tested by: sergee, otherwiseguy
-
-2008-01-17 23:42 +0000 [r99007-99014] Pari Nannapaneni <paripurnachand@digium.com>
-
- * configs/cdr.conf.sample: doh! revert a revert of a revert
- (changed by mistake in 99010)
-
- * main/manager.c, configs/cdr.conf.sample: missed that one while
- reverting
-
- * main/manager.c: reverting 99001 - We need the Max-Age for
- extending the life of cookie mansession_id
-
-2008-01-17 22:37 +0000 [r99004] Russell Bryant <russell@digium.com>
-
- * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h:
- Have IAX2 optimize the codec translation path just like chan_sip
- does it. If the caller's codec is in our codec list, move it to
- the top to avoid transcoding. (closes issue #10500) Reported by:
- stevedavies Patches: iax-prefer-current-codec.patch uploaded by
- stevedavies (license 184) iax-prefer-current-codec.1.4.patch
- uploaded by stevedavies (license 184) Tested by: stevedavies, pj,
- sheldonh
-
-2008-01-17 21:31 +0000 [r99001] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c: we should only send the Set-Cookie header to the
- browser on the first response after creating a manager session,
- not on every response (doing so causes the browser to clear any
- local cookies it may have associated with the session)
-
-2008-01-17 16:19 +0000 [r98991] Jason Parker <jparker@digium.com>
-
- * configs/zapata.conf.sample: Add a clarification about the
- immediate= option of zapata.conf Issue 11784, patch by klaus3000.
-
-2008-01-16 22:36 +0000 [r98982] Russell Bryant <russell@digium.com>
-
- * .cleancount, include/asterisk/channel.h: Add an unused pointer to
- the ast_channel struct. This makes the ast_channel structure
- retain the same size as it had in previous 1.4 releases. Also,
- all of the offsets for members in the structure are still the
- same (except for the two pointers that got replaced for the new
- spy/whisper architecture.)
-
-2008-01-16 20:34 +0000 [r98966-98973] Joshua Colp <jcolp@digium.com>
-
- * .cleancount: Bump up cleancount due to previous commit that
- changed the channel structure.
-
- * apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c,
- main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h
- (added), main/Makefile, include/asterisk/chanspy.h (removed),
- include/asterisk/channel.h, main/audiohook.c (added): Replace
- current spy architecture with backport of audiohooks. This should
- take care of current known spy issues.
-
- * channels/chan_iax2.c: Add missing NULLs at end of two
- ast_load_realtimes. (closes issue #11769) Reported by: tequ
- Patches: chaniax.patch uploaded by dimas (license 88)
-
-2008-01-16 17:20 +0000 [r98964] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c: Fix a deadlock in chan_local in
- local_hangup. There was contention because the local_pvt was held
- and it was attempting to lock a channel, which is the incorrect
- locking order. (closes issue #11730) Reported by: UDI-Doug
- Patches: 11730.patch uploaded by putnopvut (license 60) Tested
- by: UDI-Doug
-
-2008-01-16 15:08 +0000 [r98951-98960] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c: Introduce a lock into the dialing API that protects
- it when destroying the structure. (closes issue #11687) Reported
- by: callguy Patches: 11687.diff uploaded by file (license 11)
-
- * main/rtp.c: Add two more SDP names for ulaw and alaw. (closes
- issue #11777) Reported by: tootai
-
- * channels/chan_sip.c: Don't drop the old record route information
- when dealing with packets related to a reinvite. (closes issue
- #11545) Reported by: kebl0155 Patches: reinvite-patch.txt
- uploaded by kebl0155 (license 356)
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
- configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later
- versions use a separate library for some things so we need to use
- it if present in codec_speex. (closes issue #11693) Reported by:
- yzg
-
-2008-01-15 23:50 +0000 [r98943-98946] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Change a buffer in check_auth() to be a
- thread local dynamically allocated buffer, instead of a massive
- buffer on the stack. This fixes a crash reported by Qwell due to
- running out of stack space when building with LOW_MEMORY defined.
- On a very related note, the usage of BUFSIZ in various places in
- chan_sip is arbitrary and careless. BUFSIZ is a system specific
- define. On my machine, it is 8192, but by definition (according
- to google) could be as small as 256. So, this buffer in
- check_auth was 16 kB. We don't even support SIP messages larger
- than 4 kB! Further usage of this define should be avoided, unless
- it is used in the proper context.
-
- * main/rtp.c, include/asterisk/translate.h, main/frame.c,
- main/translate.c, main/abstract_jb.c, channels/chan_iax2.c,
- codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for
- some memory access errors pointed out by the valgrind2.txt output
- on issue #11698. The issue here is that it is possible for an
- instance of a translator to get destroyed while the frame
- allocated as a part of the translator is still being processed.
- Specifically, this is possible anywhere between a call to
- ast_read() and ast_frame_free(), which is _a lot_ of places in
- the code. The reason this happens is that the channel might get
- masqueraded during this time. During a masquerade, existing
- translation paths get destroyed. So, this patch fixes the issue
- in an API and ABI compatible way. (This one is for you,
- paravoid!) It changes an int in ast_frame to be used as flag
- bits. The 1 bit is still used to indicate that the frame contains
- timing information. Also, a second flag has been added to
- indicate that the frame came from a translator. When a frame with
- this flag gets released and has this flag, a function is called
- in translate.c to let it know that this frame is doing being
- processed. At this point, the flag gets cleared. Also, if the
- translator was requested to be destroyed while its internal frame
- still had this flag set, its destruction has been deffered until
- it finds out that the frame is no longer being processed.
- Admittedly, this feels like a hack. But, it does fix the issue,
- and I was not able to think of a better solution ...
-
-2008-01-15 20:08 +0000 [r98894-98934] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Based on the boundary found move over the
- correct amount. (closes issue #11750) Reported by: tasker
-
- * channels/chan_sip.c: Accept "; boundary=" not just ";boundary="
- in the multipart mixed content type. (closes issue #11750)
- Reported by: tasker
-
-2008-01-14 20:59 +0000 [r98849] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Adding in appropriate unlocks for the locks
- I added. Thanks to joetester on IRC for pointing this out.
-
-2008-01-14 17:38 +0000 [r98774] Russell Bryant <russell@digium.com>
-
- * main/translate.c: Revert a change that introduces an unacceptable
- performance hit and is causing memory leaks ... (from rev 97973)
-
-2008-01-14 16:35 +0000 [r98733-98737] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing another compilation error. I'm a bit off
- today :(
-
- * apps/app_queue.c: Oops. Last commit had compilation error.
-
- * apps/app_queue.c: Adding explicit defaults for missing options to
- init_queue. This is necessary because if a user either removes or
- comments one of these options and reloads their queues, the
- option will not reset to its default, instead maintaining the
- value from prior to the reload. Thanks to John Bigelow for
- pointing this error out to me.
-
-2008-01-12 00:05 +0000 [r98467] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c: Add a connection timeout attribute, as that was
- what was intended with the login timeout, but ODBC divides it up
- into 2 different timeouts. (Closes issue #11745)
-
-2008-01-11 22:46 +0000 [r98390] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems.
- (closes issue #11646) Reported by: caio1982 Patches:
- dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
- dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
- by: caio1982, mvanbaak
-
-2008-01-11 21:28 +0000 [r98372] Pari Nannapaneni <paripurnachand@digium.com>
-
- * main/http.c: Comment explaining how to force browser to always
- read some html files from server.
-
-2008-01-11 19:51 +0000 [r98317-98325] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: If the incoming RTP stream changes codec force the
- bridge to break if the other side does not support it. (closes
- issue #11729) Reported by: tsearle Patches:
- new_codec_patch_udiff.patch uploaded by tsearle (license 373)
-
- * res/res_agi.c: If the channel is hungup during RECORD FILE send a
- result code of -1 to be uniform with everything else. (closes
- issue #11743) Reported by: davevg Patches: res_agi.diff uploaded
- by davevg (license 209)
-
-2008-01-11 19:10 +0000 [r98315] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: Properly report the hangup cause as no answer
- when someone does not answer (closes issue #10574, reported by
- boch, patched by moy)
-
-2008-01-11 18:25 +0000 [r98266] Tilghman Lesher <tlesher@digium.com>
-
- * codecs/gsm/Makefile: Add another exception (which doesn't work)
- for -march optimization flag. Reported by: thomasmebes Patch by:
- tilghman (Closes issue #11563)
-
-2008-01-11 18:25 +0000 [r98265] Russell Bryant <russell@digium.com>
-
- * doc/security.txt, main/asterisk.c, configure,
- include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
- makeopts.in: Backport the ability to set the ToS bits on Linux
- when not running as root. Normally, we would not backport
- features into 1.4, but, I was convinced by the justification
- supplied by the supplier of this patch. He pointed out that this
- patch removes a requirement for running as root, thus reducing
- the potential impacts of security issues. (closes issue #11742)
- Reported by: paravoid Patches: libcap.diff uploaded by paravoid
- (license 200)
-
-2008-01-11 17:22 +0000 [r98219] Joshua Colp <jcolp@digium.com>
-
- * apps/app_followme.c: Ensure the return value of ast_bridge_call
- is passed back up as the application return value. This is needed
- for transfers to function so the PBX core knows to continue
- execution. (closes issue #10327) Reported by: kkiely
-
-2008-01-11 15:52 +0000 [r98164] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Back out changes from revision 97077, since
- it wasn't perfect
-
-2008-01-11 03:39 +0000 [r97976-98082] Russell Bryant <russell@digium.com>
-
- * main/frame.c: Fix samples vs. length calculations for g722
-
- * main/translate.c: Simplify this code with a suggestion from Luigi
- on the asterisk-dev list. Instead of using is16kHz(), implement a
- format_rate() function.
-
- * main/translate.c: Fix various timing calculations that made
- assumptions that the audio being processed was at a sample rate
- of 8 kHz.
-
-2008-01-10 23:08 +0000 [r97973] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c, main/translate.c: 1) When we get a
- translated frame out, clone it, because if the translator pvt is
- freed before we use the frame, bad things happen. 2) Getting a
- failure from ast_sched_delete means that the schedule ID is
- currently running. Don't just ignore it. (Closes issue #11698)
-
-2008-01-10 21:57 +0000 [r97925] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Let us leave a voicemail for ourself if we
- have logged into VoiceMailMain and chosen to leave a message.
- (closes issue #11735, reported and patched by jamessan)
-
-2008-01-10 21:37 +0000 [r97849-97889] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the
- same fixes for ael.flex as was done in 97849 for ast_expr2.fl;
- overrode the normally generate yyfree func with our own version
- that checks the pointer for non-null before passing to free().
- Also takes care of a little problem with 2.5.33 and the use of
- the __STDC_VERSION__ macro.
-
- * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a
- fix for 2 things: a problem Terry was having in OSX with null
- pointers, which was my fault, as I probably forgot to run the sed
- script last time I made mods. So, I moved the fix into the flex
- input itself. Then, I found when I used flex 2.5.33, that it was
- using __STDC_VERSION__, and that's not real good; so I added back
- in a DIFFERENT sed script to fix that little mess. Tested
- everything, a couple different ways. Hope I did no harm, at the
- least.
-
-2008-01-10 20:12 +0000 [r97847] Jason Parker <jparker@digium.com>
-
- * include/asterisk/frame.h: Fix a comment that is no longer true.
-
-2008-01-10 16:19 +0000 [r97734-97753] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample,
- pbx/kdeconsole_main.cc (removed): Remove other remnants of
- pbx_kdeconsole
-
- * pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in,
- configure, include/asterisk/autoconfig.h.in, configure.ac,
- makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't
- worked in ages, and nobody has complained. (closes issue #11706,
- reported by caio1982)
-
-2008-01-10 15:07 +0000 [r97697] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_groupcount.c: Don't try to copy the category from the
- group if no category exists. (closes issue #11724) Reported by:
- IgorG Patches: group_count.v1.patch uploaded by IgorG (license
- 20)
-
-2008-01-09 23:01 +0000 [r97640-97645] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose
- messages
-
- * pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't
- actually load on my system still (related to issue #11706)
-
-2008-01-09 20:28 +0000 [r97618-97622] Jason Parker <jparker@digium.com>
-
- * main/cli.c: Correctly display a message if a command could not be
- found. Also fix a comment which may have led to this happening.
- Issue 11718, reported by kshumard.
-
- * main/cli.c: Fix some locking and return value funkiness. We
- really shouldn't be unlocking this lock inside of a function,
- unless we locked it there too.
-
-2008-01-09 18:48 +0000 [r97575] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some
- smaller functions this time
-
-2008-01-09 18:02 +0000 [r97529] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Fix saying the parking space number to the
- caller doing the parking ...
-
-2008-01-09 17:21 +0000 [r97491] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: report the same message whether Zaptel does
- not have transcoder support loaded or no transcoders were found
-
-2008-01-09 16:44 +0000 [r97489] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Set the caller id within the gtalk_alloc
- function. As underlined in issue #10437 by Josh, we need to
- prevent a possible memory leak. We only set the name part of the
- caller id, the number part is not relevant when dealing with
- JIDs. Closes issue #11549.
-
-2008-01-09 16:11 +0000 [r97450] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Don't do conferencing totally in Zaptel if
- Monitor is running on the channel. (closes issue #11709) Reported
- by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy
- (license 371)
-
-2008-01-09 15:43 +0000 [r97410-97448] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: pass the right variable to get an error
- string... oops
-
- * channels/chan_zap.c: add error number output to ioctl failure
- messages to help with debugging
-
-2008-01-09 00:44 +0000 [r97350] Tilghman Lesher <tlesher@digium.com>
-
- * main/cli.c, main/editline/readline.c: Allow filename completion
- on zero-length modules, remove a memory leak, remove a file
- descriptor leak, and make filename completion thread-safe.
- Patched and tested by tilghman. (Closes issue #11681)
-
-2008-01-09 00:17 +0000 [r97206-97308] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: use the \retval doxygen command properly
-
- * apps/app_queue.c: Part 1 of N of adding doxygen comments to
- app_queue. I picked some of the most common functions used (which
- also happen to be some the biggest/ugliest functions too) to
- document first. I'm pretty new to doxygen so criticism is
- welcome.
-
- * apps/app_queue.c: Some coding guidelines-related cleanup
-
-2008-01-08 20:48 +0000 [r97195] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp.
- (closes issue #11443) Reported by: eferro Patches:
- dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
- 337)
-
-2008-01-08 20:47 +0000 [r97194] Tilghman Lesher <tlesher@digium.com>
-
- * main/autoservice.c, main/utils.c: Increase constants to where
- we're less likely to hit them while debugging. (Closes issue
- #11694)
-
-2008-01-08 20:42 +0000 [r97192] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Making some changes designed to not allow
- for a corrupted mailstream for a vm_state. 1. Add locking to the
- vm_state retrieval functions so that no linked list corruption
- occurs. 2. Make sure to always grab the persistent vm_state when
- mailstream access is necessary. 3. Correct an incorrect return
- value in the init_mailstream function. (closes issue #11304,
- reported by dwhite)
-
-2008-01-08 19:53 +0000 [r97093-97152] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_groupcount.c: If no group has been provided to the
- GROUP_COUNT dialplan function then use the first one specific to
- the channel. (closes issue #11077) Reported by: m4him
-
- * apps/app_queue.c: Make app_queue calls work with directed pickup.
- (closes issue #11700) Reported by: jbauer
-
-2008-01-08 18:02 +0000 [r97077] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes,
- found in issue #11386, but not completely closing that issue.
-
-2008-01-07 20:47 +0000 [r96884-96932] Russell Bryant <russell@digium.com>
-
- * configs/extensions.conf.sample, /: Merged revisions 96931 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) |
- 2 lines Change misery.digium.com to pbx.digium.com ........
-
- * res/res_smdi.c: Don't crash if something happens when setting up
- an SMDI interface and it gets destroyed before the SMDI port
- handling thread gets created.
-
-2008-01-07 14:34 +0000 [r96797-96815] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Indentation fix, makes the code easier to read
-
- * res/res_jabber.c: Compute the base64 value over the
- [authzid]\0authcid\0password string, thus excluding the trailing
- NULL byte. This change has already been committed to trunk, see
- #11644.
-
-2008-01-05 02:09 +0000 [r96644] Russell Bryant <russell@digium.com>
-
- * main/devicestate.c: Don't pass an empty string as the device
- name.
-
-2008-01-04 23:03 +0000 [r96575] Tilghman Lesher <tlesher@digium.com>
-
- * main/devicestate.c: Fix the problem of notification of a device
- state change to a device with a '-' in the name. Could probably
- do with a better fix in trunk, but this bug has been open way too
- long without a better solution. Reported by: stevedavies Patch
- by: tilghman (Closes issue #9668)
-
-2008-01-04 22:55 +0000 [r96573] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Properly continue in the dialplan if using
- PARKINGEXTEN and the slot is full. Issue 11237, patch by me.
-
-2008-01-04 19:27 +0000 [r96525] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: If you change the bindaddr in sip.conf to a
- non-bound address and reload, sip goes kablooie. Reported and
- patched by: one47 (Closes issue #11535)
-
-2008-01-04 16:19 +0000 [r96394-96449] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Make use of the temporary channel pointer
- while the pvt is unlocked. (closes issue #11675) Reported by:
- flefoll Patches: chan_zap.c.patch-store-owner-before-unlock
- uploaded by flefoll (license 244)
-
- * channels/chan_iax2.c: Don't crash if the iax2 pvt structure has
- been destroyed before we get to this point (closes issue #11672,
- reported by snuffy, patched by me)
-
-2008-01-03 21:37 +0000 [r96318] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c: Missed initialization caused crash.
- Reported and fixed by: tiziano (Closes issue #11671)
-
-2008-01-03 12:12 +0000 [r96198-96199] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: make sure frame is completely clean,
- before we send it to asterisk as DTMF. If we don't make it clean,
- it happens that one way audio occurs..
-
- * channels/chan_misdn.c: when overlapdial was used and no number
- was dialed, the call was dropped, now we just jump into the s
- extension, which makes a lot more sense.
-
-2008-01-02 23:46 +0000 [r96102] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: We need to reset the membername to NULL on each
- iteration of this loop, otherwise the result is that multiple
- members can have the same name, since the variable was not reset
- on each iteration of the loop.
-
-2008-01-02 22:14 +0000 [r96020-96024] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_config.c: Convert locks of the contexts list in
- pbx_config to the appropriate rdlock or wrlock
-
- * pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts
- list.
-
- * apps/app_macro.c: app_macro only needs a rdlock on the contexts
- list.
-
-2008-01-02 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.17 released.
-
-2008-01-02 20:24 +0000 [r95946] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Allocate a SIP refer structure when
- performing a transfer using BYE with Also so that the transfer
- information is properly stored. (AST-2008-028) (closes issue
- #11637) Reported by: greyvoip
-
-2008-01-02 17:51 +0000 [r95890] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: A change to improve the accuracy of queue
- logging in the case where a member does not answer during the
- specified timeout period. Prior to this change, there was a small
- chance that the member name recorded in this case would be blank.
- Also prior to this change, if using the ringall strategy, if no
- one answered the call during the specified timeout, the member
- name listed in the queue log would randomly be one of the members
- that was rung. (closes issue #11498, reported and tested by
- hloubser, patched by me)
-
-2007-12-31 23:43 +0000 [r95577] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Avoiding a potentially bad locking situation.
- ast_merge_contexts_and_delete writelocks the conlock, then calls
- ast_hint_extension, which attempts to readlock the same lock.
- Recursion with read-write locks is dangerous, so the inner lock
- needs to be removed. I did this by copying the "guts" of
- ast_hint_extension into ast_merge_contexts_and_delete (sans the
- extra lock). (this change is inspired by the locking problems
- seen in issue #11080, but I have no idea if this is the
- problematic area experienced by the reporters of that issue)
-
-2007-12-31 20:27 +0000 [r95470] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_env.c: Allow the default "0" to be returned if the
- STAT fails (Closes issue #11659)
-
-2007-12-28 18:24 +0000 [r95191] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Remove duplicate increment of the header
- count in the add_header() function. (closes issue #11648)
- Reported by: makoto Patch provided by sergee, committed patch by
- me, inspired by comments from putnopvut
-
-2007-12-28 00:16 +0000 [r95095] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: I found a bug while browsing the queue code and
- managed to reproduce it in a small setup. If a queue uses the
- ringall strategy, it was possible through unfortunate coincidence
- for a single member at a given penalty level to make app_queue
- think that all members at that penalty level were unavailable and
- cause the members at the next penalty level to be rung. With this
- patch, we will only move to the next penalty level if ALL the
- members at a given penalty level are unreachable.
-
-2007-12-27 21:40 +0000 [r95024] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Don't report a syntax error when an empty string
- is passed to ast_get_group. Just return 0. (closes issue #11540)
- Reported by: tzafrir Patches: group_empty.diff uploaded by
- tzafrir (license 46) -- slightly changed by me
-
-2007-12-27 20:09 +0000 [r94977] Mark Michelson <mmichelson@digium.com>
-
- * main/io.c: Fixing a typo in a comment.
-
-2007-12-27 17:32 +0000 [r94905-94924] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_h323.c: Include types.h in chan_h323 as without it
- it can not be compiled on some operating systems like FreeBSD to
- name one. (closes issue #11585) Reported by: sobomax Patches:
- chan_h323.c.diff uploaded by sobomax (license 359)
-
- * channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is
- set or not on the dialog. It is possible for it to be a pointer
- to NULL. (closes issue #11557) Reported by: FuriousGeorge
-
-2007-12-27 15:16 +0000 [r94828-94831] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Now that the contexts lock is a read/write lock, it
- should not be locked here in ast_hint_state_changed(). This makes
- it get locked recursively which now causes a deadlock. (closes
- issue #11080, thanks to callguy for the access to a deadlocked
- machine)
-
- * include/asterisk/translate.h, main/translate.c: Use the constant
- that I really meant to use here ...
-
- * main/translate.c: Change ast_translator_best_choice() to only pay
- attention to audio formats. This fixes a problem where Asterisk
- claims that a translation path can not be found for channels
- involving video. (closes issue #11638) Reported by: cwhuang
- Tested by: cwhuang Patch suggested by cwhuang, with some
- additional changes by me.
-
-2007-12-27 01:01 +0000 [r94824] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c: make this comment explain the situation in an
- even more explicit fashion
-
-2007-12-26 20:43 +0000 [r94808] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c: Workaround for what is probably a glibc bug (but
- we'll see this crop up again and again, if we don't add the
- workaround). Reported by: rolek Patch by: tilghman (Closes issue
- #11601, closes issue #11426)
-
-2007-12-26 19:04 +0000 [r94789-94801] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c: Just in case the AST_FLAG_END_DTMF_ONLY flag
- was already set before starting autoservice, remember it and
- ensure that the channel has the same setting when autoservice
- gets stopped. (pointed out by d1mas, patched up by me)
-
- * main/autoservice.c: When a channel is in autoservice, mark a flag
- on the channel that says that we only care about the END of a
- digit. That way, no magic digit emulation stuff will happen when
- all we're doing is queueing up END frames.
-
- * res/res_features.c: Don't try to send a parked call back to
- itself. (closes issue #11622, reported by djrodman, patched by
- me)
-
- * main/autoservice.c: Don't store DTMF BEGIN frames while a channel
- is in autoservice. It's just going to make ast_read() do a lot of
- extra work when the channel comes back out of autoservice.
- (closes issue #11628, patched by me)
-
- * Makefile: List include/asterisk/version.h as a .PHONY target
- because we want the commands listed for this target to be
- executed regardless of whether the file exists or not. This fixes
- having the version not up to date when running from svn. (closes
- issue #11619, reported by plack, fixed by me)
-
-2007-12-25 02:27 +0000 [r94769] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: file says... build on the builders.
-
-2007-12-24 19:36 +0000 [r94763-94767] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c: Race: we need to wait to queue a NewChannel event
- until after the channel is inserted into the channel list. The
- reason is because some manager users immediately queue requests
- from the channel when they see that event and are confused when
- Asterisk reports no such channel. (Closes issue #11632)
-
- * channels/chan_sip.c: More deadlock avoidance code (this time
- between sip_monitor and sip_hangup) Reported by: apsaras Patch
- by: tilghman (Closes issue #11413)
-
- * channels/chan_sip.c: Another bit of bad logic in realtime_peer
- Reported by: dimas Patch by: dimas (Closes issue #11631)
-
-2007-12-23 01:21 +0000 [r94660] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Argh... I suppose third time's the charm.
-
-2007-12-21 20:21 +0000 [r94468-94543] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Bunch of coding guidelines cleanup
-
- * apps/app_voicemail.c: Better quota support for using IMAP storage
- voicemail (closes issue #11415, reported by jaroth) (closes issue
- #11152, reported by selsky) Patch provided by jaroth
-
- * apps/app_voicemail.c: The mail_copy c-client function does not
- expect a full imap mailbox string, just the name of the mailbox.
- (closes issue #11419, reported and patched by jaroth, with
- additional patchwork from me)
-
- * main/dial.c: Since we are freeing list elements within a list
- traversal, we need to use the safe traversal and remove the item
- from the list before freeing it. (closes issue 11612, reported by
- dtyoo)
-
-2007-12-21 16:37 +0000 [r94466] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, include/asterisk/pbx.h: Convert the contexts lock to
- a read/write lock to resolve a deadlock. This has a nice side
- benefit of improving performance. :) (closes issue #11609)
- (closes issue #11080)
-
-2007-12-21 16:11 +0000 [r94420-94464] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Removing a debug message I accidentally just
- committed
-
- * main/say.c, apps/app_queue.c: Fixing Portuguese syntax for saying
- dates and times. Also some coding guidelines cleanup. (closes
- issue #11599, reported and patched by caio1982, coding guidelines
- cleanup by me)
-
-2007-12-21 15:07 +0000 [r94418] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: Fix for restart-as-user problem reported via the
- -dev list
-
-2007-12-20 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.16.2 released.
-
-2007-12-20 20:22 +0000 [r94215-94256] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 94255 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) |
- 5 lines Fix another potential seg fault ... (closes issue #11606)
- Reported by: dimas ........
-
- * channels/chan_zap.c: Fix a deadlock in d-channel handling in
- chan_zap. This deadlock was introduced by the fix to ensure that
- channels are properly locked when handling channel variables.
- There were sections of this code where the channel pvt was locked
- before the channel lock, when in fact it _must_ be the other way
- around. (closes issue #11582) Reported by: bugi
-
-2007-12-19 23:02 +0000 [r94122] Mark Michelson <mmichelson@digium.com>
-
- * res/res_monitor.c: Sox versions 13.0.0 and newer do not have
- "soxmix" and instead use sox -m. res_monitor needs to use this if
- the user does not have soxmix. (closes issue #11589, reported by
- amessina, patch inspired by amessina but with a flourish from me)
-
-2007-12-19 22:48 +0000 [r94077] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: Check
- for the existence of the soxmix application on the target
- platform and have the result available in autoconfig.h. (part of
- issue #11589)
-
-2007-12-19 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.16.1 released.
-
-2007-12-19 17:29 +0000 [r93955] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Make the 1.4 builders happy, ensure var is
- NULL.
-
-2007-12-19 17:04 +0000 [r93949] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Avoid segfault in chan_iax when peer isn't
- defined (Closes issue #11602)
-
-2007-12-18 22:42 +0000 [r93764] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: FreeBSD also does not have byte swap
- functions. Issue 11586, patch by sobomax.
-
-2007-12-18 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.16 released.
-
-2007-12-18 18:45 +0000 [r93668-93676] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
- 93667 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007)
- | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........
-
-2007-12-18 17:02 +0000 [r93625] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: Rework deadlock avoidance used in ast_write,
- since it meant that agent channels which were being monitored had
- one audio file recorded and one empty audio file saved. (closes
- issue #11529, reported by atis patched by me)
-
-2007-12-17 22:56 +0000 [r93381-93420] Jason Parker <jparker@digium.com>
-
- * main/translate.c: What was I thinking when I wrote this
- masterpiece? -1 + 1 = 0.. who woulda thunk it?.
-
-2007-12-17 22:28 +0000 [r93377] Joshua Colp <jcolp@digium.com>
-
- * main/utils.c: Do not try to access information about a lock when
- printing out a trylock attempt. It is possible for the lock that
- it references to no longer be valid. This would have caused
- segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
- Reported by: callguy (closes issue #11100) Reported by: callguy
-
-2007-12-17 21:12 +0000 [r93336] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/time.h: Today is tomorrow's yesterday, and
- yesterday's tomorrow is today, and tomorrow's tomorrow is the day
- after tomorrow, so who cares if you recycle anyway? If this
- confuses you, that's nothing compared to what this fixes. ;-)
-
-2007-12-17 19:53 +0000 [r93291] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: We need to create the directory for a
- voicemail user even if they are using IMAP storage since
- greetings are stored in the filesystem. (closes issue #11388,
- reported by spditner, patch by me inspired by a patch by
- spditner)
-
-2007-12-17 18:05 +0000 [r93250] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c: If a call is received with a called number
- IE containing nothing go to the 's' extension. (closes issue
- #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt
- uploaded by Corydon76 (license 14)
-
-2007-12-17 07:21 +0000 [r93183] Kevin P. Fleming <kpfleming@digium.com>
-
- * funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile,
- res/Makefile, channels/Makefile, formats/Makefile: fix some
- copy-and-paste leftovers
-
-2007-12-17 07:15 +0000 [r93182] Olle Johansson <oej@edvina.net>
-
- * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
- apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add
- dependencies on res_monitor and res_features. I wonder if
- Asterisk can run at all without res_features. My guess is that
- there's propably a lot of more modules and the core that depends
- on it. Reported by: caio1982 (closes issue #11574)
-
-2007-12-17 06:44 +0000 [r93180] Kevin P. Fleming <kpfleming@digium.com>
-
- * formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
- configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in,
- formats/Makefile, pbx, res, channels, funcs/Makefile, codecs,
- include/asterisk/autoconfig.h.in, build_tools/make_version, apps,
- configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps
- (removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In
- http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
- rizzo brought up some issues related to the way that the metadata
- required for menuselect and the rest of the build system is
- extracted from the source files. Since I had a few hours to kill
- on an airplane today, I decided to improve this situation... so
- now the system caches the extracted metadata and uses it to build
- the menuselect 'tree' as much as it can. The result of this is
- that when a single source file is changed, only the metadata for
- that file needs to be extracted again, and the rest is used from
- the cache files. I also reduced the number of forked processes
- required to do the metadata extraction; it was actually possible
- to do most of what we needed in the Makefiles themselves without
- using any shell scripts at all! On my laptop, these changes
- resulted in an 80% decrease in the time required for the
- 'menuselect.makeopts' automatic check to occur after editing a
- single source file. While doing this work I also cleaned up a few
- minor things in the Makefiles, adding a check for 'awk' to the
- configure script and changed all remaining places we use 'grep'
- or 'awk' to use the ones found by the configure script, and
- changed the 'prep_tarball' script to build the menuselect
- metadata so that tarballs of Asterisk will include it and won't
- require the user to wait while it is extracted after unpacking.
-
-2007-12-14 17:36 +0000 [r93000] Russell Bryant <russell@digium.com>
-
- * main/config.c: There are a lot of existing systems that #include
- non-existent files. So, to make the transition to treating this
- as an error a bit less painless, just issue a huge error message
- for now. Then, later, we can reinstate the code that treats it as
- a failure. (Thanks to philippel for the feedback)
-
-2007-12-14 15:16 +0000 [r92937] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Up the length of the format on the SIP
- channel since it can now be rather long. (closes issue #11552)
- Reported by: francesco_r
-
-2007-12-14 15:05 +0000 [r92934] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS
- state setting and overlap_task thread starting.
-
-2007-12-14 15:01 +0000 [r92933] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Change help documentation to match actual behavior
- (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
- (Closes issue #11548)
-
-2007-12-14 01:24 +0000 [r92875] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS,
- don't spam the CLI with messages about possible deadlocks.
- Instead just print the intended single message every five
- seconds. (closes issue 11537, reported and patched by dimas)
-
-2007-12-13 21:28 +0000 [r92815] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: Properly initialize polarity statuses, so
- that they are detected properly. Reported by: julianjm Patch by:
- julianjm (Closes issue #10238)
-
-2007-12-13 20:13 +0000 [r92809] Jason Parker <jparker@digium.com>
-
- * main/pbx.c: Make application help text a little more clear about
- the use of extensions in a filename.
-
-2007-12-13 20:03 +0000 [r92803-92807] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Prevent another potential fd leak
-
- * apps/app_voicemail.c: Prevent a possible fd leak.
-
-2007-12-13 00:11 +0000 [r92696] Jason Parker <jparker@digium.com>
-
- * main/config.c, channels/chan_sip.c, channels/chan_h323.c,
- channels/chan_iax2.c: If a typo is found in a config file, we
- previous continued on with what was already loaded. We do not
- want to do this (see bug below for details). This makes it so
- that if a [ is found without a ], the entire config will fail,
- and nothing in it will be loaded. Isue #10690.
-
-2007-12-12 22:00 +0000 [r92656] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: emit a warning message when we drop a G.729B
- CNG frame destined for the transcoder
-
-2007-12-12 21:15 +0000 [r92617] Jason Parker <jparker@digium.com>
-
- * apps/app_meetme.c: Don't increment user count until after name
- has been recorded (if enabled). Issue 11048, tested by pep.
-
-2007-12-12 19:40 +0000 [r92556] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: resolve compiler warning
-
-2007-12-12 17:46 +0000 [r92510] Mark Michelson <mmichelson@digium.com>
-
- * res/res_features.c: Correctly detect where a dynamic feature was
- activated. Before this patch, the channel which initiated the
- bridge was always assumed to have been the one which activated
- the dynamic feature. This patch corrects this. (closes issue
- #11529, reported and patched by nic_bellamy)
-
-2007-12-12 16:52 +0000 [r92463] Tilghman Lesher <tlesher@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: Test
- directly for the API that fixed AST-2007-026, to ensure that
- older versions of PostgreSQL are no longer acceptable. (Closes
- issue #11526)
-
-2007-12-12 16:08 +0000 [r92443] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Removing an unused variable.
-
-2007-12-11 19:51 +0000 [r92363] Joshua Colp <jcolp@digium.com>
-
- * main/global_datastores.c: Fix potential memory leak with the
- dialed interfaces list if another memory allocation fails.
- (closes issue #11507) Reported by: eliel Patches:
- global_datastores.c.patch uploaded by eliel (license 64)
-
-2007-12-11 17:42 +0000 [r92323] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing autofill to be more accurate.
- Specifically, if calls ahead of the current caller were ringing
- members (but not yet bridged) there could be available members
- and waiting callers who would not get matched up. The member
- availability checker was correctly determining the number of
- available members in this scenario, but the queue itself did not
- parallelly reflect this status on the pending calls. This commit
- corrects the issue. (closes issue #11459, reported by
- equissoftware, patched by me)
-
-2007-12-10 16:36 +0000 [r92204] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Add G729A as another possible payload name for G729.
- Some devices use this instead of G729, which is perfectly normal
- since the payload number itself is defined and can't be used by
- anything else so the name doesn't matter that much. (closes issue
- #11483) Reported by: revolution Patches: rtp.diff uploaded by
- revolution (license 346)
-
-2007-12-10 16:29 +0000 [r92202] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: If there are no members in a queue, then the
- loop where the datastore for detecting duplicate dialed numbers
- will be skipped, meaning the datastore isn't created. This means
- that when we try to free it, there's a crash. This stops that
- crash from occurring. (closes issue #11499, reported by slavon,
- patched by eliel)
-
-2007-12-10 16:13 +0000 [r92200] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: It is possible for nativeformats to contain
- more then one codec, so print out multiple ones. (closes issue
- #11366) Reported by: ovi
-
-2007-12-10 14:04 +0000 [r92158] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Avoid reinvite race situations with two
- Asterisks trying to reinvite each other in 1.4 and trunk. This
- patch implements support for the 491 error code that Asterisk 1.4
- generates on situations where we get an incoming INVITE and
- already has one in progress. Thanks to mavetju for reporting and
- to Raj Jain for an excellent explanation of the problem. Patch by
- myself. Tested with 8 Asterisk servers connected to each other in
- a training network. Closes issue #10481
-
-2007-12-07 23:29 +0000 [r91890] Jason Parker <jparker@digium.com>
-
- * main/dsp.c: We need to make sure we free the input frame if we
- return a different frame in ast_dsp_process. Issue 11273, pointed
- out by dimas, with a patch by eliel.
-
-2007-12-07 22:30 +0000 [r91870] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: even though Asterisk explicitly requests that
- endpoints using G.729 do *not* use Annex B (silence detection and
- comfort noise generation) some do anyway; the transcoder card
- interface does not currently work properly with CNG frames, so
- trim off the CNG before sending the data
-
-2007-12-07 21:24 +0000 [r91777-91830] Russell Bryant <russell@digium.com>
-
- * main/utils.c: Make the lock protecting each thread's list of
- locks it currently holds recursive. I think that this will fix
- the situation where some people have said that "core show locks"
- locks up the CLI. (related to issue #11080)
-
- * include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS
- code. The ast_mutex_init() function had the mutex attribute
- object marked as static. This means that multiple threads
- initializing locks at the same time could step on each other and
- end up with improperly initialized locks. (found when tracking
- down locking issues related to issue #11080)
-
- * include/asterisk/lock.h: I love fixing lock related errors in the
- lock debugging code. That's about as ironic as it gets in
- Asterisk programming land. Anyway, I spotted this bug while
- trying to track down why systems are locking up and acting weird
- in issue #11080. The mutex attribute object was marked as static
- in this function when it should not have been.
-
- * apps/app_dial.c: * Add channel locking around datastore
- operations that expect the channel to be locked. * Document why
- we don't record Local channels in the dialed interfaces list. *
- Remove the dialed variable as it isn't needed. * Restructure some
- code for clarity and coding guidelines stuff
-
- * apps/app_queue.c: * Add channel locking around datastore
- operations that expect the channel to be locked. * Document why
- we don't record Local channels in the dialed interfaces list. *
- Handle memory allocation failure. * Remove the dialed variable,
- as it wasn't actually needed. * Tweak some formatting to conform
- to coding guidelines.
-
- * main/autoservice.c: * Add a bit more of a verbose comment as to
- why a hangup frame needs to be queued up if autoservice gets a
- NULL return from ast_read(). * Make the process of queueing the
- hangup frame more efficient by putting the frame where it is
- going to end up and avoiding some locking and extra memory
- allocations and freeing.
-
-2007-12-07 15:39 +0000 [r91737] Mark Michelson <mmichelson@digium.com>
-
- * main/autoservice.c: Hangups that happen during autoservice were
- not processed appropriately. This is because a hangup actually
- causes a NULL frame to be received, not a hangup frame. Queueing
- a hangup if we receive a NULL frame during autoservice corrects
- this problem (closes issue #11467, reported by jmls, patched by
- me)
-
-2007-12-07 02:51 +0000 [r91675-91693] Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c: Don't unlock the dialed_interfaces list until
- we're done messing with the iterator.
-
- * apps/app_dial.c, apps/app_queue.c: Allow dialing local channels
- from Queue() and Dial() again. There was a slight flaw in the
- code to prevent call forwards from looping that caused this
- problem. (related to issue #11486)
-
- * apps/app_queue.c: Fix in an issue in the call forwarding handling
- code that was causing crashes on every call into a queue. I'm not
- entirely sure about the logic in this part of the code, so I want
- to look at it some more tomorrow. However, this makes it safe and
- keeps it from crashing. (closes issue #11486, reported by adamg,
- patched by me)
-
-2007-12-07 00:52 +0000 [r91637] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c: At the end of a call, when we're reporting, RTCP may
- already be partially torn down, so check for NULL dereference
- Reported by: blitzrage Patch by: tilghman (Closes issue #11450)
-
-2007-12-06 20:25 +0000 [r91541] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: IMAP storage did not honor the maxmsg
- setting in voicemail.conf, and it also had the possibility of
- crashing if a user had more than 256 messages in their voicemail.
- This patch kills two birds with one stone by adding maxmsg
- support and also setting a hard limit on the number of messages
- at 255 so that the crashes cannot happen. (closes issue #11101,
- reported by Skavin, patched by me)
-
-2007-12-06 19:11 +0000 [r91501] Russell Bryant <russell@digium.com>
-
- * main/loader.c, include/asterisk/module.h: Add a new module flag
- to indicate that a build sum is present. Modules built against
- older Asterisk 1.4 headers will now load properly with just a
- warning indicating that they are old and may cause problems.
- (patch by paravoid)
-
-2007-12-06 16:49 +0000 [r91439-91450] Joshua Colp <jcolp@digium.com>
-
- * main/udptl.c: Fix various in the udptl implementation. It could
- return empty modem frames, have an incorrect sequence number on
- packets, and display the wrong sequence number in the debug
- messages. (closes issue #11228) Reported by: Cache Patches:
- udptl-4.patch uploaded by dimas (license 88)
-
- * channels/chan_sip.c: Add support for accepting and sending T.38
- in the initial INVITE. (closes issue #9402) Reported by: thdei
-
-2007-12-06 12:54 +0000 [r91366] Olle Johansson <oej@edvina.net>
-
- * main/loader.c, include/asterisk/logger.h, main/logger.c: Make
- sure logger is reloaded at general reload in the cli. (Discovered
- during Asterisk training in Portugal)
-
-2007-12-05 22:57 +0000 [r91273-91292] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Reverting extra stuff I didn't mean to
- commit
-
- * apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial()
- did not properly handle the case where only a label was provided.
- This was due to the fact that the answering channel did not have
- an extension set, so ast_parseable_goto would fail. This fix
- eliminates the call to ast_parseable_goto on the answering
- channel since it is a wasteful call. The answering channel and
- the calling channel are both directed to the same extension and
- context, just different priorities, so we can just copy the
- values from the calling channel to the answering channel and
- increment the answering channel's priority. (closes issue #11382,
- reported by jon, patch by me with correction by jon)
-
-2007-12-05 21:38 +0000 [r91237] Tilghman Lesher <tlesher@digium.com>
-
- * sounds/Makefile: Upgrade to the latest version of extra sounds
-
-2007-12-05 17:31 +0000 [r90967-91192] Russell Bryant <russell@digium.com>
-
- * main/threadstorage.c: Make the lock in the threadstorage
- debugging code untracked to avoid a deadlock on thread
- destruction. (closes issue #11207) Reported by: ys Patches:
- threadstorage.c.diff uploaded by ys (license 281) Also fixes an
- open bug report: (closes issue #11446)
-
- * main/utils.c: When DEBUG_THREADS is enabled, we only have the
- details about who is holding a lock that we are waiting on for a
- mutex, not rwlocks. This should fix the problem where people have
- reported "core show locks" crashing sometimes.
-
- * include/asterisk/lock.h: Fix some crashes in chan_iax2 that were
- reported as happening on Mac systems. It turns out that the
- problem was the Mac version of the ast_atomic_fetchadd_int()
- function. The Mac atomic add function returns the _new_ value,
- while this function is supposed to return the old value. So, the
- crashes happened on unreferencing objects. If the reference count
- was decreased to 1, ao2_ref() thought that it had been decreased
- to zero, and called the destructor. However, there was still an
- outstanding reference around. (closes issue #11176) (closes issue
- #11289)
-
- * include/asterisk/file.h, configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/compiler.h: Modify file.h to maintain API
- compatibility with earlier versions. If a recent compiler is
- being used, then a warning will show up for any modules still
- using the old name "private" instead of "_private". (patch
- suggested by paravoid)
-
- * main/pbx.c: Make some changes to some additions I made recently
- for doing channel autoservice when looking up extensions. This
- code was added to handle the case where a dialplan switch was in
- use that could block for a long time. However, the way that I
- added it, it did this for all extension lookups. However, lookups
- in the in-memory tree of extensions should _not_ take long enough
- to matter. So, move the autoservice stuff to be only around
- executing a switch.
-
-2007-12-04 17:28 +0000 [r90876] Jason Parker <jparker@digium.com>
-
- * main/channel.c: If we fail to create a channel after allocating a
- timing fd, we need to make sure to close it. Issue 11454, patch
- by eliel.
-
-2007-12-04 05:29 +0000 [r90798] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Fix build issue on the build cluster.
-
-2007-12-03 23:50 +0000 [r90736-90753] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/compat.h: Solaris requires the inclusion of
- sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
- snuffy,tilghman (Closes issue #11430)
-
- * res/res_config_pgsql.c: If both dbhost and dbsock were not set, a
- NULL deref could result Reported by: xrg Patch by: tilghman
- (Closes issue #11387)
-
-2007-12-03 23:12 +0000 [r90735] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, main/channel.c, main/global_datastores.c
- (added), channels/chan_local.c, main/Makefile,
- include/asterisk/channel.h, include/asterisk/global_datastores.h
- (added), apps/app_queue.c: A big one... This is the merge of the
- forward-loop branch. The main change here is that call-forwards
- can no longer loop. This is accomplished by creating a datastore
- on the calling channel which has a linked list of all devices
- dialed. If a forward happens, then the local channel which is
- created inherits the datastore. If, through this progression of
- forwards and datastore inheritance, a device is attempted to be
- dialed a second time, it will simply be skipped and a warning
- message will be printed to the CLI. After the dialing has been
- completed, the datastore is detached from the channel and
- destroyed. This change also introduces some side effects to the
- code which I shall enumerate here: 1. Datastore inheritance has
- been backported from trunk into 1.4 2. A large chunk of code has
- been removed from app_dial. This chunk is the section of code
- which handles the call forward case after the channel has been
- requested but before it has been called. This was removed because
- call-forwarding still works fine without it, it makes the code
- less error-prone should it need changing, and it made this set of
- changes much less painful to just have the forwarding handled in
- one place in each module. 3. Two new files, global_datastores.h
- and .c have been added. These are necessary since the datastore
- which is attached to the channel may be created and attached in
- either app_dial or app_queue, so they need a common place to find
- the datastore info. This approach was taken in case similar
- datastores are needed in the future, there will be a common place
- to add them.
-
-2007-12-03 22:06 +0000 [r90696] Jason Parker <jparker@digium.com>
-
- * apps/app_meetme.c: Make sure we always close the conference fd if
- we have an open one. Issue 11383, reported by markmhy, patch by
- eliel.
-
-2007-12-03 20:59 +0000 [r90639] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_mgcp.c: Changing some bad logic when calculating
- the interdigit timeout. (closes issue #11402, reported and
- patched by eferro)
-
-2007-12-03 20:51 +0000 [r90607] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Fix crash in ParkAndAnnounce application.
- Issue #11436, reported by lytledd, patch by eliel.
-
-2007-12-03 20:05 +0000 [r90548-90588] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Do not create a smoother for G723.1 frames, they need
- to be left alone to their native 20/24 byte size.
-
- * .cleancount, main/channel.c, include/asterisk/channel.h: Preserve
- the indication currently playing on a channel when a masquerade
- operation happens. (issue #BE-88)
-
-2007-12-03 18:20 +0000 [r90546] Jason Parker <jparker@digium.com>
-
- * channels/chan_iax2.c: Only log debug messages if debug is
- enabled. Closes issue #11416, patch by casper.
-
-2007-12-02 18:18 +0000 [r90470] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: The other day when I went through making
- changes as a result of the ao2_link() change, I added some code
- to set pointers to NULL after they were unreferenced. This
- pointed out that in this place, the object was unreferenced
- before the code was done using it. So, move the unref down a
- little bit. (crash reported by jmls on IRC)
-
-2007-12-02 09:34 +0000 [r90432] Tilghman Lesher <tlesher@digium.com>
-
- * main/autoservice.c: Clarify the return value on autoservice.
- Specifically, if you started autoservice and autoservice was
- already on, it would erroneously return an error. Reported by:
- adiemus Patch by: dimas (Closes issue #11433)
-
-2007-11-30 19:26 +0000 [r90310-90348] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c, main/manager.c, include/asterisk/astobj2.h,
- apps/app_queue.c, channels/chan_iax2.c: Change the behavior of
- ao2_link(). Previously, in inherited a reference. Now, it
- automatically increases the reference count to reflect the
- reference that is now held by the container. This was done to be
- more consistent with ao2_unlink(), which automatically releases
- the reference held by the container. It also makes it so it is no
- longer possible for a pointer to be invalid after ao2_link()
- returns.
-
- * include/asterisk/astobj2.h: Add some notes on the behavior of
- ao2_unlink() after a discussion with Tilghman
-
-2007-11-30 14:43 +0000 [r90269] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix locking issues under one legged replaces
- scenarios. (closes issue #11420) Reported by: irroot Patches:
- chan_sip_oneleg.patch uploaded by irroot (license 52)
-
-2007-11-30 00:16 +0000 [r90231] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_mgcp.c: Clear the DTMF buffer if the call times
- out. (closes issue #11418, reported and patched by eferro)
-
-2007-11-29 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.15 released.
-
-2007-11-29 19:48 +0000 [r90166] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_pgsql.c: Properly escape cdr->src and cdr->dst and ensure
- we use thread-safe escaping (Fixes AST-2007-026)
-
-2007-11-29 19:38 +0000 [r90163] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This patch handles the case where a queue
- member with a negative penalty is added via the manager. If a
- negative value is submitted for a member penalty, we set it to 0.
- (closes issue #11411, reported and patched by Laureano)
-
-2007-11-29 19:24 +0000 [r90154-90160] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c: Properly escape input buffers (Fixes
- AST-2007-025)
-
- * formats/format_g726.c, include/asterisk/file.h,
- formats/format_wav.c, formats/format_pcm.c,
- formats/format_ogg_vorbis.c, main/file.c, formats/format_h263.c,
- formats/format_h264.c, formats/format_wav_gsm.c: Use of "private"
- as a field name in a header file messes with C++ projects
- Reported by: chewbacca Patch by: casper (Closes issue #11401)
-
- * sounds/Makefile: Upgrade the core sounds release version
-
-2007-11-29 00:36 +0000 [r90142-90147] Russell Bryant <russell@digium.com>
-
- * funcs/func_callerid.c: fix some formatting i accidentally changed
-
- * funcs/func_callerid.c, main/channel.c,
- include/asterisk/channel.h: This set of changes is to make some
- callerID handling thread-safe. The ast_set_callerid() function
- needed to lock the channel. Also, the handlers for the CALLERID()
- dialplan function needed to lock the channel when reading or
- writing callerid values directly on the channel structure.
-
- * include/asterisk/file.h, main/file.c: Merge a change from
- team/russell/chan_refcount ... This makes ast_stopstream()
- thread-safe.
-
-2007-11-28 22:59 +0000 [r90101] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Fix a few memory leaks. (closes issue #11405)
- Reported by: eliel Patches: load_realtime.patch uploaded by eliel
- (license 64)
-
-2007-11-28 22:30 +0000 [r90098] Kevin P. Fleming <kpfleming@digium.com>
-
- * configs/users.conf.sample, main/manager.c: it is impossible to
- set permissions for manager accounts created by users.conf
- (reported internally, patched by me)
-
-2007-11-28 22:08 +0000 [r89999-90059] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Removing some seemingly pointless code. This sets a
- channel variable for every priority executed in the dialplan if
- you have debug set to anything non-zero. This seems pointless due
- to the fact that these channel variables are not referenced
- anywhere else in the code and their names are esoteric enough
- that they would not be practical to reference in the dialplan.
- Plus the fact that this behavior isn't documented anywhere means
- that the change is not likely to cause any disruption. If
- anything, this may actually cause a slight performance increase
- if running with debug on. The motivating influence for this code
- change is the eventwhencalled option for queues. If set to vars,
- all channel variables will be output to the manager. These
- unnecessary channel variables make the output a lot more
- difficult to deal with.
-
- * apps/app_voicemail.c: Recording greetings when using IMAP storage
- was causing zero-length files to be stored. Since greetings are
- not retrieved from IMAP anyway, it is pointless to attempt
- storing them there. (closes issue #11359, reported by spditner,
- patched by me)
-
-2007-11-28 00:20 +0000 [r89839-89893] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, include/asterisk/pbx.h: - update documentation for
- some of the goto functions to note that they handle locking the
- channel as needed - update ast_explicit_goto() to lock the
- channel as needed
-
- * main/autoservice.c: Don't do frame processing if ast_read()
- returned NULL.
-
- * apps/app_queue.c: Instead of depending on the return value of
- ast_true(), explicitly set the eventwhencalled variable to 1.
-
- * main/pbx.c: Don't start/stop autoservice in
- pbx_extension_helper() unless a channel exists
-
-2007-11-27 23:10 +0000 [r89837] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Two changes with regards to the
- 'eventwhencalled' option of queues.conf 1) Due to some signed vs.
- unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
- did exactly the same thing. Thus the sign change of the ast_true
- call. 2) The vars2manager function overwrote a \n for every
- channel variable it parsed, resulting in bizarre output for the
- channel variables. This patch remedies this. (related to issue
- #11385, however I'm not sure if this will actually be enough to
- close it)
-
-2007-11-27 21:45 +0000 [r89790] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, main/pbx.c: Merge changes from
- team/russell/autoservice_1.4 This set of changes fixes an issue
- that was reported to me on IRC yesterday. The user, d1mas, was
- using chan_zap for incoming calls and was having DTMF recognition
- issues in some situations. Specifically, he noticed that the
- problem occurred when using DISA or WaitExten. He also noticed
- that when using Read, the problem did not occur. His system also
- used DUNDi for dialplan lookups. So, he theorized that if the
- DUNDi lookups blocked for some period of time, that audio from
- the zap channel could get lost. If the audio got lost, then it
- wouldn't be run through the DTMF detector, and digits could get
- lost. He was correct, and the following set of changes fixes the
- problem. However, the changes go a little bit further than what
- was necessary to fix this exact problem. 1) I updated
- pbx_extension_helper() to autoservice the associated channel to
- handle cases where extension lookups may take a long time. This
- would normally be a dialplan switch that does some lookup over
- the network, such as the DUNDi or IAX2 switches. This ensures
- that even while a DUNDi lookup is blocking, the channel will be
- continuously serviced. 2) I made a change to the autoservice
- code. This is actually something that has bothered me for a long
- time. When a channel is in autoservice, _all_ frames get thrown
- away. However, some frames really shouldn't be thrown away. The
- most notable examples are signalling (CONTROL) frames, and DTMF.
- So, this patch queues up important frames while a channel is in
- autoservice. When autoservice is stopped on the channel, the
- queued up frames get stuck back on the channel so that they can
- get processed instead of thrown away. 3) I made another change to
- the autoservice code to handle the case where autoservice is
- started on channels recursively. Previously, you could call
- ast_autoservice_start() multiple times on a channel, and it would
- stop the first time ast_autoservice_stop() gets called. Now, it
- will ensure that autoservice doesn't actually stop until the
- final call to ast_autoservice_stop().
-
-2007-11-27 20:22 +0000 [r89727] Mark Michelson <mmichelson@digium.com>
-
- * res/res_config_pgsql.c: Changing some calls from free() to
- ast_free() since they were allocated with ast_calloc(). (closes
- issue #11390, reported and patched by Laureano)
-
-2007-11-27 20:16 +0000 [r89701-89709] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/app.c: on second thought... revert all the other changes
- i've made in app options parsing leaving only one: if an empty
- argument is supplied for an option, set that argument pointer to
- point to an empty string rather than NULL, so that the
- application can do normal checks on it without worrying about it
- being NULL
-
- * main/app.c: generate a warning when an application option that
- requires an argument is ignored due to lack of an argument
-
-2007-11-27 16:12 +0000 [r89634] Russell Bryant <russell@digium.com>
-
- * configs/voicemail.conf.sample: Add a note to the sample voicemail
- config noting that when using IMAP storage, only the first format
- specified will be attached to the message.
-
-2007-11-27 15:38 +0000 [r89631] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_env.c: Default result of STAT should be "0" not "".
- Reported via the -users mailing list, fixed by me.
-
-2007-11-27 15:23 +0000 [r89624-89630] Olle Johansson <oej@edvina.net>
-
- * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: If we
- get a codec offer using a well-known payload type, but using it
- for another codec that we don't know, Asterisk did not remove
- that codec from the list. With this patch, we remove the codec
- from audio and video rtp objects and deny it ever existed. Thanks
- to lasse for testing. (closes issue #11376) Reported by: lasse
- Patches: bug11376.txt uploaded by oej (license 306) Tested by:
- lasse
-
- * configs/sip.conf.sample: Clarify limitonpeers=yes (closes issue
- #11304) Reported by: pj
-
-2007-11-27 06:24 +0000 [r89622] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, main/cdr.c, configs/cdr.conf.sample,
- include/asterisk/cdr.h: closes issue #11379; OK, this is an
- attempt to make both sides happy. To the cdr.conf file, I added
- the option 'unanswered', which defaults to 'no'. In this mode,
- you will see a cdr for a call, whether it was answered or not.
- The disposition will be NO ANSWER or ANSWERED, as appropriate.
- The src is as you'd expect, the destination channel will be one
- of the channels from the Dial() call, usually the last in the
- list if more than one chan was specified. With unanswered set to
- 'yes', you will still see this cdr entry in both cases. But in
- the case where the dial timed out, you will also see a cdr for
- each line attempted, marked NO ANSWER, with no destination
- channel name. The new option defaults to 'no', so you don't see
- the pesky extra cdr's by default, and you will not see the
- irritating 'not posted' messages.
-
-2007-11-26 23:10 +0000 [r89616-89618] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_playback.c: After issuing a "say load new", if a caller
- hangs up during the middle of playback of a number, app_playback
- will continue to try to play the remaining files. With this
- change, no more files will be played back upon hangup. (closes
- issue #11345, reported and patched by IgorG)
-
- * apps/app_playback.c: After issuing a "say load new" tons of
- warning messages are printed out to the CLI every time do_say in
- app_playback is called. Removing these warnings
-
-2007-11-26 21:10 +0000 [r89599-89610] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c: Fix issues with async dialing with an application
- executing. The application has to be terminated and control
- returned to the thread before hanging things up. (issue #BE-252)
-
- * res/res_features.c: Add module counting removal for error
- conditions. (closes issue #11333) Reported by: Laureano Patches:
- res_features_v2.c.patch uploaded by Laureano (license 265)
-
-2007-11-26 17:41 +0000 [r89594] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Add channel locking to a function that needed to be
- doing it. This is just a little something I noticed while working
- on a completely unrelated issue.
-
-2007-11-26 17:36 +0000 [r89587-89592] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_config.c: Use ast_free to free memory, or else we shall
- implode if MALLOC_DEBUG is enabled. (closes issue #11347)
- Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys
- (license 281)
-
- * apps/app_mixmonitor.c: Close the audio file before sending it to
- the post processing application. (closes issue #11357) Reported
- by: reformed Patches: mixmonitor.patch uploaded by reformed
- (license 330)
-
-2007-11-26 17:20 +0000 [r89586] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/app.c: when parsing application options that take arguments,
- don't indicate that the option was supplied unless a
- non-zero-length argument was found for it
-
-2007-11-26 15:48 +0000 [r89580] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Revert vmu->email back to an empty string
- if it was empty when imap_store_file was called. This prevents
- sending a duplicate e-mail. (closes issue #11204, reported by
- spditner, patched by me)
-
-2007-11-26 15:34 +0000 [r89571-89577] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: If channel allocation fails because the alert
- pipe could not be created also free the scheduler context.
- (closes issue #11355) Reported by: eliel Patches:
- main.channel.c.patch uploaded by eliel (license 64)
-
- * apps/app_meetme.c: When unloading app_meetme destroy any auto
- created contexts created by SLA. (closes issue #11367) Reported
- by: eliel
-
-2007-11-25 17:17 +0000 [r89559] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c, configs/res_odbc.conf.sample,
- include/asterisk/res_odbc.h, res/res_config_odbc.c: We previously
- attempted to use the ESCAPE clause to set the escape delimiter to
- a backslash. Unfortunately, this does not universally work on all
- databases, since on databases which natively use the backslash as
- a delimiter, the backslash itself needs to be delimited, but on
- other databases that have no delimiter, backslashing the
- backslash causes an error. So the only solution that I can come
- up with is to create an option in res_odbc that explicitly
- specifies whether or not backslash is a native delimiter. If it
- is, we use it natively; if not, we use the ESCAPE clause to make
- it one. Reported by: elguero Patch by: tilghman (Closes issue
- #11364)
-
-2007-11-24 16:59 +0000 [r89534-89545] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_adsi.c: Free some frames that would otherwise leak on
- error. Reported by: Laureano Patch by: Laureano,tilghman (Closes
- issue #11351)
-
- * apps/app_voicemail.c, main/app.c: Currently, zero-length
- voicemail messages cause a hangup in VoicemailMain. This change
- fixes the problem, with a multi-faceted approach. First, we do
- our best to avoid these messages from being created in the first
- place, and second, if that fails, we detect when the voicemail
- message is zero-length and avoid exiting at that point. Reported
- by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083)
-
- * main/manager.c: Up until this point, the XML output of the
- manager has been technically invalid, due to the repetition of
- certain parameters in a single event. This caused various issues
- for XML parsers, some of which refused to parse at all, given the
- invalidity of the rendered XML. So this commit fixes the XML
- output, ensuring that each entity parameter has a unique name,
- thus ensuring valid XML. Reported by: msetim Patch by: tilghman
- (Closes issue #10220)
-
- * res/res_config_odbc.c: Use ESCAPE clause for the first parameter,
- not just 2nd-Nth parameters. Reported by: apsaras Patch by:
- tilghman (Closes issue #11353)
-
-2007-11-22 17:29 +0000 [r89527] Russell Bryant <russell@digium.com>
-
- * configs/agents.conf.sample: mvanbaak pointed out a spelling error
- in this sample configuration file. While I was at it, I went
- ahead and tweaked it a little bit more.
-
-2007-11-21 19:27 +0000 [r89493-89495] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fix a small error I made in my previous commit
-
- * apps/app_queue.c: Changing an inaccurate debug message to be less
- inaccurate. Under the circumstances, this message would always
- report that there were 0 members available, even though that may
- not be true.
-
-2007-11-21 18:59 +0000 [r89491] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: If a channel gets masqueraded in the middle
- of a park, don't play the announcement to the masqueraded
- channel, and dial back to the original channel on timeout.
-
-2007-11-20 19:16 +0000 [r89461-89462] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/module.h: re-doxygen some comments
-
- * main/loader.c, include/asterisk/module.h,
- build_tools/make_buildopts_h: bring back compile-option checking
- when loading modules, only this time use a string-based storage
- and comparison mechanism because it is easier to support on other
- platforms
-
-2007-11-20 17:50 +0000 [r89457] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: According to comments in main/pbx.c, it is essential
- that if we are going to lock the conlock as well as the hints
- lock, it must be locked in that respective order. In order to
- prevent a potential deadlock, we need to lock the conlock prior
- to locking the hints lock in ast_hint_state_changed (see the call
- stack example on issue #11323 for how this can happen). (closes
- issue #11323, reported by eelcob, suggestion for patch by eelcob,
- patch by me)
-
-2007-11-20 15:22 +0000 [r89450] Steve Murphy <murf@digium.com>
-
- * doc/queues-with-callback-members.txt: closes issue #11324; break
- statements missing in switch cases.
-
-2007-11-20 13:40 +0000 [r89445] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: added RR patch from iroot #10908, thanks.
-
-2007-11-19 15:53 +0000 [r89416-89419] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: Print out the correct filename
- (features.conf) in the log message when parkpos options are
- incorrect. (closes issue #11295) Reported by: Laureano Patches:
- res_features.c.patch uploaded by Laureano (license 265)
-
- * doc/localchannel.txt: Clarify documentation a bit, include that a
- frame has to pass through the core in order for the Local channel
- optimization to happen. (closes issue #11246) Reported by: jon
-
-2007-11-16 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.14 released.
-
-2007-11-16 22:26 +0000 [r89339] Russell Bryant <russell@digium.com>
-
- * main/loader.c, include/asterisk/module.h,
- build_tools/make_buildopts_h: Temporarily revert revision 89325,
- which added md5 magic for keeping track of what build options
- were used. We agreed that we should remove this before making a
- 1.4 release, and then we can put it back in. Then, we can take a
- month or so to play around with it to get it how we want it.
-
-2007-11-16 16:47 +0000 [r89325] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/loader.c, include/asterisk/module.h,
- build_tools/make_buildopts_h: To help combat problems where
- people build external modules (asterisk-addons or others) and
- then change the build options of the Asterisk build in a way that
- makes the incompatible without warning, this commit introduces an
- MD5 signature of the important build-time options and includes
- that signature into modules when they are built. When the loader
- loads one of these modules and notices the problem, it will emit
- a warning to console and refuse to initialize the module, as
- doing so could cause the system to be unstable or even crash. If
- you upgrade to this version of Asterisk, you must rebuild *all*
- of your modules that came from other sources before trying to run
- this version. If you are using Digium's G.729 binary codec
- module, you will need v33 or newer.
-
-2007-11-16 15:28 +0000 [r89323] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Make realtime queues accessible from the
- QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
- patched by atis, with small modifications from me)
-
-2007-11-15 18:37 +0000 [r89298-89302] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile: Start Asterisk in Debian at a more reasonable time
- (since zaptel is at level 20)
-
- * channels/misdn/isdn_lib.c: Fix an uninitialized memory read found
- by valgrind
-
- * channels/chan_iax2.c: Yet another memory corruption issue.
- Reported by: atis Patch by: tilghman Fixes issue #10923
-
-2007-11-15 17:19 +0000 [r89296] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Update the SLAStation application to account
- for the case where the SLA thread has a call out to the station,
- but the user has pressed a line button to answer the call instead
- of picking up the handset. If they do, the phone sends out a new
- INVITE. So, the SLAStation app must check to see if it is picking
- up a ringing trunk, and ensure that the other stations stop
- ringing. (reported internally, patched by me, tested by mogorman)
-
-2007-11-15 14:57 +0000 [r89286-89288] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c: Undoing previous commit since I realize it was
- wrong
-
- * main/manager.c: Adding a missing mutex unlock. (closes issue
- 11256, reported and patched by ys)
-
-2007-11-15 11:26 +0000 [r89280-89281] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't send re-invites during pending INVITE
- transactions. Patch by one47 - thanks! Closes issue #9305
-
- * channels/chan_sip.c: Improve support for multipart messages. Code
- by gasparz, changes by me (mostly formatting). Thanks, gasparz!
- Closes issue #10947
-
-2007-11-14 23:23 +0000 [r89275] Tilghman Lesher <tlesher@digium.com>
-
- * main/app.c: When a recording ends with '#', we are improperly
- trimming an extra 200ms from the recording. Reported by: sim
- Patch by: tilghman Closes issue #11247
-
-2007-11-14 01:15 +0000 [r89260] Joshua Colp <jcolp@digium.com>
-
- * main/srv.c: Return the proper value when the srv_callback
- function executes properly. (closes issue #11240) Reported by:
- jtodd
-
-2007-11-13 21:07 +0000 [r89248-89254] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer
- systems which require a third arg to open() when using O_CREAT.
- Issue 11238, reported by puzzled.
-
- * res/res_features.c: Revert change from revision 67064. It is
- documented behavior that if a parking extension already exists
- while using PARKINGEXTEN, dialplan execution will continue. If
- blind transferring to a Park with PARKINGEXTEN, you must keep
- this in mind, and handle the failure yourself. Issue 11237,
- reported by jon.
-
-2007-11-13 17:34 +0000 [r89246] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: If we set a value for qualify, we should
- actually pay attention to it, instead of overriding the value
-
-2007-11-13 16:02 +0000 [r89241] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_mixmonitor.c: Reverting commit made in revision 89205
- since it is unnecessary. Thanks to Kevin for pointing this out
-
-2007-11-13 13:51 +0000 [r89239] Tilghman Lesher <tlesher@digium.com>
-
- * main/utils.c: Debugging is running into the 16-lock limit.
- Increase to avoid. (This define is only effective when debugging
- is turned on, so there's no effect for most installations.)
-
-2007-11-13 00:56 +0000 [r89205] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If
- only 1 argument is given, then the args.options and
- args.post_process strings are uninitialized and could contain
- garbage. This change handles this situation properly by only
- using arguments that we have parsed.
-
-2007-11-12 20:46 +0000 [r89194] Jason Parker <jparker@digium.com>
-
- * main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev
-
-2007-11-12 20:16 +0000 [r89184-89191] Tilghman Lesher <tlesher@digium.com>
-
- * main/config.c: If two config writes collide, file corruption
- could result. Use a mkstemp() file, instead. Reported by:
- paravoid Patch by: tilghman Closes issue #10781
-
- * main/channel.c, channels/chan_sip.c: Fix two cases of memory
- corruption caused by background threads. Reported by: atis Patch
- by: tilghman Fixes issue #10923
-
-2007-11-12 11:26 +0000 [r89169-89173] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and
- no number was dialed and overlapdial is set, we wait for the ISDN
- timeout instead of starting our own timer. added a comment for
- the misdn.conf.sample for the overlapdial config option.
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
- channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
- restart all interfaces Restart_Indicator, to automatically send a
- RESTART after the L2 of a PTP Port comes up. Also fixed some
- places where we have send a RELEASE without need for it.
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
- state/event issue with overlapdial=yes when no extension matched.
- removed the general sending of a RELEASE_COMPLETE when we receive
- a RELEASE, this is done by mISDNuser/mISDN. This makes it
- possible to use asterisk-1.4 with mISDN trunk, but requires users
- of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6
- (when using the NT mode at all)
-
- * channels/misdn/isdn_lib.c: fixed the support for CW and therefore
- for the reject_cause option.
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- channels/misdn/isdn_lib.h, channels/chan_misdn.c,
- channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
- aded ntkeepcalls option, to avoid droÃpping calls when the L2
- goes down on a PTP link. There are some pbx which do turn off the
- L1 for a very short while and restart it immediately. normally
- T310 should be started and after 10 seconds or so the calls
- should be dropped, this is a simple fix wihtout this timer.
-
-2007-11-08 23:52 +0000 [r89125] Jason Parker <jparker@digium.com>
-
- * main/say.c: Properly say the seconds here.. Issue 11203, fix
- described by vma.
-
-2007-11-08 21:00 +0000 [r89119] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Rework of the commit I made yesterday to use
- the already built-in ast_uri_decode function as opposed to my
- home-rolled one. Also added comments. Thanks to oej for pointing
- me in the right direction
-
-2007-11-08 18:45 +0000 [r89115] Jason Parker <jparker@digium.com>
-
- * configs/res_odbc.conf.sample: Avoid warnings on load when using
- sample configuration files. Issue 11195, patch by eliel.
-
-2007-11-08 16:47 +0000 [r89111] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: I made this same adjustment in trunk to fix
- a bug, and it makes sense to do it in 1.4 as well. If an
- imapfolder is specified in voicemail.conf, don't ever explicitly
- connect to INBOX since it may not exist.
-
-2007-11-08 05:26 +0000 [r89105] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/srv.c: fix a glaring bug in the new SRV record handling that
- would cause incorrect weight sorting
-
-2007-11-08 04:55 +0000 [r89103] Tilghman Lesher <tlesher@digium.com>
-
- * doc/valgrind.txt: Typo
-
-2007-11-08 02:26 +0000 [r89095-89101] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Do not add a sip: to the beginning of the To
- URI unless needed. (closes issue #10756) Reported by: goestelecom
-
- * channels/chan_sip.c: Improve the devicestate logic for multiple
- devices. If any are available then the extension is considered
- available. (closes issue #10164) Reported by: nic_bellamy
- Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic
- (license 299)
-
- * channels/chan_sip.c: Add support for allowing one outgoing
- transaction. This means if a response comes back out of order
- chan_sip will still handle it. I dream of a chan_sip with real
- transaction support. (closes issue #10946) Reported by: flefoll
- (closes issue #10915) Reported by: ramonpeek (closes issue #9567)
- Reported by: atca_pres
-
- * channels/chan_sip.c: If callerid is configured in sip.conf use
- that for checking the presence of an extension in the dialplan.
- (closes issue #11185) Reported by: spditner
-
-2007-11-07 23:39 +0000 [r89093] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c: The member refcount must be incremented, to
- avoid using it after deallocation. A huge thanks go to lvl- for
- patiently providing the necessary valgrind output that was
- necessary to finding this problem of memory corruption. Reported
- by: lvl- Patch by: tilghman Closes issue #11174
-
-2007-11-07 22:40 +0000 [r89090] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: This patch makes it possible for SIP phones
- to dial extensions defined with '#' characters in extensions.conf
- AND maintain their escaped characters when forming URI's (closes
- issue #10681, reported by cahen, patched by me, code review by
- file)
-
-2007-11-07 21:40 +0000 [r89088] Steve Murphy <murf@digium.com>
-
- * cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to
- 10578, I just ran 1.4 thru valgrind; some of the config leakage
- I've already fixed, but it doesn't hurt to double check. I found
- and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major,
- tho.
-
-2007-11-07 15:56 +0000 [r89085] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c: Fixing a segfault in the manager "core show
- channels concise" command. (closes issue #11183, reported by arnd
- and patched by ys)
-
-2007-11-07 04:07 +0000 [r89079] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.ael.sample: Suppress AEL warnings on load.
- Reported by: eliel Patch by: eliel Closes issue #11178
-
-2007-11-06 20:18 +0000 [r89053] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: Fix init_classes() so that classes that
- actually do have files loaded aren't treated as empty, and
- immediately destroyed ...
-
-2007-11-06 19:09 +0000 [r89046] Jason Parker <jparker@digium.com>
-
- * codecs/codec_zap.c: Correctly set the total number of channels
- from a zaptel transcoder board. SPD-49, patch by Matthew
- Nicholson.
-
-2007-11-06 19:09 +0000 [r89045] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: We went to the trouble of creating a
- method of tracking failed trylocks, then never turned it on
- (oops).
-
-2007-11-06 18:53 +0000 [r89042] Olle Johansson <oej@edvina.net>
-
- * main/tdd.c: Bug fixes to tdd support in zaptel.
-
-2007-11-06 18:20 +0000 [r89037] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: If someone were to delete the files used
- by an existing MOH class, and then issue a reload, further use of
- that class could result in a crash due to dividing by zero. This
- set of changes fixes up some places to prevent this from
- happening. (closes issue #10948) Reported by: jcomellas Patches:
- res_musiconhold_division_by_zero.patch uploaded by jcomellas
- (license 282) Additional changes added by me.
-
-2007-11-06 17:52 +0000 [r89036] Steve Murphy <murf@digium.com>
-
- * main/config.c: closes issue #8786 - where the [catname](!) and
- [catname](othercat1,othercat2,...) notation gets dropped across a
- ConfigUpdate (or any other thing that would cause a config file
- to be written). While I was at it, I also cleaned up some of the
- destroy routines to free up comments, which was not being done.
- Made sure the new struct I introduced is also cleaned up properly
- at destruction time. My code handles multiple template
- inclusions. Many thanks to ssokol for his patch, which, while not
- literally used in the final merge, served as a foundation for the
- fix.
-
-2007-11-06 17:08 +0000 [r88994-89032] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make it so that if a peer is determined to
- be unreachable using qualify their devicestate will report back
- unavailable. (closes issue #11006) Reported by: pj
-
- * channels/chan_zap.c: Fix improbable but possible memory leaks in
- chan_zap. (closes issue #11166) Reported by: eliel Patches:
- chan_zap.c.patch uploaded by eliel (license 64)
-
-2007-11-06 13:50 +0000 [r88931] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: Remove some checks to see if locks are
- initialized from the non-DEBUG_THREADS versions of the lock
- routines. These are incorrect for a number of reasons: - It
- breaks the build on mac. - If there is a problem with locks not
- getting initialized, then the proper fix is to find that place
- and fix the code so that it does get initialized. - If additional
- debug code is needed to help find the problem areas, then this
- type of things should _only_ be put in the DEBUG_THREADS
- wrappers.
-
-2007-11-06 02:52 +0000 [r88862] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/srv.h: update comment to match the state of the
- code
-
-2007-11-05 23:29 +0000 [r88826] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: Reworked deadlock avoidance in __ast_read.
- Restored audio to callback agents. (closes issue #11071, reported
- by callguy, patched by me, tested by callguy and Ted Brown)
-
-2007-11-05 22:07 +0000 [r88709-88805] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, include/asterisk/pbx.h: After seeing crashes related
- to channel variables, I went looking around at the ways that
- channel variables are handled. In general, they were not handled
- in a thread-safe way. The channel _must_ be locked when reading
- or writing from/to the channel variable list. What I have done to
- improve this situation is to make pbx_builtin_setvar_helper() and
- friends lock the channel when doing their thing. Asterisk API
- calls almost all lock the channel for you as necessary, but this
- family of functions did not. (closes issue #10923, reported by
- atis) (closes issue #11159, reported by 850t)
-
- * channels/chan_sip.c: When traversing the list of channel
- variables here in transmit_invite(), the asterisk channel must be
- locked, as this data may change at any time. (I have seen
- numerous reports of crashes related to the handling of channel
- variables. There are a couple of issues on the bug tracker
- related to it, but it has also been noted on IRC and mailing
- lists. So, I am finding and fixing some places where channel
- variables are handled improperly.)
-
- * channels/chan_sip.c: Fix up some indentation.
-
- * main/srv.c, include/asterisk/srv.h: Merge changes from
- asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
- record support in Asterisk was broken. There was no guarantee on
- what record Asterisk would choose to actually use. This set of
- changes improves the situation by ensuring that Asterisk will
- choose the highest priority record.
-
- * main/channel.c: Merge the last bit of changes from
- asterisk/team/russell/readq-1.4 The issue here is that the
- channel frame readq handling got broken when the code was
- converted to use the linked list macros. It caused corruption of
- the list head and tail pointers. So, I fixed up the usage of the
- linked list macros and in passing, simplified the code. I also
- documented what the code is doing, as it was a bit difficult to
- figure out at first. This bug showed itself with crashes showing
- messed up head/tail pointers for the readq. However, there are a
- couple of crashes that aren't quite as obvious, but I think may
- be related. So, if your bug gets closed by this commit, but you
- still have a problem, please reopen or create a new bug report.
- (closes issue #10936) (closes issue #10595) (closes issue #10368)
- (closes issue #11084) (closes issue #10040) (closes issue #10840)
-
-2007-11-05 18:47 +0000 [r88671] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If a SIP channel is put on hold multiple
- times do not keep incrementing the onHold value. (closes issue
- #11085) Reported by: francesco_r Tested by: blitzrage (closes
- issue #10474) Reported by: acennami
-
-2007-11-05 17:46 +0000 [r88624] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Fix up datastore handling in ast_do_masquerade().
- The code is intended to move any channel datastores from the old
- channel to the new one. However, it did not use the linked list
- macros properly to accomplish the task. The existing code would
- only work if there was only a single datastore on the old
- channel.
-
-2007-11-05 17:19 +0000 [r88585] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Make sure we destroy the config structure on
- configuration failure. Issue 11163, patch by eliel.
-
-2007-11-05 16:20 +0000 [r88539] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c: Don't check used pooled connections for
- connection status, as it will cause issues for prepared queries.
- Reported by: Nick Gorham (via -dev list) Patch by: tilghman
-
-2007-11-04 22:38 +0000 [r88471] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/stringfields.h, main/channel.c,
- apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c:
- Rename ast_string_field_free_pool to
- ast_string_field_free_memory, and ast_string_field_free_all to
- ast_string_field_reset_all to avoid misuse (due to too similar
- names and an error in documentation). Fix two related memory
- leaks in app_meetme. No need to merge to trunk, different fix
- already applied there. Not applicable to 1.2
-
-2007-11-02 20:49 +0000 [r88328-88366] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make subscribecontext behave as advertised.
- It will now look for the presence of a hint in the given context
- (be it subscribecontext or context). (closes issue #10702)
- Reported by: slavon
-
- * channels/chan_sip.c: If an INFO request within a dialog is
- received with a content length of 0 simply send back a 200 OK. It
- is valid to do this and the remote side is probably using it to
- make sure the signalling is still alive. (closes issue #5747)
- Reported by: chandi Patches: infofix-81430-1.patch uploaded by
- IgorG (license 20)
-
-2007-11-02 16:51 +0000 [r88283] Jason Parker <jparker@digium.com>
-
- * main/say.c: We need to make sure to specify a language to
- ast_fileexists, otherwise it may fail for anything besides en
- Issue 11147, fix discovered by both citats and myself
- (independently), with input from Corydon76
-
-2007-11-02 13:03 +0000 [r88116-88210] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy
- Patch by: ys Closes issue #11143
-
- * doc/valgrind.txt (added): Add some notes on using valgrind
-
-2007-11-01 16:21 +0000 [r88078] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Make sure we set the poll fds to NULL after
- free()ing it. Part of issue 11017, patch by tzafrir.
-
-2007-11-01 13:27 +0000 [r87970-88026] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Fix up commit for my Zap channel with spies in
- Meetme fix. (thanks Tony Mountifield!)
-
- * apps/app_meetme.c: If a Zap channel contains a spy or a spy is
- added take it out of the conference in kernel space and make it
- go through Asterisk so the spy gets audio from both sides.
- (closes issue #10060) Reported by: mparker
-
-2007-10-31 21:23 +0000 [r87906-87908] Jason Parker <jparker@digium.com>
-
- * res/res_jabber.c: Make sure we free some allocated memory before
- returning. Issue 11131, patch by eliel.
-
- * channels/chan_gtalk.c: Don't try to allocate memory that we're
- just going to re-allocate later anyways. Issue 11130, patch by
- eliel.
-
-2007-10-31 18:03 +0000 [r87852] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile: Create samples for ALL of the available options in
- asterisk.conf
-
-2007-10-31 17:49 +0000 [r87775-87849] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan
- save' cli command saves a file where the semicolon is not
- escaped. Fixed this; User also wanted comments to be preserved
- across dialplan save, but this is impossible at this point in
- time, because comments are not stored in the dialplan. They are
- 'compiled' out of extensions.conf. The only way to preserve those
- comments is to use the config file reader/writer that the GUI
- uses to allow online user edits. extensions.conf is first and
- foremost, a config file, and is read in by the normal config-file
- reading routines. Then, it is processed into a dialplan
- (context/exten structs).
-
- * pbx/pbx_ael.c: Included some verbage in the check_includes func,
- to inform the user that included contexts that have no match in
- the AEL, might be OK, as AEL cannot check in the extensions.conf
- or the in-memory contexts, as they may not be there at the time
- of the check.
-
-2007-10-30 23:02 +0000 [r87739] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD
- Reported by: ys Fixed by: ys Closes issue #11116
-
-2007-10-30 21:19 +0000 [r87686] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Merge the changes from
- team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a
- race condition related to the handling of POKEing peers.
- Essentially, a reference to a peer is held by the scheduler when
- there are pending callbacks, but the reference count didn't
- reflect it. So, it was possible for a peer to hit a reference
- count of zero and have its destructor begin to be called at the
- same time that the scheduler thread ran a POKE related callback.
- If that happened, a crash would likely occur. (closes issue
- #11082, closes issue #11094)
-
-2007-10-30 20:29 +0000 [r87650] Jason Parker <jparker@digium.com>
-
- * channels/Makefile: Only try to clean out h323/ if the
- h323/Makefile exists.
-
-2007-10-30 16:13 +0000 [r87571] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: Add two more checks before printing out a
- warning message about bridging. If either channel has hungup of
- course the bridge will have failed. (closes issue #10009)
- Reported by: dimas
-
-2007-10-30 15:45 +0000 [r87567] Jason Parker <jparker@digium.com>
-
- * main/editline/np/vis.c: Fix build of editline on Solaris. Issue
- 11113, patch by snuffy.
-
-2007-10-30 15:10 +0000 [r87534] Joshua Colp <jcolp@digium.com>
-
- * apps/app_followme.c: Return 1.4 to a state where it builds.
- Changing the arguments to a function and not changing where they
- are used is bad, mmmk?
-
-2007-10-30 14:31 +0000 [r87514] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_followme.c: Fix issue where the recorded name wasn't
- getting removed correctly. (closes issue #11115) Reported by:
- davevg Patches: followme-v3.diff
-
-2007-10-29 22:13 +0000 [r87460-87465] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/gsm: missed one directory
-
- * codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs,
- codecs/lpc10, main/db1-ast, main/editline, main,
- codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile,
- codecs/lpc10/Makefile, utils, codecs, agi,
- main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr:
- clean up (and ignore) assembler and preprocessor intermediate
- files if any are created during the build
-
- * Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is
- already there (used when debugging preprocessor issues) because
- the compiler will whine about each compile command
-
-2007-10-29 21:06 +0000 [r87427] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing a completely unnecessary quota
- check from IMAP code.
-
-2007-10-29 20:22 +0000 [r87373-87396] Russell Bryant <russell@digium.com>
-
- * main/utils.c, include/asterisk/lock.h: Add some more details to
- the output of "core show locks". When a thread is waiting for a
- lock, this will now show the details about who currently has it
- locked. (inspired by issue #11100)
-
- * main/astmm.c: Remove a lock that doesn't make any sense. The
- regions lock needs to be held when traversing the list of
- allocated chunks so that they can be printed out to the CLI.
- (Thanks to eliel on #asterisk-dev for pointing this out!)
-
-2007-10-29 17:20 +0000 [r87342] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix issue where if both sides of the dialog
- cancelled the dialog at the same time chan_sip could kepe
- retransmitting a response for no reason. (closes issue #9566)
- Reported by: atca_pres Patches: bug9566.patch uploaded by oej
-
-2007-10-29 17:13 +0000 [r87340] Jason Parker <jparker@digium.com>
-
- * funcs/func_realtime.c, funcs/func_cut.c: Allow some function
- modules to compile under dev mode. Issue 11104, patch by andrew.
-
-2007-10-29 14:23 +0000 [r87294] Joshua Colp <jcolp@digium.com>
-
- * main/utils.c: Fix issue with ast_unescape_semicolon going into an
- endless loop. (closes issue #10550) Reported by: ramonpeek
- Patches: unescape-85177-1.patch uploaded by IgorG (license 20)
-
-2007-10-28 13:46 +0000 [r87262] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c,
- funcs/func_cut.c: Add autoservice to several more functions which
- might delay in their responses. Also, make sure that func_odbc
- functions have a channel on which to set variables. Reported by
- russell Fixed by tilghman Closes issue #11099
-
-2007-10-26 16:34 +0000 [r87168] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c,
- pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c,
- include/asterisk/ael_structs.h, pbx/ael/ael.tab.h,
- utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16,
- pbx/ael/ael.flex: closes issue #11086 where a user complains that
- references to following contexts report a problem; The problem
- was REALLy that he was referring to empty contexts, which were
- being ignored. Reporter stated that empty contexts should be OK.
- I checked it out against extensions.conf, and sure enough, empty
- contexts ARE ok. So, I removed the restriction from AEL. This,
- though, highlighted a problem with multiple contexts of the same
- name. This should be OK, also. So, I added the extend keyword to
- AEL, and it can preceed the 'context' keyword (mixed with
- 'abstract', if nec.). This will turn off the warnings in AEL if
- the same context name is used 2 or more times. Also, I now call
- ast_context_find_or_create for contexts now, instead of just
- ast_context_create; I did this because pbx_config does this. The
- 'extend' keyword thus becomes a statement of intent. AEL can now
- duplicate the behavior of pbx_config,
-
-2007-10-26 13:54 +0000 [r87120] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_curl.c: The addition of autoservice to func_curl
- additionally made func_curl dependent on the existence of a
- channel, with no real reason. This should make func_curl once
- again work without a channel. Reported by jmls. Fixed by
- tilghman. Closes issue #11090
-
-2007-10-25 23:03 +0000 [r87069] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/channel.c, include/asterisk/linkedlists.h: appending one
- list to another should leave the first list empty, and not
- require the user to do that
-
-2007-10-25 22:53 +0000 [r87067] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_cut.c: Backport alternate encoding of newline
- delimiters from trunk to 1.4, as approved by Russell Reported by
- blitzrage Closes issue #10903
-
-2007-10-24 20:56 +0000 [r86982] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Correctly respect hidecalleridname
- configuration option. Simplify code slightly in the process.
- Issue 11079, reported by ddv2005
-
-2007-10-24 04:14 +0000 [r86880-86936] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable
- to specify app:spec in hint arguments
-
- * funcs/func_logic.c: closes issue #11052 -- where nothing after
- the ? will allow un-initialized variable values to corrupt and
- crash asterisk on 64-bit platforms
-
- * main/Makefile: this update to Makefile corrects how ast_expr2f.c
- should be generated
-
- * main/ast_expr2f.c: This should get rid of a really, really
- irritating warning generated by some 64-bit platforms from libc,
- where free(0) is frowned upon
-
-2007-10-22 21:36 +0000 [r86836] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: If lock tracking is not enabled, then we
- can not attempt to log any mutex failures. If so, we could end up
- in infinite recursion. The only lock that is affected by this is
- a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes
- issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
- (license 281)
-
-2007-10-22 17:38 +0000 [r86787] Tilghman Lesher <tlesher@digium.com>
-
- * main/astmm.c: Minor FreeBSD build fix
-
-2007-10-22 16:35 +0000 [r86754-86756] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: After reading online I have confirmed that
- Record-Route headers should be copied to 1xx responses as well.
- (closes issue #10113) Reported by: makoto
-
- * apps/app_controlplayback.c: Make sure res is a positive value
- before performing the check to determine whether the user stopped
- it or not. (closes issue #11023) Reported by: cfc
-
-2007-10-22 15:52 +0000 [r86726-86750] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Don't leak a frame in the case that an END frame
- is received and the time since the BEGIN is less than that of the
- defined minimum DTMF duration. (closes issue #11051) Reported by:
- casper Patches: channel.c.86664.diff uploaded by casper (license
- 55)
-
- * include/asterisk/lock.h: Update the static mutex initializer to
- include the initialization of the internal mutex used to protect
- the lock debugging data. (closes issue #11044, patch suggested by
- Ivan)
-
-2007-10-22 14:48 +0000 [r86694] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Account for the fact that sometimes headers
- may be terminated with \r\n instead of just \n (closes issue
- #11043, reported by yehavi)
-
-2007-10-22 14:27 +0000 [r86630-86663] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Move log message to before the frame it
- references is freed. (closes issue #11050) Reported by: slavon
- Patches: channel.c.86662.diff uploaded by casper (license 55)
-
- * pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes
- issue #11041) Reported by: jsmith Patches:
- asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
- (license 176)
-
- * main/loader.c: Fixes for building under OpenSolaris. (closes
- issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
- uploaded by snuffy (license 35)
-
-2007-10-22 09:21 +0000 [r86598] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send
- DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan
- does not match after an overlap call. Also added out_cause=1
-
-2007-10-19 16:38 +0000 [r86469-86502] Joshua Colp <jcolp@digium.com>
-
- * main/app.c: When returning a DTMF digit from
- ast_control_streamfile cast it as a char so that 0 does not
- overlap with the success return code. (closes issue #11023)
- Reported by: cfc
-
- * channels/chan_sip.c: Fix two issues with domains and transfers.
- If a port was given in the hostname it was treated as part of the
- hostname. If domains were configured but external domains were
- not enabled all transfers would be considered remote. (closes
- issue #11027) Reported by: ramonpeek Patches: 11027-1.diff
- uploaded by ramonpeek (license 266)
-
- * channels/chan_sip.c: Set port number in received as information
- for registrations as well. (closes issue #11028) Reported by:
- brad-x
-
-2007-10-19 01:45 +0000 [r86438] TransNexus OSP Development <support@transnexus.com>
-
- * apps/app_osplookup.c: Fixed OSP module did not report
- source/devinfo IP in correct format.
-
-2007-10-18 22:01 +0000 [r86405-86406] Jason Parker <jparker@digium.com>
-
- * Makefile: Correct documentation. I removed the wrong line..
-
- * Makefile: Add documentation for options in asterisk.conf Issue
- 11029, patch by eserra
-
-2007-10-18 21:16 +0000 [r86330-86372] Russell Bryant <russell@digium.com>
-
- * configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous
- commit.
-
- * configs/iax.conf.sample, channels/chan_iax2.c: Add support for
- setting the maximum trunk size for IAX2 trunking
-
- * main/channel.c, include/asterisk/channel.h: The channel needs to
- stay locked while running timer callbacks, as they access and
- modify channel data that may change elsewhere. I went through
- every timer callback in the source tree to make sure that none of
- them did any additional locking that could introduce deadlocks,
- and all is well. (closes issue #10765) Reported by: Ivan Patches:
- ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license
- 229)
-
-2007-10-18 17:38 +0000 [r86328] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: If a non-existent file is specified to be
- played either as a periodic announcement or as a hold/position
- announcement, the caller would be kicked out of the queue. No
- longer does this happen.
-
-2007-10-18 15:45 +0000 [r86237-86296] Russell Bryant <russell@digium.com>
-
- * codecs/codec_zap.c: Execute the RELEASE operation on transcoder
- channels in the destroy callback. (patch from jsloan)
-
- * main/utils.c: Revert a change that I made for issue #10979 which,
- as has been pointed out to me in issue #11018, doesn't really
- make sense. There is no reason to have the base64 decode function
- force a '\0' terminated buffer, when the result is almost always
- binary, anyway. In fact, this caused some breakage, as some code
- in res_crypto passed in a buffer exactly the right size to get
- its binary result, which got stomped on by this patch. (closes
- issue #11018, reported by dimas)
-
-2007-10-17 21:39 +0000 [r86202] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Changing the strategy field of the call_queue
- struct to be signed instead of unsigned, since the code attempts
- to set the strategy to -1 if you specify a bogus strategy. While
- this isn't a huge issue in 1.4, it could be a problem for someone
- who, say, tries to use the roundrobin strategy in trunk (despite
- all the deprecation warnings in 1.4).
-
-2007-10-17 17:57 +0000 [r86149] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: If Asterisk is in the middle of shutting
- down, respond to OPTIONS with 503 Unavailable. (closes issue
- #10994) Reported by: eserra Patches: sip-options-503.patch
- uploaded by eserra (license 45)
-
-2007-10-17 16:58 +0000 [r86117] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Whoops, forgot to remove the original
- sip_scheddestroy. (closes issue #11010) Reported by: vadim
-
-2007-10-17 15:23 +0000 [r86066] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: When runuser/rungroup is specified, a remote
- console could only be attained by root (Closes issue #9999)
-
-2007-10-17 15:06 +0000 [r86063] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't schedule dialog destruction if a
- MESSAGE is received using an existing dialog. (closes issue
- #11010) Reported by: vadim
-
-2007-10-16 23:35 +0000 [r86028-86032] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample: Since monitor-join is deprecated now,
- remove the example from the sample queues.conf file
-
- * UPGRADE.txt: Updating UPGRADE.txt to reflect the deprecation of
- the monitor-join queue option
-
- * apps/app_queue.c: Adding deprecated warning to monitor-join
- option, since the plan is to no longer support this in favor of
- monitor-type = mixmonitor (related to issue #10885)
-
-2007-10-16 22:36 +0000 [r85994-85997] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: really picky formatting tweak ...
-
- * include/asterisk/lock.h: Some locking errors exposed the fact
- that the lock debugging code itself was not thread safe. How
- ironic! Anyway, these changes ensure that the code that is
- accessing the lock debugging data is thread-safe. Many thanks to
- Ivan for finding and fixing the core issue here, and also thanks
- to those that tested the patch and provided test results. (closes
- issue #10571) (closes issue #10886) (closes issue #10875) (might
- close some others, as well ...) Patches: (from issue #10571)
- ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license
- 229) - a few small changes by me
-
-2007-10-16 21:14 +0000 [r85958] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Trying to remove a non-dynamic queue member via
- dynamic means can lead to some interesting (read nasty)
- situations. This patch clears up the issue by making only dynamic
- queue members removable via dynamic methods.
-
-2007-10-16 19:41 +0000 [r85921] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/localtime.c: Also set up gmtoff (this is used in the
- %z gnu extension to strftime) Reported and fixed by jcmoore
- Closes issue #11002
-
-2007-10-16 19:10 +0000 [r85896] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Remove a pointless lock.
-
-2007-10-16 15:21 +0000 [r85852] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing a double free which happens in the
- statechange thread. (closes issue #10987, reported by andrew)
-
-2007-10-16 14:52 +0000 [r85818-85850] Joshua Colp <jcolp@digium.com>
-
- * apps/app_hasnewvoicemail.c: Check to make sure a value has been
- given to the VMCOUNT dialplan function. (closes issue #10996)
- Reported by: marsosa
-
- * main/threadstorage.c: Fix memory allocation issue in
- threadstorage. (closes issue #10995) Reported by: snuffy Patches:
- new-patch.diff uploaded by snuffy (license 35)
-
-2007-10-16 10:46 +0000 [r85800] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Fix the output for this channel help CLI
- command
-
-2007-10-15 21:10 +0000 [r85717-85720] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Ensure that no pending state changes are leaked
- when the device state change thread gets stopped on module
- unload.
-
- * apps/app_queue.c: Previously, app_queue created a thread to
- handle every single device state change. I changed this a while
- ago in trunk for performance reasons. However, bug 8407 points
- out that it is actually a race condition, causing device state
- changes to get processed in random order. So, I backported my
- changes from trunk to 1.4. (closes issue #8407, patch provided by
- tim_ringenbach, committed patch by me)
-
-2007-10-15 20:29 +0000 [r85687] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: Don't execute a gosub if the arguments is
- zero-len (not just NULL) Reported by davevg Fixed by me Closes
- issue #10985
-
-2007-10-15 20:21 +0000 [r85686] Russell Bryant <russell@digium.com>
-
- * main/say.c: Add a small fix for the tw version of saying dates.
- (closes issue #7827) Reported by: sharkey Patches: say.nits.patch
- uploaded by sharkey (license 172)
-
-2007-10-15 20:15 +0000 [r85684] Jason Parker <jparker@digium.com>
-
- * Makefile: Properly use DESTDIR in 'config' target. Do not try to
- run chkconfig or similar if using DESTDIR. Issue 10938, patch by
- cabal95.
-
-2007-10-15 19:22 +0000 [r85604-85649] Russell Bryant <russell@digium.com>
-
- * main/utils.c: Be pedantic about handling memory allocation
- failure.
-
- * main/utils.c: The loop in the handler for the "core show locks"
- could potentially block for some amount of time. Be a little bit
- more careful and prepare all of the output in an intermediary
- buffer while holding a global resource. Then, after releasing it,
- send the output to ast_cli().
-
- * channels/chan_sip.c: Make the default for the srvlookup option to
- be yes. It doesn't really make sense for it to default to off.
- The default configuration file has it on, and proper RFC
- behavior, as indicated by a comment in the code, is for it to be
- on. So, let's have it on by default to make lives easier. (closes
- issue #10954, suggested by jtodd)
-
-2007-10-15 16:39 +0000 [r85571] Joshua Colp <jcolp@digium.com>
-
- * configs/features.conf.sample: Document that DTMF based features
- only work when two channels are bridged together. (closes issue
- #10773) Reported by: pbayley
-
-2007-10-15 16:34 +0000 [r85561] Russell Bryant <russell@digium.com>
-
- * include/asterisk/strings.h: Make a few changes so that characters
- in the upper half of the ISO-8859-1 character set don't get
- stripped when reading configuration. (closes issue #10982,
- dandre)
-
-2007-10-15 16:22 +0000 [r85559] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Bring both DTMF begin and end frames up through to
- the core for DTMF feature handling. (closes issue #10826)
- Reported by: dimas
-
-2007-10-15 15:40 +0000 [r85556] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Ensure the buffer passed to
- ast_canmatch_extension() is properly initialized so that it is
- null terminated. (issue #10977) Reported by: dimas Patches:
- pbxdundi.patch uploaded by dimas (license 88) - small mods by me
-
-2007-10-15 14:55 +0000 [r85552] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: If Monitor or a spy was added to a P2P or native
- bridged channel bring the channel back to the generic bridging
- core so the monitor or spy operations work. (closes issue #10943)
- Reported by: julianjm
-
-2007-10-15 13:16 +0000 [r85540-85548] Russell Bryant <russell@digium.com>
-
- * main/db.c: Suppress a LOG_DEBUG message if debug is not enabled.
- (closes issue #10980) Reported by: casper Patches:
- db.c.84633.diff uploaded by casper (license 55)
-
- * main/asterisk.c: Make sure remote consoles unmute themselves
- again after reconnecting. (closes issue #10847) Reported by: atis
- Patches: console_unmute_on_reconnect.patch uploaded by atis
- (license 242)
-
- * main/utils.c: Make sure that the base64 decoder returns a
- terminated string. (closes issue #10979) Reported by: ys Patches:
- util.c.diff uploaded by ys (license 281) - small mods by me
-
- * pbx/pbx_config.c: Don't create the context for users in
- users.conf until we know at least one user exists. (closes issue
- #10971) Reported by: dimas Patches: pbxconfig.patch uploaded by
- dimas (license 88)
-
-2007-10-13 15:26 +0000 [r85536] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.ael.sample: Remove deprecated syntax from
- sample ael file Reported and patched by: dimas Closes issue
- #10967
-
-2007-10-13 05:48 +0000 [r85532-85533] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, main/cli.c, include/asterisk/logger.h: Fix an
- issue with console verbosity when running asterisk -rx to execute
- a command and retrieve its output. The issue was that there was
- no way for the main Asterisk process to know that the remote
- console was connecting in the -rx mode. The way that James has
- fixed this is to have all remote consoles muted by default. Then,
- regular remote consoles automatically execute a CLI command to
- unmute themselves when they first start up. (closes issue #10847)
- Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded
- by jamesgolovich (license 176)
-
- * main/asterisk.c, main/cli.c, include/asterisk/cli.h: Properly
- handle the case where read() may return the text for more than
- one CLI command at once for a remote console. (closes issue
- #10888) Reported by: jamesgolovich Patches:
- asterisk-climultiple.diff.txt uploaded by jamesgolovich (license
- 176)
-
-2007-10-12 18:30 +0000 [r85523] Tilghman Lesher <tlesher@digium.com>
-
- * doc/asterisk-mib.txt, doc/PEERING, LICENSE: Change Digium address
-
-2007-10-12 15:45 +0000 [r85515-85517] Russell Bryant <russell@digium.com>
-
- * res/res_smdi.c: Fix a spelling error in a log message. SMDI, not
- SDMI. (closes issue #10959)
-
- * pbx/pbx_realtime.c: Fix the potential use of an uninitialized
- buffer in a log message. (closes issue #10958) Reported by: dimas
- Patches: realtime.patch uploaded by dimas (license 88)
-
-2007-10-11 15:26 +0000 [r85397] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: When creating a new packet don't try to stop
- retransmission of it. It was just allocated/created so it's
- impossible for it to have already been scheduled. (closes issue
- #10945) Reported by: flefoll Patches:
- chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll
- (license 244)
-
-2007-10-11 04:35 +0000 [r85356] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: A dollar sign by itself, not indicating a start of a
- variable or expression prematurely ends substitution (closes
- issue #10939)
-
-2007-10-10 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.13 released.
-
-2007-10-10 15:56 +0000 [r85316] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h: I introduced a new member to the
- ast_filestream struct in 1.4.12, but put it in the middle of the
- struct, instead of at the end. One of the Debian folks, paravoid,
- pointed out that this breaks binary compatability with modules
- compiled against older headers. So, I'm moving the new member to
- the end of the struct to resolve the situation.
-
-2007-10-10 15:51 +0000 [r85315] Mark Michelson <mmichelson@digium.com>
-
- * main/utils.c: The thread ID should be unsigned.
-
-2007-10-10 14:42 +0000 [r85277-85280] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If devicestate is passed a port number strip
- it out. (closes issue #10930) Reported by: ibc
-
- * channels/chan_sip.c: Add support for handling a 182 Queued
- response. (closes issue #10924) Reported by: ramonpeek Patches:
- queued-182.diff uploaded by ramonpeek (license 266)
-
-2007-10-10 14:26 +0000 [r85276] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: A bunch of changes from sprintf to
- snprintf. See security advisory AST-2002-022
-
-2007-10-10 14:14 +0000 [r85242] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Close voicemail message description file if
- duration did not meet the minimum, or else we will eventually run
- out of file descriptors. (closes issue #10918) Reported by:
- brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license
- 279)
-
-2007-10-10 06:24 +0000 [r85195] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/frame.h: use a macro instead of an inline
- function, so that backtraces will report the caller of
- ast_frame_free() properly
-
-2007-10-09 21:55 +0000 [r85158] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, main/utils.c, include/asterisk/lock.h: This
- commit fixes the following issues: - Deadlock in ast_write (issue
- #10406) - Deadlock in ast_read (issue #10406) - Possible mutex
- initialization error in lock.h (issue #10571)
-
-2007-10-09 14:30 +0000 [r84990-85093] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't perform a reinvite if a transfer is in
- progress. (issue #10915) Reported by: ramonpeek
-
- * main/rtp.c: Only update codec information if the channel has a
- technology private structure. (issue #10915) Reported by:
- ramonpeek
-
- * main/rtp.c: Update codec information as well as address when
- doing hold reinvites. (issue #10868) Reported by: mavince
-
- * main/channel.c: Don't keep trying to native bridge if either of
- the channels are involved in a masquerade operation to be done.
- (closes issue #10696) Reported by: tbelder
-
-2007-10-08 03:28 +0000 [r84957] Russell Bryant <russell@digium.com>
-
- * Makefile.rules: Enable file dependency tracking for _all_ builds,
- and not just for builds with dev-mode enabled. I have seen enough
- problems caused by this that I don't think it's worth keeping. I
- want to continue to encourage anybody that is interested to
- continue to run Asterisk from svn. Furthermore, I do not want
- their systems to break when we change a structure definition in a
- header file. :)
-
-2007-10-07 16:15 +0000 [r84890-84902] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Presence packets from a client who's connected
- with our Jabber ID are valid, therefore, those clients must be
- considered as buddies. The resource string helps us make the
- distinction between clients. Closes issue #10707, reported by
- yusufmotiwala.
-
- * res/res_jabber.c: Prevent Asterisk from crashing when receiving a
- presence packet without resource from a buddy that is known to
- have a resource list. Revert a change I previously made, where
- Asterisk could point to a freed memory location.
-
-2007-10-05 19:42 +0000 [r84851] Tilghman Lesher <tlesher@digium.com>
-
- * main/db.c: Log exactly why we can't open the database, if we fail
- (closes issue #10887)
-
-2007-10-05 18:55 +0000 [r84818] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Update the remembered RTP peer information when
- putting an endpoint on hold or taking it off hold so that the RTP
- stack does not initiate a needless reinvite. (closes issue
- #10868) Reported by: mavince
-
-2007-10-05 16:44 +0000 [r84783] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Do deadlock avoidance in a couple more
- places. You can't lock two channels at the same time without
- doing extra work to make sure it succeeds. (closes issue #10895,
- patch by me)
-
-2007-10-05 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.12.1 released. (This is mainly to include the
- app_queue fix for a memory leak on reload, but includes a couple
- of other bug fixes, as well.)
-
-2007-10-05 01:39 +0000 [r84742] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Fix a copy/paste error in the description of
- UpdateConfig that was pointed out by JerJer on #asterisk-dev
-
-2007-10-04 21:57 +0000 [r84692] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Don't allocate space for queue members unless
- it's needed. You end up deleting dynamic members on a reload. Not
- good. closes issue (#10879, reported by dazza76, patched by me)
-
-2007-10-04 21:36 +0000 [r84690] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: callers of sig2str already add the word
- 'signalling' in the appropriate place, so don't duplicate it
-
-2007-10-04 14:51 +0000 [r84637] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Create a duplicate of the channel's member name
- as the tab completion stuff will free it. (closes issue #10884)
- Reported by: adamg
-
-2007-10-03 22:59 +0000 [r84581] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c: When an RFC 2833 event is sent that we don't
- recognize, ignore it, don't queue a NULL digit (closes issue
- #10877)
-
-2007-10-03 18:20 +0000 [r84511-84544] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: closes issue #10870 ; where a CUT() function call
- in a switch expr doesn't execute correctly, because the commas in
- the function args are not converted to vertbars before the func
- is called. I modified just the switch code to convert the commas
- to vertbars if there, but if more of these sort of probs are
- found, I may have to resort to something a little more
- fundamental. We'll see, I guess.
-
- * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-vtest13,
- 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-test3, pbx/pbx_ael.c,
- pbx/ael/ael-test/ref.ael-test5: closes issue #10834 ; where a
- null input to a switch statement results in a hangup; since
- switch is implemented with extensions, and the default case is
- implemented with a '.', and the '.' matches 1 or more remaining
- characters, the case where 0 characters exist isn't matched, and
- the extension isn't matched, and the goto fails, and a hangup
- occurs. Now, when a default case is generated, it also generates
- a single fixed extension that will match a null input. That
- extension just does a goto to the default extension for that
- switch. I played with an alternate solution, where I just tack an
- extra char onto all the patterns and the goto, but not the
- default case's pattern. Then even a null input will still have at
- least one char in it. But it made me nervous, having that extra
- char in , even if that's a pretty secret and low-level issue.
-
-2007-10-02 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.12 released.
-
-2007-10-02 20:06 +0000 [r84474] Russell Bryant <russell@digium.com>
-
- * Makefile, build_tools/prep_tarball: * Don't build the
- menuselect-tree for the tarball, as it requires running the
- configure script first * Change the Makefile to note that
- menuselect-tree depends on the configure script.
-
-2007-10-02 19:01 +0000 [r84410-84437] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Fix some odd formatting I missed..
-
- * res/res_features.c: Finish up on transferee channel before return
- on failure. Issue 10821, patch by Ivan
-
-2007-10-02 14:12 +0000 [r84370] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Use snprintf instead of sprintf in one
- place. There is no vulnerability here due to various buffer sizes
- around the code, but I still didn't like seeing a non
- length-limited copy of data coming off of the wire into a stack
- buffer, as this would be a problem in the future if buffer sizes
- elsewhere got changed or size limitations removed ...
-
-2007-10-02 09:48 +0000 [r84345] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: terminate USERUSER String with 0
-
-2007-10-01 21:52 +0000 [r84291] Jason Parker <jparker@digium.com>
-
- * Makefile, Makefile.rules, channels/Makefile: Add dist-clean
- support for subdirs. Change h323 to only remove the Makefile on a
- dist-clean, rather than a clean. This fixes a bug I found with
- trying to run make after a make clean
-
-2007-10-01 21:25 +0000 [r84274] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/channel.c, main/manager.c, channels/chan_agent.c: moved
- get_base_channel() code from action_redirect to
- ast_channel_masquerade() for issue 7706 and BE-160
-
-2007-10-01 21:18 +0000 [r84273] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: Anything to keep gcc 4.2 happy...
-
-2007-10-01 21:07 +0000 [r84271] Russell Bryant <russell@digium.com>
-
- * main/utils.c, include/asterisk/lock.h: Fulfull a feature request
- from Qwell on the "core show locks" output. It will now note the
- lock type for each lock that a thread holds. (mutex, rdlock, or
- wrlock)
-
-2007-10-01 20:27 +0000 [r84239] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: closes issue
- #10777 -- by returning a null for the parse tree when there's
- really nothing there, and making sure we don't try to do checking
- on a null tree.
-
-2007-10-01 19:56 +0000 [r84166-84236] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: Add another sanity check in the AGI read loop. We
- really don't care about EAGAIN unless we didn't read an entire
- line. If there is a newline at the end if the read buffer, break,
- because we got the whole thing. (reported and patched by bmd)
-
- * include/asterisk/lock.h: Show rwlocks in the "core show locks"
- output. Before, it only showed mutexes.
-
- * channels/Makefile: Remove another file in "make clean". (closes
- issue #10814, paravoid)
-
- * apps/app_dial.c: Simplify the CAN_EARLY_BRIDGE macro a bit.
-
-2007-10-01 14:10 +0000 [r84158-84163] Joshua Colp <jcolp@digium.com>
-
- * configs/usbradio.conf.sample (removed): Remove chan_usbradio
- config file from tree, it is not present in here. (closes issue
- #10839) Reported by: casper
-
- * res/res_musiconhold.c: Fix randomness. save_pos was being set to
- 0 initially instead of -1, causing it to jump to position 0 when
- moh started. (closes issue #10859) Reported by: jamesgolovich
- Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich
- (license 176)
-
- * apps/app_dial.c: Only attempt early bridging if the options given
- to Dial() permit it. (closes issue #10861) Reported by: peekyb
-
-2007-09-30 20:02 +0000 [r84146] Russell Bryant <russell@digium.com>
-
- * include/asterisk/module.h: Fix the AST_MODULE_INFO macro for C++
- modules. The load and reload parameters were in the wrong place.
- (closes issue #10846, alebm)
-
-2007-09-29 23:00 +0000 [r84133-84135] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ael-ntest22/t1/a.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1/b.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1/c.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/d.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/e.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/f.ael (added),
- pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22
- (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added),
- pbx/ael/ael-test/ref.ael-test3,
- pbx/ael/ael-test/ael-ntest22/t3/h.ael (added),
- pbx/ael/ael-test/ref.ael-test4,
- pbx/ael/ael-test/ael-ntest22/t3/i.ael (added),
- pbx/ael/ael-test/ael-ntest22/t3/j.ael (added),
- pbx/ael/ael-test/ael-ntest22/qq.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1 (added),
- pbx/ael/ael-test/ael-ntest22/t2 (added),
- pbx/ael/ael-test/ael-ntest22/t3 (added),
- pbx/ael/ael-test/ael-ntest22/extensions.ael (added),
- pbx/ael/ael-test/ael-ntest22 (added): This is a regression update
- that matches what I did in 84134 for AEL regressions.
-
- * pbx/ael/ael_lex.c, pbx/ael/ael.flex: This issue sort of closes
- 10786; All config files support #include with globbing (you know,
- *,[chars],?,{list,list},etc), so I've updated the AEL system to
- support this also.
-
-2007-09-28 14:13 +0000 [r84049-84078] Tilghman Lesher <tlesher@digium.com>
-
- * main/say.c: Correct pronunciations of numbers for .nl (Closes
- issue #10837)
-
- * main/channel.c: Avoid a deadlock with ALL of the locks in the
- masquerade function, not just the pairs of channels. (Closes
- issue #10406)
-
-2007-09-27 23:12 +0000 [r84018] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/manager.c, channels/chan_agent.c,
- include/asterisk/channel.h: if an Agent is redirected, the base
- channel should actually be redirected. This was causing multiple
- issues, especially issue 7706 and BE-160
-
-2007-09-27 00:01 +0000 [r83976] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: remove a todo item that has been completed
-
-2007-09-26 23:53 +0000 [r83974] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_alsa.c: avoid the weird usage of assert() in the
- ALSA header files that gcc 4.2 wants to complain about
-
-2007-09-26 21:35 +0000 [r83910-83943] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: I changed my mind ... I think this should be
- a LOG_NOTICE.
-
- * channels/chan_sip.c: Add a log message that was requested by the
- masses in the developer tutorial session at Astricon. chan_sip
- did not output any message when a call was rejected because the
- extension was not found. This adds a verbose message (at verbose
- level 3) to note when this happens.
-
- * channels/chan_misdn.c: Fix building chan_misdn under dev-mode.
- (please run the configure script with --enable-dev-mode so this
- doesn't happen again ...)
-
-2007-09-26 18:35 +0000 [r83879] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: Remove unused 4k of memory on the program
- stack (closes issue #10827)
-
-2007-09-25 14:13 +0000 [r83637-83773] Tilghman Lesher <tlesher@digium.com>
-
- * main/app.c: jmls pointed out that unsetting the group and setting
- the group to the blank string aren't quite the same.
-
- * build_tools/make_defaults_h: In the source, keys are relative to
- the datadir, not varlib (which is the same in most cases, but
- it's good to be accurate). Closes issue #10811
-
- * doc/realtime.txt: Oops. Removed the unworkable workaround. This
- note should never have been in the release.
-
- * main/app.c: Making change to group splitting, as discussed on the
- -dev list. The main effect of this will be to permit
- Set(GROUP([cat])=), i.e. unsetting a group.
-
-2007-09-24 07:54 +0000 [r83620] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed round_robin group dial method, this
- never worked well on BRI Ports (2 channels)
-
-2007-09-22 19:39 +0000 [r83558-83589] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: This closes issue #10788 -- The exact same fixes
- are made here for the first arg in the for(arg1; arg2; arg3) {}
- statement, as were done for the 3rd arg. It can now be an
- assignment that will embedded in a Set() app, or a macro call, or
- an app call.
-
- * pbx/pbx_ael.c: This closes issue #10788 -- the 3rd arg in the for
- statement is now wrapped in Set() only if there's an '=' in that
- string. Otherwise, if it begins with '&', then a Macro call is
- generated; otherwise it is made into an app call. A bit more
- accomodating, keeps the new guys happy, and the guys with ael-1
- code should be happy, too
-
-2007-09-21 14:37 +0000 [r83432] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c,
- channels/chan_misdn.c, pbx/ael/ael.tab.c, main/ast_expr2f.c,
- main/file.c, include/asterisk/sched.h, channels/chan_h323.c,
- pbx/pbx_dundi.c, utils/ael_main.c, main/ast_expr2.fl,
- channels/chan_mgcp.c, main/sched.c, res/res_config_pgsql.c,
- main/dnsmgr.c, channels/chan_sip.c, pbx/ael/ael.y,
- main/db1-ast/hash/hash.c, include/asterisk/channel.h,
- channels/chan_iax2.c: gcc 4.2 has a new set of warnings dealing
- with cosnt pointers. This set of changes gets all of Asterisk
- (minus chan_alsa for now) to compile with gcc 4.2. (closes issue
- #10774, patch from qwell)
-
-2007-09-21 13:34 +0000 [r83400] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix video under certain circumstances. It
- would have been possible for the formats on the channel to not
- contain the video format. (closes issue #10782) Reported by:
- cwhuang
-
-2007-09-20 21:16 +0000 [r83316-83348] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: When daemonizing, don't change working directory
- to "/". It makes it not be able to do a core dump when not
- running as uid=root. (closes issue #10766, xrg)
-
- * contrib/scripts/safe_asterisk: Change safe_asterisk to explicitly
- ask for /bin/bash, as it uses bashisms. (closes issue #10772,
- reported by culrich)
-
-2007-09-20 17:09 +0000 [r83246] Jason Parker <jparker@digium.com>
-
- * apps/app_disa.c: If # is pressed after dialing an extension in
- DISA, stop trying to collect more digits. (issue #10754) Reported
- by: atis Patches: app_disa.c.branch.patch uploaded by atis
- (license 242) app_disa.c.trunk.patch uploaded by atis (license
- 242)
-
-2007-09-20 16:25 +0000 [r83230-83232] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make sure the minimum T1 timer value is
- obeyed in all cases. (closes issue #10768) Reported by: flefoll
- Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll
- (license 244) chan_sip.c.br14.83070.retrans-patch uploaded by
- flefoll (license 244)
-
- * channels/chan_sip.c: Fix a minor spelling error. (closes issue
- #10769) Reported by: flefoll Patches:
- chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license
- 244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll
- (license 244)
-
-2007-09-19 19:50 +0000 [r83121-83179] Russell Bryant <russell@digium.com>
-
- * apps/app_system.c: The System() and TrySystem() applications can
- take a substantial amount of time to execute while not servicing
- the channel. So, put the channel in autoservice while the command
- is being executed. (closes issue #10726, reported by mnicholson)
-
- * funcs/func_curl.c: Using curl can take a substantial amount of
- time, so the channel should be autoserviced while waiting for it
- to complete. (closes issue #10725, reported by mnicholson)
-
- * channels/chan_iax2.c: When handling a reload of chan_iax2, don't
- use an ao2_callback() to POKE all peers. Instead, use an
- iterator. By using an iterator, the peers container is not locked
- while the POKE is being done. It can cause a deadlock if the
- peers container is locked because poking a peer will try to lock
- pvt structs, while there is a lot of other code that will hold a
- pvt lock when trying to go lock the peers container. (reported to
- me directly by Loic Didelot. Thank you for the debug info!)
-
- * main/manager.c: Fix up another potential race condition. Do the
- loop decrementing use count on events with the eventq protected
- from being changed. (reported on IRC by Ivan)
-
-2007-09-19 13:47 +0000 [r83070-83074] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Protect the CDR record from modification by
- pbx_exec so that the application data contains the Queue data.
- (closes issue #10761) Reported by: snar Patches:
- app-queue-mixmonitor.patch uploaded by snar (license 245)
-
- * channels/chan_sip.c: (closes issue #10760) Reported by: dimas
- Patches: chan_sip.patch uploaded by dimas (license 88) Read in
- subscribecontext option in general to be the default.
-
-2007-09-19 09:32 +0000 [r83023-83024] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: removed comment which violates the coding
- guidelines.
-
- * channels/misdn_config.c, channels/chan_misdn.c,
- channels/misdn/chan_misdn_config.h: added 'astdtmf' option to
- allow configuring the asterisk dtmf detector instead of the
- mISDN_dsp ones. also added the patch from irroot #10190, so that
- dtmf tones detected by the asterisk detector are passed outofband
- to asterisk, to make any use of dtmf tones at all.
-
-2007-09-19 00:19 +0000 [r82992] Russell Bryant <russell@digium.com>
-
- * apps/app_flash.c: Change the description of app_flash to note how
- it can be a useful tool instead of just saying that it is
- generally a worthless feature. (Thanks to Jim Van Meggelen for
- pointing it out and providing the proposed text)
-
-2007-09-18 23:41 +0000 [r82961] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Initialize a variable to NULL to make the world
- happy.
-
-2007-09-18 22:42 +0000 [r82929] Russell Bryant <russell@digium.com>
-
- * include/asterisk/agi.h, res/res_agi.c: Add a new patch to handle
- interrupting the fgets() call when using FastAGI. This version of
- the patch maintains the original behavior of the code when not
- using FastAGI. (closes issue #10553) Reported by: juggie Patches:
- res_agi_fgets-4.patch uploaded by juggie (license 24)
- res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight
- mods by me Tested by: juggie, festr
-
-2007-09-18 21:49 +0000 [r82887-82913] Doug Bailey <dbailey@digium.com>
-
- * main/manager.c: Corrected patch applied in revision r82887.
-
- * main/manager.c: Fixed a bug where http manager sessions prevented
- the eventq from being cleaned out because http manager sessions
- do not have a valid file descriptor.
-
-2007-09-18 20:56 +0000 [r82867] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Fix a memory leak that can occur on systems under
- higher load. The issue is that when events are appended to the
- master event queue, they use the number of active sessions as a
- use count so it will know when all active sessions at the time
- the event happened have consumed it. However, the handling of the
- number of sessions was not properly synchronized, so the use
- count was not always correct, causing an event to disappear
- early, or get stuck in the event queue for forever. (closes issue
- #9238, reported by bweschke, patch from Ivan, modified by me)
-
-2007-09-18 20:09 +0000 [r82865] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Moving the logic for handling an empty
- membername to the create_member function so that there is a
- common place where this occurs instead of being spread out to
- several different places.
-
-2007-09-18 18:59 +0000 [r82834] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_queue.c: there is no need for conditional logic to
- select ->interface or ->membername, snince ->membername will
- always be populated
-
-2007-09-18 16:31 +0000 [r82802] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: When copying the contents from the wildcard
- peer, do a deep copy instead of shallow copy so that it doesn't
- crash when beging destroyed. (closes issue #10546, patch by me)
-
-2007-09-18 15:28 +0000 [r82751] Jason Parker <jparker@digium.com>
-
- * configs/sip.conf.sample: Correct the allowexternaldomains option
- in SIP sample config. Issue 10753
-
-2007-09-17 20:16 +0000 [r82594-82676] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, main/stdtime/localtime.c: Put a memset in
- ast_localtime() instead of a couple places in app_voicemail to
- prevent the problem everywhere instead of just a couple of
- places. (related to issue #10746)
-
- * apps/app_voicemail.c: Initialize some memory to fix crashes when
- leaving voicemail. This problem was fixed by running Asterisk
- under valgrind. (closes issue #10746, reported by arcivanov,
- patched by me) *** IMPORTANT NOTE: We need to check to see if
- this same bug exists elsewhere.
-
- * res/res_features.c: Handle the case where there are multiple
- dynamic features with the same digit mapping, but won't always
- match the activated on/by access controls. In that case, the code
- needs to keep trying features for a match. (reported by Atis on
- the asterisk-dev list, patched by me)
-
-2007-09-17 16:40 +0000 [r82590-82592] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: revert a change that wasn't supposed to be
- committed... doh!
-
- * apps/app_queue.c, channels/chan_iax2.c: fix a couple of places
- where a logical member name (if specified) was not used, but
- instead the direct interface was listed
-
-2007-09-17 02:00 +0000 [r82514] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: (closes issue #10734) Reported by: asgaroth Instead
- of passing a NULL pointer into snprintf pass "". It makes Solaris
- much happier.
-
-2007-09-14 21:19 +0000 [r82444] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: closes issue #10668; thanks to arkadia for his patch;
- had to leave out the bit about ending the previous cdr in the
- fork; it would destroy current implementations.
-
-2007-09-14 21:17 +0000 [r82435] Russell Bryant <russell@digium.com>
-
- * configs/zapata.conf.sample: Add a note to help clarify the value
- set with the echocancel option. (inspired by Malcolm's blog post
- on blogs.digium.com about HPEC)
-
-2007-09-14 18:35 +0000 [r82396-82398] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Crap, I broke the build. Fixed.
-
- * apps/app_queue.c: Adding member name field to manager events
- where they were missing before (closes issue #10721, reported by
- snar)
-
-2007-09-14 17:48 +0000 [r82394] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: If a channel does not have an owner, do not
- try to set a channel variable. This will end up making the
- channel variable global, which is not right. Closes issue #10720,
- patch by flefoll.
-
-2007-09-14 15:50 +0000 [r82382-82385] Russell Bryant <russell@digium.com>
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
- checking for libusb here, so nobody has to deal with conflicts in
- the chan_usbradio-1.4 branch every time the configure script gets
- changed
-
- * channels/chan_usbradio.c (removed), channels/xpmr (removed),
- channels/Makefile: Remove chan_usbradio from the main 1.4 branch.
- It can't live here because we have a strict policy to not include
- new features in release branches. However, I'm going to merge it
- into trunk, and I also have a special 1.4 based branch that
- includes this module. svn co
- http://svn.digium.com/svn/asterisk/team/jdixon/chan_usbradio-1.4
-
-2007-09-14 14:42 +0000 [r82376] Mark Michelson <mmichelson@digium.com>
-
- * doc/CODING-GUIDELINES: Fixing a typo in the coding guidelines
- (closes issue #10717, reported and patched by leedm777)
-
-2007-09-14 01:24 +0000 [r82368] Jim Dixon <telesistant@hotmail.com>
-
- * apps/app_rpt.c: Fixed problem with changes made to cdr
- functionality
-
-2007-09-14 00:52 +0000 [r82367] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_usbradio.c: this new driver may not live in this
- branch for long (since it is a new feature), but it definitely
- should not be built by default
-
-2007-09-14 00:34 +0000 [r82366] Jim Dixon <telesistant@hotmail.com>
-
- * apps/app_rpt.c, channels/xpmr/xpmr_coef.h (added),
- channels/chan_usbradio.c (added), channels/xpmr/xpmr.h (added),
- channels/xpmr (added), channels/xpmr/LICENSE (added),
- channels/xpmr/sinetabx.h (added), configs/usbradio.conf.sample
- (added), channels/Makefile, channels/xpmr/xpmr.c (added): Added
- channel driver for USB Radio device and support thereof.
-
-2007-09-13 23:11 +0000 [r82358] Jason Parker <jparker@digium.com>
-
- * pbx/pbx_spool.c: Fix a small typo. retrytime > waittime
-
-2007-09-13 20:16 +0000 [r82346] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Preemptively fixing a possible segfault. It is
- possible that queuename is NULL (meaning pause ALL queues), so
- use q->name instead.
-
-2007-09-13 20:11 +0000 [r82344] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_csv.c: Fix a crash that could occur in cdr_csv when
- mutliple threads tried to close the same file. Do we actually
- need the locking here? What happens if you open the same file
- twice, and two threads try to write to it at the same time? Is
- fputs() going to write out the entire line at once? I suspect
- that it could be possible for the second fopen to run during the
- first fputs, so the position could be in the middle of the
- previously written line... Issue 10347, initial patch by
- explidous (but I removed all of the paranoia stuff..)
-
-2007-09-13 18:57 +0000 [r82337-82339] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c: resolve a warning when not building under dev
- mode
-
- * main/astobj2.c, main/asterisk.c, include/asterisk.h: Only compile
- in tracking astobj2 statistics if dev-mode is enabled. Also, when
- dev mode is enabled, register the CLI command that can be used to
- run the astobj2 test and print out statistics.
-
-2007-09-13 18:12 +0000 [r82335] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, LICENSE: Merged revisions 82334 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007)
- | 2 lines clarify the OpenSSL and OpenH323 license exceptions
- ........
-
-2007-09-13 16:25 +0000 [r82326] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Added logic to handle the unlikely case that
- someone has two queues with the same name. Asterisk will log a
- warning message letting the user know that one was already
- defined with that name and is it skipping all further instances.
- This also will work for realtime queues but in order for that to
- happen, the user would have to trigger a perfectly timed reload
- as a realtime queue is being looked up, which is highly unlikely
- (but taken care of nonetheless).
-
-2007-09-13 11:47 +0000 [r82309] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Closes issue #9401, reported and patched
- by irrot, with slight modifications by me. Handle DTMF sent by
- Asterisk properly.
-
-2007-09-12 21:56 +0000 [r82296] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: Fix a check of the wrong pointer, as pointed out
- by an XXX comment left in the code. The problem was harmless,
- however.
-
-2007-09-12 21:28 +0000 [r82291] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/tzfile.h: Oops, wrong location for FreeBSD zone
- files
-
-2007-09-12 20:24 +0000 [r82286] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * apps/app_meetme.c: remove a race condition for the creation of
- recordthread's, and fix a small memory leak. This closes issue#
- 10636
-
-2007-09-12 20:12 +0000 [r82285] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/private.h, main/stdtime/tzfile.h,
- include/asterisk/localtime.h, main/stdtime/localtime.c: Working
- on issue #10531 exposed a rather nasty 64-bit issue on
- ast_mktime, so we updated the localtime.c file from source. Next
- we'll have to write ast_strptime to match.
-
-2007-09-12 15:16 +0000 [r82278-82280] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Clean up the output of "asterisk -h". This
- tweaks the wording and wraps lines at 80 characters. (closes
- issue #10699, seanbright)
-
- * res/res_agi.c: revert patch from issue #10553, as someone not
- using fastagi reported that this broke their system.
-
-2007-09-12 14:30 +0000 [r82274-82276] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Accidentally committed changes to
- app_voicemail which do NOT need to be in the 1.4 branch yet.
- reverting...
-
- * apps/app_voicemail.c, apps/app_queue.c: We should only initialize
- a realtime queue when it is allocated, not every time we access
- it. This prevents the members ao2_container from being
- reallocated every time the queue is accessed. I also removed a
- debug message I had accidentally left in on a previous commit.
-
-2007-09-11 22:37 +0000 [r82267] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix incorrect uses of ao2_find(). Every one of
- these calls was reading bogus memory ...
-
-2007-09-11 21:41 +0000 [r82265] Joshua Colp <jcolp@digium.com>
-
- * codecs/gsm/src/lpc.c, codecs/gsm/src/long_term.c: (closes issue
- #10679) Reported by: andrew Build under dev mode when K6OPTS is
- enabled.
-
-2007-09-11 20:49 +0000 [r82263] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix another missing unref of member objects.
- This one was pointed out by Marta. When building the outgoing
- list in try_calling(), a member reference is stored in each
- outgoing entry. However, when this list got destroyed, the
- reference was not released.
-
-2007-09-11 20:36 +0000 [r82261] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: this change should fix issue # 10659 -- what I worry
- about is how many other bug reports it may generate. Hopefully,
- we can please the/a majority. Hopefully. We shall see. Calls not
- marked ANSWERED and with only one channel name will not be
- posted. This should eliminate the double CDR's.
-
-2007-09-11 16:05 +0000 [r82252] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: All instances of ao2_iterators which were just
- named 'i' have been renamed to 'mem_iter' so that when refcounted
- queues are merged into trunk, there will be little confusion
- regarding iterator names, especially when a queue and member
- iterator are used in the same function.
-
-2007-09-11 16:03 +0000 [r82250] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: The sample dundi.conf claims support for a
- wildcard peer entry - [*], but the code did not support it. This
- patch makes it work. (closes issue #10546, patch by dds, with
- some changes by me)
-
-2007-09-11 16:01 +0000 [r82249] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
- hold/retrieve issue.
-
-2007-09-11 15:26 +0000 [r82245] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: (closes issue #10553) Reported by: juggie Patches:
- res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by:
- juggie When using fastagi, fgets() can return before a full line
- is read. Add explicit handling for the case where it gets
- interrupted.
-
-2007-09-11 14:56 +0000 [r82243] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_dundi.c: (closes issue #10577) Reported by: jamesgolovich
- Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich
- (license 176) Don't leak memory when unloading DUNDi.
-
-2007-09-11 14:34 +0000 [r82198-82240] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Add a couple more missing unrefs of queue
- member objects
-
- * apps/app_queue.c: Add a missing unref of a queue member in an
- error handling block
-
- * apps/app_queue.c: Document why membercount can not simply be
- replaced by ao2_container_count()
-
- * main/astobj2.c: backport astobj2 race condition fix. This
- function is the exact same as trunk so it applies here as well.
-
-2007-09-10 18:02 +0000 [r82155] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c: Convert struct member to use refcounts (closes
- issue #10199)
-
-2007-09-10 15:02 +0000 [r82091] Mark Michelson <mmichelson@digium.com>
-
- * configs/misdn.conf.sample: Removing non-existent options from
- misdn configuration sample. (closes issue #10678, reported and
- patched by IgorG)
-
-2007-09-09 02:35 +0000 [r82028] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Fix inline compiles on really old
- compilers (who uses gcc 2.7 anymore, really?)
-
-2007-09-08 18:41 +0000 [r81952-81997] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Fix a small memory leak. ast_unregister_atexit()
- did not free the entry it removed.
-
- * .cleancount: (closes issue #10672) Bump the cleancount so that a
- "make clean" will be forced. This is needed because my fix in
- revision 81599 made a change to a data structure in file.h, and
- since file dependency tracking is only on with dev-mode enabled,
- file format modules that don't get rebuilt may crash, as is the
- case with this issue. This makes me wonder - how much faster does
- the code build without the file dependency tracking enabled? If
- it doesn't make much of a difference, then it may be worth just
- keeping it on all of the time, or perhaps just not in release
- tarballs, so that this type of issue is avoided.
-
-2007-09-07 19:48 +0000 [r81923] Jason Parker <jparker@digium.com>
-
- * apps/app_queue.c: Allow the MEMBERINTERFACE variable to be used
- as the mixmonitor filename. This moves the setting of the
- MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch
- by sim.
-
-2007-09-07 15:25 +0000 [r81886] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample: Moving the explanation for joinempty
- to a more appropriate place
-
-2007-09-06 22:28 +0000 [r81832] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: (closes issue #9724, closes issue #10374)
- Reported by: kenw Patches: 9724.txt uploaded by russell (license
- 2) Tested by: kenw, russell Resolve a deadlock that occurs when
- doing a SIP transfer to parking. I come across this type of
- deadlock fairly often it seems. It is very important to mind the
- boundary between the channel driver and the core in respect to
- the channel lock and the channel-pvt lock. Channel drivers lock
- to lock the pvt and then the channel once it calls into the core,
- while the core will do it in the opposite order. The way this is
- avoided is by having channel drivers either release their pvt
- lock while calling into the core, or such as in this case,
- unlocking the pvt just long enough to acquire the channel lock.
-
-2007-09-06 22:05 +0000 [r81778-81826] Jason Parker <jparker@digium.com>
-
- * Makefile: We added COPTS for ASTCFLAGS additions, but not LDOPTS
- for ASTLDFLAGS. This adds LDOPTS
-
- * include/asterisk/astobj2.h: This should fix a build issue that
- people building against uClibc were seeing with the addition of
- astobj2
-
-2007-09-06 19:40 +0000 [r81776] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: (closes issue #10122) Reported by:
- stevefeinstein Patches: meetme-unmute-manager.diff uploaded by
- qwell (license 4) Tested by: stevefeinstein After looking over
- the code I agree with Qwell. Setting the file descriptor to
- conference each time just causes a fight back and forth.
-
-2007-09-06 16:56 +0000 [r81743] Philippe Sultan <philippe.sultan@gmail.com>
-
- * include/asterisk/jabber.h, channels/chan_gtalk.c: Various string
- length fixes. Removed an unused variable in aji_client structure
- (context)
-
-2007-09-06 16:25 +0000 [r81682-81713] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixes an issue where valid DTMF had to be
- pressed twice to exit a queue if a member's phone was ringing.
- (closes issue #10655, reported by strider2k, patched by me)
-
- * res/res_features.c: Fixes a memory leak (closes issue #10658,
- reported and patched by Ivan)
-
-2007-09-06 14:20 +0000 [r81650] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: According to both RFC 3920 - section 9.1.2 -
- and Google's XMPP server complaint, if set, the 'from' attribute
- must be set to the user's full JID.
-
-2007-09-05 20:53 +0000 [r81599] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h, main/say.c, res/res_features.c,
- main/file.c, include/asterisk/channel.h: Fix an issue that can
- occur when you do an attended transfer to parking. If you
- complete the transfer before the announcement of the parking spot
- finishes, then the channel being parked will hear the remainder
- of the announcement. These changes make it so that will not
- happen anymore. Basically, res_features sets a flag on the
- channel is playing the announcement to so that the file streaming
- core knows that it needs to watch out for a channel masquerade,
- and if it occurs, to abort the announcement. (closes BE-182)
-
-2007-09-05 17:18 +0000 [r81569] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Solaris x86 compatibility fix
-
-2007-09-05 15:19 +0000 [r81525] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing the build...
-
-2007-09-05 15:14 +0000 [r81523] Jason Parker <jparker@digium.com>
-
- * channels/chan_phone.c: Do not try to unregister a NULL channel
- tech. Also changed load_module function to use defines rather
- than numbers for return values. Issue 10651, patch by
- rbraun_proformatique, with additions by me.
-
-2007-09-05 15:03 +0000 [r81520] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Reverting behavior of QUEUE_MEMBER_COUNT to
- only count members who are logged in and available. (related to
- issue #10652, reported by wuwu)
-
-2007-09-05 13:11 +0000 [r81492] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: (closes issue #10650) Reported by: tacvbo Only
- print out that the spy was removed while holding the spy lock.
-
-2007-09-04 20:54 +0000 [r81453-81455] Jason Parker <jparker@digium.com>
-
- * apps/app_followme.c: Rather than attempt to play a file, we can
- just check whether it exists. Issue 10634, patch by me, testing
- by pabelanger, sanity checked by bweschke
-
- * configs/followme.conf.sample: Change default followme config file
- to point to the correct files. Issue 10644, patch by pabelanger
-
-2007-09-04 18:37 +0000 [r81448] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
- Remove the typedefs on ao2_container and ao2_iterator. This is
- simply because we don't typedef objects anywhere else in
- Asterisk, so we might as well make this follow the same
- convention.
-
-2007-09-04 16:40 +0000 [r81442] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: there is no point in sending 401
- Unauthorized to a UAS that sent us a properly-formatted
- Authentication header with the expected username and nonce but an
- incorrect response (which indicates the shared secret does not
- match)... instead, let's send 403 Forbidden so that the UAS
- doesn't retry with the same authentication credentials repeatedly
-
-2007-09-04 14:23 +0000 [r81435-81439] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: (closes issue #10632) Reported by:
- jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded
- by jamesgolovich (license 176) Fix memory leak when unloading
- chan_iax2. The firmware files were not being freed.
-
- * main/channel.c: (closes issue #10476) Reported by: mdu113 Only
- look for the end of a digit when waiting for a digit. This in
- turn disables emulation in the core.
-
- * main/dns.c: (closes issue #10610) Reported by: john Patches:
- dns.c.patch uploaded by john (license 218) Tested by: mvanbaak
- Don't return a match if no SRV record actually exists.
-
-2007-09-03 18:57 +0000 [r81433] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Remove a couple of calls to
- ast_string_field_free_pools() on peers in error handling blocks
- in the code for building peers. The peer object destructor does
- this and doing it twice will cause a crash. (closes issue #10625,
- reported by and patched by pnlarsson)
-
-2007-09-01 15:57 +0000 [r81426-81428] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Changed a comment to be more accurate. (really
- this is just a test to make sure I can commit properly from home)
-
- * main/astobj2.c, include/asterisk/astobj2.h: Making match_by_addr
- into ao2_match_by_addr and making it available everywhere since
- it could be a handy callback to have
-
-2007-08-31 21:27 +0000 [r81418] Russell Bryant <russell@digium.com>
-
- * include/asterisk/astobj2.h: Remove references to a debugging
- parameter that does not exist
-
-2007-08-31 19:48 +0000 [r81416] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixed broken behavior of a reload on realtime
- queues. Prior to this patch, if a reload was issued and a
- realtime queue had callers waiting in it, then the queue would be
- removed from the queue list, but it would not actually be freed
- (in fact, a debug message warning about a memory leak would come
- up). With this patch, reloads do not touch realtime queues at
- all.
-
-2007-08-31 19:16 +0000 [r81415] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_logic.c: The IF() function was not allowing true
- values that had embedded colons (closes issue #10613)
-
-2007-08-31 18:44 +0000 [r81412] Jason Parker <jparker@digium.com>
-
- * apps/app_dial.c: Re-order dial options to be in line with the
- existing alpha order. Issue 10621, initial patch by junky
-
-2007-08-31 17:38 +0000 [r81410] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Make the 'gtalk show channels' CLI command
- available. Closes issue 10548, reported by keepitcool.
-
-2007-08-31 15:53 +0000 [r81406] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c: Make it the engine's responsible to check for
- the presence of results.
-
-2007-08-31 15:51 +0000 [r81405] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: add missing "transcoder show" (and deprecated
- "show transcoder") CLI commands that were in 1.2 but never added
- to 1.4
-
-2007-08-31 14:38 +0000 [r81401-81403] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (closes issue #10618) Reported by: dimas
- Don't pass through the stopped sounds frame.... just drop it.
-
- * res/res_features.c: (closes issue #10009) Reported by: dimas
- Don't output a bridge failed warning message if it failed because
- one of the channels was part of the masquerade process. That is
- perfectly normal.
-
-2007-08-30 22:05 +0000 [r81397] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Removing an extraneous (and possibly
- misleading) log message. Firstly, if the announce file isn't
- found, the streaming functions will report it. Secondly, not all
- non-zero returns from play_file mean that the announce file
- wasn't found. Positive return values simply mean that a digit was
- pressed (most likely to skip through the announcement). (closes
- issue #10612, reported and patched by dimas)
-
-2007-08-30 21:23 +0000 [r81395] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #10514) Reported by: casper
- Patches: chan_sip.c.80129.diff uploaded by casper (license 55)
- Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible
- for it to ever be that value.
-
-2007-08-30 21:11 +0000 [r81392] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: via issue 10599, where 'CDR already initialized'
- messages are being generated. Since all channels will have an
- init'd CDR attached at creation time, this message is now
- particularly useless. Removed.
-
-2007-08-30 15:38 +0000 [r81383] Russell Bryant <russell@digium.com>
-
- * channels/h323/ast_h323.cxx: Add missing checks for the PTRACING
- define. (closes issue #10559, paravoid)
-
-2007-08-30 15:35 +0000 [r81381] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Changed some manager event messages to reflect
- whether a queue member is a realtime member or not
-
-2007-08-30 15:33 +0000 [r81379] Russell Bryant <russell@digium.com>
-
- * configs/modem.conf.sample (removed), configs/enum.conf.sample,
- configs/extensions.ael.sample: Fix a typo, update a reload
- command, and remove an unused configuration file. (closes issue
- #10606, casper)
-
-2007-08-30 14:53 +0000 [r81375] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: (closes issue #10603) Reported by: jmls Patches:
- pbx.diff uploaded by jmls (license 141) Backport changes from
- 81372. Add REASON dialplan variable for when an originated call
- fails and the failed extension is executed.
-
-2007-08-30 14:43 +0000 [r81373] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: Fixed some warnings.
-
-2007-08-30 14:23 +0000 [r81369] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (issue #10599) Reported by: dimas Handle the
- -1 control subclass during feature dialing (it indicates to stop
- sounds).
-
-2007-08-30 08:31 +0000 [r81367] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: Fixed a severe
- issue where a misdn_read would lock the channel, but read would
- not return because it blocks. later chan_misdn would try to queue
- a frame like a AST_CONTROL_ANSWER which could result in a
- deadlock situation. misdn_read will now not block forever
- anymore, and we don't queue the ANSWER frame at all when we
- already was called with misdn_answer -> answer would be called
- twice. Also we don't explicitly send a RELEASE_COMPLETE on
- receiption of a RELEASE anymore, because mISDN does that for us,
- this resulted in a problem on some switches, which would block
- our port after some calls for a short while.
-
-2007-08-29 16:35 +0000 [r81346-81349] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This patch, in essence, will correctly pause a
- realtime queue member and reflect those changes in the realtime
- engine. (issue #10424, reported by irroot, patch by me) This
- patch creates a new function called update_realtime_member_field,
- which is a generic function which will allow any one field of a
- realtime queue member to be updated. This patch only uses this
- function to update the paused status of a queue member, but it
- lays the foundation for persisting the state of a realtime member
- the same way that static members' state is maintained when using
- the persistentmembers setting
-
- * apps/app_queue.c: Changed some tabs to spaces
-
-2007-08-29 15:57 +0000 [r81342] Russell Bryant <russell@digium.com>
-
- * main/Makefile: If chan_h323 is not being built, don't use g++ to
- do the final link of Asterisk. (in response to a question on the
- asterisk-dev list)
-
-2007-08-29 15:52 +0000 [r81340] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This fix creates a more accurate way of
- detecting whether realtime members were deleted. (closes issue
- 10541, reported by Alric, patched by me) The REALLY nice things
- about this patch is that queue members now have a "realtime"
- field which will be true if the member is a realtime member. This
- means we can check this value prior to certain processing if it
- should ONLY be done for realtime members.
-
-2007-08-29 14:13 +0000 [r81331] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #9690) Reported by: mattv Make
- rtp timeouts work even if two RTP streams are directly bridged in
- the RTP stack.
-
-2007-08-28 21:38 +0000 [r81226-81291] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Change the message about receiving a
- mini-frame before the first full voice frame to a DEBUG message.
-
- * pbx/pbx_dundi.c: revert unintentional changes in rev 81226
-
- * configs/indications.conf.sample, pbx/pbx_dundi.c: Add Russian
- tones. (closes issue #7953, hanabana)
-
-2007-08-28 14:12 +0000 [r81120-81189] Mark Michelson <mmichelson@digium.com>
-
- * contrib/scripts/vmail.cgi: Fixes a forwarding problem when using
- res_config_mysql (closes issue #10573, reported by chrisvaughan,
- patch suggested by chrisvaughan as well)
-
- * apps/app_queue.c: Resolve a potential deadlock. In this case, a
- single queue is locked, then the queue list. In changethread(),
- the queue list is locked, and then each individual queue is
- locked. Under the right circumstances, this could deadlock. As
- such, I have unlocked the individual queue before locking the
- queue list, and then locked the queue back after the queue list
- is unlocked.
-
- * channels/chan_agent.c: DTMF begin frames should be ignored so
- that when an agent acks a call with the '#' key, he doesn't cause
- a queue's announce file to be interrupted. Also went ahead and
- did the same for the '*' key and for ending a call. (closes issue
- #10528, reported by deskhack, patched by me)
-
-2007-08-27 17:27 +0000 [r81042-81074] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Add a \todo to note that this module leaks most
- of the memory it allocates on unload and should be fixed (when
- I'm not in the middle of something else ...).
-
- * pbx/pbx_dundi.c: explicity define a variable as a boolean
-
- * res/res_musiconhold.c: (closes issue #10419) Reported by:
- mustardman Patches: asterisk-mohposition.diff.txt uploaded by
- jamesgolovich (license 176) This patch fixes a few problems with
- music on hold. * Fix issues with starting at the beginning of a
- file when it shouldn't. * Fix the inuse counter to be decremented
- even if the class had not been set to be deleted when not in use
- anymore * Don't arbitrarily limit the number of MOH files to 255
-
-2007-08-27 15:01 +0000 [r81012] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #10561) Reported by: jesselang
- Patches: chan_sip-ChannelReload-20080825.patch uploaded by
- jesselang (license 202) Remove an extra \r\n to make the
- ChannelReload event conform with every other event.
-
-2007-08-27 14:55 +0000 [r81010] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Found a case where the queue's membercount is
- off. It does not take into account dynamic members on a reload.
-
-2007-08-27 13:20 +0000 [r80974] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: (closes issue #10562) Reported by: idkpmiller Correct
- jitter value output in the CLI to be as expected.
-
-2007-08-26 18:11 +0000 [r80932] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Remove an extra signal_condition() for the
- scheduler thread. (closes issue #10564, patch from casper)
-
-2007-08-25 17:37 +0000 [r80895] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix some issues with the handling of the
- scheduler in chan_iax2. Most of the places that scheduled items
- to be executed by the scheduler thread did not signal the
- scheduler thread to wake up so that it could recalculate the time
- until the next action. These changes will make the scheduler
- thread more responsive and ensure that actions get executed as
- close to when intended as possible instead of it being possible
- for very long delays.
-
-2007-08-24 22:59 +0000 [r80878] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * apps/app_zapateller.c: An empty string is an empty callerid ...
- so zap it. This closes issue #10502, which was pointed out by
- dswartz. Thank you, and may the swartz be with you
-
-2007-08-24 21:22 +0000 [r80820-80849] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: If dnsmgr is in use, and no DNS servers are
- available when Asterisk first starts, then don't give up on
- poking peers. Allow the poke to get rescheduled so that it will
- work once the dnsmgr is able to resolve the host. (closes issue
- #10521, patch by jamesgolovich)
-
- * main/dsp.c: Improve the debouncing logic in the DTMF detector to
- fix some reliability issues. Previously, this code used a shift
- register of hits and non-hits. However, if the start of the digit
- isn't clean, it is possible for the leading edge detector to miss
- the digit. These changes replace the flawed shift register logic
- and also does the debouncing on the trailing edge as well.
- (closes issue #10535, many thanks to softins for the patch)
-
-2007-08-24 19:52 +0000 [r80818] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_queue.c: A minor correction to the available logic of
- autofill. If a queue member is paused, they're not really
- "available" so don't count them as such. Somewhat related to
- issue #10155
-
-2007-08-24 18:52 +0000 [r80789] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: From a complaint by jmls, I realize that the message
- in cdr_disposition is unnecessary. To get failure disposition,
- just return -1; no use having more than one case do that.
-
-2007-08-24 15:51 +0000 [r80750] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix a possible crash in IMAP voicemail.
-
-2007-08-24 15:41 +0000 [r80747] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, UPGRADE.txt: Make the deprecation warning inline with
- the code, instead of only in documentation (closes issue #10549)
-
-2007-08-24 15:28 +0000 [r80722] Russell Bryant <russell@digium.com>
-
- * utils/ael_main.c: Tweak the formatting of this MODULEINFO block.
- I think this would have caused a "*" to get in the
- menuselect-tree file.
-
-2007-08-24 14:48 +0000 [r80689-80717] Steve Murphy <murf@digium.com>
-
- * utils/ael_main.c: This change addresses JerJer's complaint that
- aelparse builds and installs even if pbx_ael is unchecked in the
- menuselect stuff.
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test6:
- backport of 80649, a fix to an unreported problem in the ael
- parser, that results in a crash on a 64bit machine
-
-2007-08-24 11:42 +0000 [r80661] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Closes issue #10509 Googletalk calls are
- answered too early, which results in CDRs wrongly stating that a
- call was ANSWERED when the calling party cancelled a call before
- before being established. We must not answer the call upon
- reception of a 'transport-accept' iq packet, but this packet
- still needs to be acknowledged, otherwise the remote peer would
- close the call (like in #8970).
-
-2007-08-23 21:34 +0000 [r80601-80617] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/misdn/isdn_lib.c: make misdn/isdn_lib compile without
- warnings
-
- * channels/chan_misdn.c: make chan_misdn compile without warnings
-
-2007-08-23 20:16 +0000 [r80539-80573] Russell Bryant <russell@digium.com>
-
- * include/asterisk/features.h, res/res_features.c: When executing a
- dynamic feature, don't look it up a second time by digit pattern
- after we already looked it up by name. This causes broken
- behavior if there is more than one feature defined with the same
- digit pattern. (closes issue #10539, reported by bungalow, patch
- by me)
-
- * funcs/func_timeout.c: Revert very broken fix for issue #10540 ...
- none of these values take ms so I don't know what I was thinking
-
- * funcs/func_timeout.c: Fix func_timeout to take values in floating
- point so 1.5 actually means 1.5 seconds instead of being rounded.
- (closes issue #10540, reported by spendergrass, patch by me)
-
-2007-08-23 17:14 +0000 [r80505-80507] Jason Parker <jparker@digium.com>
-
- * /: *sigh*
-
- * /: use autotagged externals
-
-2007-08-23 17:08 +0000 [r80501] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: report the actual channel number that was
- unregistered, instead of assuming that the interface list
- consists of channels 1 through <x> with no gaps in the sequence
-
-2007-08-23 17:02 +0000 [r80360-80499] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix some code where it was possible for a
- reference to a peer to not get released when it should. Thank you
- to Marta Carbone for pointing this out!
-
- * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
- This is a hack to maintain old behavior of chan_iax2. This
- ensures that if the peers and users are being stored in a linked
- list, that they go in the list in the same order that the older
- code used. This is necessary to maintain the behavior of which
- peers and users get matched when traversing the container.
-
- * res/res_agi.c: Revert res_agi fix that didn't quite work until we
- get it right ...
-
- * include/asterisk/astobj2.h: Add some more documentation on
- iterating ao2 containers. The documentation implies that is
- possible to miss an object or see an object twice while
- iterating. After looking through the code and talking with
- mmichelson, I have documented the exact conditions under which
- this can happen (which are rare and harmless in most cases).
-
- * main/astobj2.c: When converting this code to use the list macros,
- I changed it so objects are added to the head of a bucket instead
- of the tail. However, while looking over code with mmichelson, we
- noticed that the algorithm used in ao2_iterator_next requires
- that items are added to the tail. This wouldn't have caused any
- huge problem, but it wasn't correct. It meant that if an object
- was added to a container while you were iterating it, and it was
- added to the same bucket that the current element is in, then the
- new object would be returned by ao2_iterator_next, and any other
- objects in the bucket would be bypassed in the traversal.
-
- * channels/chan_sip.c: Don't crash when using realtime in chan_sip
- without an insecure setting in the database. (closes issue
- #10348, reported by link55, fixed by me)
-
- * main/astobj2.c (added), main/Makefile, include/asterisk/astobj2.h
- (added), doc/iax.txt, UPGRADE.txt, include/asterisk/strings.h,
- channels/chan_iax2.c: Merge changes from
- team/russell/iax_refcount. This set of changes fixes problems
- with the handling of iax2_user and iax2_peer objects. It was very
- possible for a thread to still hold a reference to one of these
- objects while a reload operation tries to delete them. The fix
- here is to ensure that all references to these objects are
- tracked so that they can't go away while still in use. To
- accomplish this, I used the astobj2 reference counted object
- model. This code has been in one of Luigi Rizzo's branches for a
- long time and was primarily developed by one of his students,
- Marta Carbone. I wanted to go ahead and bring this in to 1.4
- because there are other problems similar to the ones fixed by
- these changes, so we might as well go ahead and use the new
- astobj if we're going to go through all of the work necessary to
- fix the problems. As a nice side benefit of these changes, peer
- and user handling got more efficient. Using astobj2 lets us not
- hold the container lock for peers or users nearly as long while
- iterating. Also, by changing a define at the top of chan_iax2.c,
- the objects will be distributed in a hash table, drastically
- increasing lookup speed in these containers, which will have a
- very big impact on systems that have a large number of users or
- peers. The use of the hash table will be made the default in
- trunk. It is not the default in 1.4 because it changes the
- behavior slightly. Previously, since peers and users were stored
- in memory in the same order they were specified in the
- configuration file, you could influence peer and user matching
- order based on the order they are specified in the configuration.
- The hash table does not guarantee any order in the container, so
- this behavior will be going away. It just means that you have to
- be a little more careful ensuring that peers and users are
- matched explicitly and not forcing chan_iax2 to have to guess
- which user is the right one based on secret, host, and access
- list settings, instead of simply using the username. If you have
- any questions, feel free to ask on the asterisk-dev list.
-
- * res/res_agi.c: Juggie in #asterisk-dev was reporting problems
- where fgets would return without reading the whole line when
- using fastagi. When this happens, errno was set to EINTR or
- EAGAIN. This patch accounts for the possibility and lets fgets
- continue in that case.
-
-2007-08-22 18:53 +0000 [r80302-80330] Jason Parker <jparker@digium.com>
-
- * Makefile, build_tools/mkpkgconfig, build_tools/make_build_h,
- build_tools/strip_nonapi, build_tools/prep_moduledeps,
- build_tools/make_buildopts_h: Fix a few build issues in Solaris
- (and likely others). Use GREP and ID variables from autoconf.
- Reported to me in #asterisk-dev I forgot who reported this -
- sorry. :(
-
- * Makefile: Change a syntax that the GNU make in Solaris dislikes.
-
- * build_tools/make_version: Fix a bashism (we explicitly request
- /bin/sh). Remove some oddly placed quotes I found in passing.
-
-2007-08-22 16:21 +0000 [r80257] Russell Bryant <russell@digium.com>
-
- * Makefile: Honor the contents of the COPTS variable as custom
- target CFLAGS. Apparently this is what openwrt does. (reported by
- Brian Capouch on the asterisk-dev list, patch by me)
-
-2007-08-22 16:14 +0000 [r80255] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: (closes issue #10526) Reported by: sinistermidget
- Revert commit from issue #10355 and return timestamp skew to 640.
-
-2007-08-21 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.11 released.
-
-2007-08-21 18:42 +0000 [r80183] Russell Bryant <russell@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@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@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@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@digium.com>
-
- * apps/app_queue.c: Fix the build of app_queue
-
-2007-08-20 21:39 +0000 [r80049-80086] Mark Michelson <mmichelson@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@digium.com>
-
- * pbx/pbx_ael.c: memset really, really needs to be used here.
-
-2007-08-13 21:57 +0000 [r79334] Joshua Colp <jcolp@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@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@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@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@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@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@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@digium.com>
-
- * Asterisk 1.4.10.1 released.
-
-2007-08-10 15:20 +0000 [r78995] Russell Bryant <russell@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@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@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@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@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@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@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@digium.com>
-
- * doc/jabber.txt: Fix mogs email address.
-
-2007-08-08 18:16 +0000 [r78575-78620] Mark Michelson <mmichelson@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@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@digium.com>
-
- * Asterisk 1.4.10 released.
-
-2007-08-07 20:57 +0000 [r78488] Russell Bryant <russell@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@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@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@digium.com>
-
- * channels/chan_sip.c: Allow chan_sip to build in devmode
-
-2007-08-07 19:09 +0000 [r78415] Tilghman Lesher <tilghman@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@digium.com>
-
- * res/res_features.c: (closes issue #10327) Reported by: kkiely
- Instead of directly mucking with the extension/context/priority
- of the channel we are transferring when it has a PBX simply call
- ast_async_goto on it. This will ensure that the channel gets
- handled properly and sent to the right place.
-
- * main/channel.c: (closes issue #10301) Reported by: fnordian
- Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
- (license 110) Restore previous behavior where if we failed to
- lock the channel we wanted we would return to exactly the same
- point as if we had just reentered the function.
-
- * /, apps/app_macro.c: Merged revisions 77767 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4
- lines (closes issue #10334) Reported by: ramonpeek Pass through
- the return value from macro_exec through the MacroIf application.
- ........
-
-2007-07-27 18:15 +0000 [r77571] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Missing newline
-
-2007-07-27 17:04 +0000 [r77536-77540] Joshua Colp <jcolp@digium.com>
-
- * cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois
- Patches: cdr_pgsql.patch uploaded by prashant (license 114)
- Finish the Postgresql connection after the log messages are
- printed so we don't access invalid memory.
-
- * channels/chan_sip.c: (closes issue #10323) Reported by: julianjm
- Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by
- julianjm (license 99) Clear ONHOLD flag when decrementing the
- onHold peer count. If we did not do this the count may keep
- decreasing.
-
-2007-07-27 14:30 +0000 [r77490] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: "re-invite" was misspelled
-
-2007-07-26 23:19 +0000 [r77460] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: (closes issue #10302) Reported by: litnialex If a
- DTMF end frame comes from a channel without a begin and it is
- going to a technology that only accepts end frames (aka INFO)
- then use the minimum DTMF duration if one is not in the frame
- already.
-
-2007-07-26 22:16 +0000 [r77424-77429] Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/mp3.txt: change protocol for downloads as well
-
- * doc/mp3.txt, sounds/Makefile: use new canonical name for download
- server
-
-2007-07-26 21:23 +0000 [r77410] Russell Bryant <russell@digium.com>
-
- * Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined
- in trunk, but not in 1.4. When Asterisk is compiled under dev
- mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
- define it in the same way that trunk does. Also, revert the
- change that added this define in the Makefile The advantage to
- doing it this way is that buildopts.h gets installed when you
- install Asterisk. Then, when building any out of tree modules, or
- building asterisk-addons, these modules know which options the
- rest of Asterisk was built with.
-
-2007-07-26 20:35 +0000 [r77380] Mark Michelson <mmichelson@digium.com>
-
- * Makefile, main/logger.c: Fixes to get ast_backtrace working
- properly. The AST_DEVMODE macro was never defined so the majority
- of ast_backtrace never attempted compilation. The makefile now
- defines AST_DEVMODE if configure was run with --enable-dev-mode.
- Also, changes were made to acccomodate 64 bit systems in
- ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for
- their roles in allowing me to get this committed
-
-2007-07-26 19:32 +0000 [r77348-77350] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/logger.c: Missed one
-
- * main/logger.c: Oops, that builtin define should be all-lowercase.
-
-2007-07-26 18:30 +0000 [r77318] Mark Michelson <mmichelson@digium.com>
-
- * cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur,
- meaning free gets called on the same variable twice. This patch
- sets the connection to NULL after calls to PQfinish so that the
- problem does not occur. Also in this patch, prashant_jois
- informed me that it is safe to pass a null pointer to PQfinish,
- so I have removed the check for conn's existence from
- my_unload_module. (closes issue 10295, reported by junky, patched
- by me with input from prashant_jois)
-
-2007-07-25 22:39 +0000 [r77191] Steve Murphy <murf@digium.com>
-
- * apps/app_meetme.c: This fix solves problem with intense squelch
- noise when someone joins conf in bug 9430; We repro'd the problem
- with meetme opts of 'CciMo'; Josh Colp supplied this patch, and
- I'm applying it. It looks like playing the recorded username will
- louse up the next thing played into the channel. Josh rearranged
- the code so as to start things over before playing data directly
- into the conference.
-
-2007-07-25 22:16 +0000 [r77176] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd
- Add SPEECH_DTMF_TERMINATOR variable so the user can specify the
- digit to terminate a DTMF string with. If none is specified then
- no terminator will be used.
-
-2007-07-25 21:52 +0000 [r77154] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: chan->emulate_dtmf_duration is an unsigned int,
- not a signed int, so use %u instead of %d in the format string
-
-2007-07-25 20:23 +0000 [r77116-77136] Jason Parker <jparker@digium.com>
-
- * /: so are my fingers...
-
- * /: autotagexternals script is still obviously misbehaving...
-
- * /: use autotagged externals
-
-2007-07-25 17:14 +0000 [r77071] Joshua Colp <jcolp@digium.com>
-
- * configure, acinclude.m4: Fix autoconf logic for finding OpenH323
- when it is not in the first place searched (/usr/share/openh323).
-
-2007-07-25 09:34 +0000 [r77022] Luigi Rizzo <rizzo@icir.org>
-
- * main/rtp.c: set the sequence number in a frame for all frame
- types
-
-2007-07-25 00:18 +0000 [r76983] Steve Murphy <murf@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1
- line this fixes bug 10293, where the error message because
- defaultzone or loadzone was not defined was confusing ........
-
-2007-07-24 22:12 +0000 [r76891-76937] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24
- Jul 2007) | 2 lines Oops, res contains the error code, not errno.
- I was wondering why a mutex was reporting "No such file or
- directory"... ........
-
- * main/app.c: Found another place where we should be using the
- umask (thanks jcmoore)
-
-2007-07-24 Jason Parker <jparker@digium.com>
-
- * Asterisk 1.4.9 released.
-
-2007-07-24 16:42 +0000 [r76803-76805] Jason Parker <jparker@digium.com>
-
- * channels/chan_iax2.c: Don't create the Asterisk channel until we
- are starting the PBX on it. (ASA-2007-018)
-
-2007-07-24 16:26 +0000 [r76801] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Added a membercount variable to call_queue
- struct which keeps track of the number of logged in members in a
- particular queue. This makes it so that the 'n' option for
- Queue() can act properly depending on which strategy is used. If
- the strategy is roundrobin, rrmemory, or ringall, we want to ring
- each phone once before moving on in the dialplan. However, if any
- other strategy is used, we will only ring one phone since it
- cannot be guaranteed that a different phone will ring on
- subsequent attempts to ring a phone. As a side effect of this,
- the QUEUE_MEMBER_COUNT dialplan function now just reads the
- membercount variable instead of traversing through the member
- list to figure out how many members there are. Special thanks to
- blitzrage for helping to test this out. (closes issue #10127,
- reported by bcnit, patched by me, tested by blitzrage)
-
-2007-07-23 22:38 +0000 [r76708] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: It was our stated intention for 1.4 that
- files created in app_voicemail should depend upon the umask.
- Unfortunately, mkstemp() creates files with mode 0600, regardless
- of the umask. This corrects that deficiency.
-
-2007-07-23 18:59 +0000 [r76656] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix some incorrect softkey labels in
- messages. Don't try to play dialtone in some unimplemented
- features.
-
-2007-07-23 18:29 +0000 [r76654] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 76653 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul
- 2007) | 4 lines (closes issue #5866) Reported by: tyler Do not
- force channel format changes when a generator is present. The
- generator may have changed the formats itself and changing them
- back would cause issues. ........
-
-2007-07-23 17:57 +0000 [r76620] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Don't try to queue up hold/unhold frames
- on a non-existent channel. Issue 10276.
-
-2007-07-23 17:48 +0000 [r76519-76618] Joshua Colp <jcolp@digium.com>
-
- * apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux
- by putting the value of the digit char in an int.
-
- * /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6
- lines (closes issue #10236) Reported by: homesick Patches:
- rpid_1.4_75840.patch uploaded by homesick (license 91) Accept
- Remote Party ID on guest calls. ........
-
- * channels/chan_skinny.c: (closes issue #10268) Reported by:
- mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak
- (license 7) Add another OS that has to use the Macros for byte
- ordering.
-
-2007-07-23 12:25 +0000 [r76485] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Use a signed integer for storing the number
- of bytes in the packet read from the network. Using an unsigned
- value here made it impossible to handle an error returned from
- recvfrom(). Furthermore, in the case that recvfrom() did return
- an error, this would cause a crash due to a heap overflow.
- (closes issue #10265, reported by and fix suggested by
- timrobbins)
-
-2007-07-21 02:02 +0000 [r76227] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) |
- 4 lines Backport a fix for a memory leak that was fixed in trunk
- in reivision 76221 by rizzo. The memory used for the localaddr
- list was not freed during a configuration reload. ........
-
-2007-07-20 21:36 +0000 [r76211] Steve Murphy <murf@digium.com>
-
- * sounds/Makefile: This patch from 10249 is worth applying! It
- prevents downloading sound files if they are already downloaded.
- Darn Practical, if you ask me
-
-2007-07-20 21:03 +0000 [r76174-76178] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Allow getting a call from an existing
- "sub" channel. Cancel ringing if endpoint hangs up before
- answering. Fixes were backported from trunk (there was apparently
- a bit of confusion during merge of a previous patch). (closes
- issue #10241)
-
- * main/manager.c: Eliminate a compiler warning with gcc 4.2 by
- constifying a char *
-
- * channels/chan_skinny.c: It's possible for sub->owner to be NULL
- here if you cancel the call immediately after/during sending a
- digit.
-
-2007-07-20 18:42 +0000 [r76139] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_directory.c: When using users.conf for the entries in
- the directory, if multiple users had the same last name, only the
- first user listed would be available in the directory. (closes
- issue #10200, reported by mrskippy, patched by me)
-
-2007-07-20 18:22 +0000 [r76132] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Use the define that specifies the default length
- of an artificially created DTMF digit in the ast_senddigit()
- function. The define is set to 100ms by default, which is the
- same thing that this function was using. But, using the define
- lets changes take effect in this case, as well as the others
- where it was already used.
-
-2007-07-20 17:20 +0000 [r76054-76087] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6
- lines (closes issue #10247) Reported by: fkasumovic Patches:
- chan_sip.patch uploaded by fkasumovic (license #101) Drop any
- peer realm authentication entries when reloading so multiple
- entries do not get added to the peer. ........
-
- * res/res_convert.c: (closes issue #10246) Reported by: fkasumovic
- Patches: res_conver.patch uploaded by fkasumovic (license #101)
- Use the last occurance of . to find the extension, not the first
- occurance.
-
- * apps/app_queue.c: Move makeannouncement variable declaration to
- proper place.
-
-2007-07-19 20:36 +0000 [r75980] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Remove some duplicate code.
-
-2007-07-19 18:59 +0000 [r75969-75978] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: The diff on this looks pretty big but all I did
- was remove a pointless if statement (always evaluates true).
-
- * apps/app_queue.c: Changes in handling return values of several
- functions in app_queue. This all started as a fix for issue
- #10008 but now includes all of the following changes: 1.
- Simplifying the code to handle positive return values from ast
- API calls. 2. Removing the background_file function. 3. The fix
- for issue #10008 (closes issue #10008, reported and patched by
- dimas)
-
-2007-07-19 15:53 +0000 [r75928] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) |
- 6 lines When processing full frames, take sequence number
- wraparound into account when deciding whether or not we need to
- request retransmissions by sending a VNAK. This code could cause
- VNAKs to be sent erroneously in some cases, and to not be sent in
- other cases when it should have been. (closes issue #10237,
- reported and patched by mihai) ........
-
-2007-07-18 22:59 +0000 [r75807] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Need to make sure we set milliseconds and
- timestamp - pointed out by the recent ast_ time stuff from
- Tilghman
-
-2007-07-18 21:09 +0000 [r75759] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) |
- 5 lines When traversing the queue of frames for possible
- retransmission after receiving a VNAK, handle sequence number
- wraparound so that all frames that should be retransmitted
- actually do get retransmitted. (issue #10227, reported and
- patched by mihai) ........
-
-2007-07-18 20:40 +0000 [r75749] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007)
- | 2 lines Store prior to copy (closes issue #10193) ........
-
-2007-07-18 20:17 +0000 [r75732] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Umm, why are we transmitting dialtone on
- cfwdall?
-
-2007-07-18 20:00 +0000 [r75712] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c,
- pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these
- Asterisk won't build under devmode using GCC 4.2.
-
-2007-07-18 19:54 +0000 [r75707-75711] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fixes for 7935/7936 conference phones.
- Issue 9245, patch by slimey.
-
- * channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue
- 9887, patches by DEA
-
-2007-07-18 17:56 +0000 [r75658] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 75657 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007)
- | 1 line removed the word 'pissed' from ast_log(...) function
- call for BE-90 ........
-
-2007-07-18 15:44 +0000 [r75583-75623] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Few more places that needs to check for
- onhold state.
-
- * channels/chan_sip.c: (closes issue #10165) Reported by: elandivar
- It is possible for hold status to exist without call limits set,
- so we need to ensure update_call_counter is executed regardless.
-
- * channels/chan_h323.c: Don't bother reloading chan_h323 if it did
- not load successfully in the first place. This would otherwise
- cause a crash.
-
- * pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record
- the threadid of each running thread before shutting them down as
- the thread themselves may change the value.
-
-2007-07-18 12:29 +0000 [r75529] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_meetme.c: Using a freed frame causes crashes (closes
- issue #9317)
-
-2007-07-17 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.8 released.
-
-2007-07-17 20:57 +0000 [r75441-75450] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17
- Jul 2007) | 3 lines Properly check for the length in the skinny
- packet to prevent an invalid memcpy. (ASA-2007-016) ........
-
- * main/rtp.c: cast arguments to ast_log so that it builds without
- warnings for me
-
- * channels/iax2-parser.c, channels/iax2-parser.h, /,
- channels/chan_iax2.c: Merged revisions 75444 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) |
- 5 lines Ensure that when encoding the contents of an ast_frame
- into an iax_frame, that the size of the destination buffer is
- known in the iax_frame so that code won't write past the end of
- the allocated buffer when sending outgoing frames. (ASA-2007-014)
- ........
-
- * /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) |
- 4 lines After parsing information elements in IAX frames, set the
- data length to zero, so that code later on does not think it has
- data to copy. (ASA-2007-015) ........
-
-2007-07-17 20:40 +0000 [r75439] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Ensure that the pointer to STUN data does not go to
- unaccessible memory. (ASA-2007-017)
-
-2007-07-17 20:33 +0000 [r75437] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: (issue #10210) Reported by: juggie Patches:
- 10210-1.4-grr.patch uploaded by juggie (license #24) Tested by:
- juggie, blitzrage Log a warning if someone uses DeadAGI on a live
- channel.
-
-2007-07-17 20:03 +0000 [r75405] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c: Fixing an error I made earlier. ast_fileexists
- can return -1 on failure, so I need to be sure that we only enter
- the if statement if it is successful. Related to my fix to issue
- #10186
-
-2007-07-17 20:01 +0000 [r75401-75403] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: (closes issue #10209) Reported by: juggie Patches:
- 10209-trunk-2.patch uploaded by juggie Tested by: juggie,
- blitzrage In ast_pbx_run(), mark a channel as hung up after an
- application returned -1, or when it runs out of extensions to
- execute. This is so that code can detect that this channel has
- been hung up for things like making sure DeadAGI is used on
- actual dead channels, and is beneficial for other things, like
- making sure someone doesn't try to start spying on a channel that
- is about to go away.
-
- * res/res_agi.c: Remove a duplicated newline character in AGI debug
- output. (closes issue #10207, patch by seanbright)
-
-2007-07-16 20:53 +0000 [r75258-75306] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/dns.c, /: Merged revisions 75304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007)
- | 3 lines provide proper copyright/license attribution for this
- structure that was copied from a BSD-licensed header file long,
- long ago... ........
-
- * /: another fix that is not needed here (finishing up 75251)
-
-2007-07-16 18:16 +0000 [r75253] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c: Restoring functionality from 1.2 wherein
- Retrydial will not exit if there is no announce file specified.
- This change makes it so that if there is no announce file
- specified, the application will continue until finished (or
- caller hangs up). If a bogus announce file is specified, then a
- warning message will be printed saying that the file could not be
- found, but execution will still continue. (closes issue #10186,
- reported by jon, patched by me)
-
-2007-07-16 18:12 +0000 [r75252] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: block change that is not relevant here
-
-2007-07-13 20:36 +0000 [r75108] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13
- Jul 2007) | 3 lines Fix a couple potential minor memory leaks.
- load_moh_classes() could return without destroying the loaded
- configuration. ........
-
-2007-07-13 20:15 +0000 [r75078] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul
- 2007) | 5 lines Fixed an issue where chanspy flags were
- uninitialized if no options were passed. What triggered this
- investigation was an IRC chat where some people's quiet flags
- were set while others' weren't even though none of them had
- specified the q option. ........
-
-2007-07-13 20:10 +0000 [r75053-75067] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13
- Jul 2007) | 6 lines Ensure that adding a user to the list of
- users of a specific music on hold class is not done at the same
- time as any of the other operations on this list to prevent list
- corruption. Using the global moh_data lock for this is not ideal,
- but it is what is used to protect these lists everywhere else in
- the module, and I am only changing what is necessary to fix the
- bug. ........
-
- * channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) |
- 12 lines (closes issue #9660) Reported by: mmacvicar Patches
- submitted by: bbryant, russell Tested by: mmacvicar, marco,
- arcivanov, jmhunter, explidous When using a TDM400P (and probably
- other analog cards) there was a chance that you could hang up and
- pick the phone back up where it has been long enough to be not
- considered a flash hook, but too soon such that the device
- reports that it is busy and the person on the phone will only
- hear silence. This patch makes chan_zap more tolerant of this and
- gives the device a couple of seconds to succeed so the person on
- the phone happily gets their dialtone. ........
-
-2007-07-12 23:00 +0000 [r74998] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Change to my previous fix regarding agent
- logoff soft. Now uses deferlogoff instead of loginstart since
- loginstart is used after logoff. Thanks to makoto for pointing
- this out and suggesting the fix. (closes issue #10178, reported
- and patched by makoto, with modification by me)
-
-2007-07-12 20:42 +0000 [r74955] Steve Murphy <murf@digium.com>
-
- * channels/chan_sip.c: This patch resolves 10143; thanks to irroot
- for the patch; looked acceptable. Let the community decide if it
- messes things up
-
-2007-07-12 19:17 +0000 [r74888-74922] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Whoops... didn't want this to be returned to 0
- each iteration.
-
- * main/channel.c: When waiting for a digit ensure that a begin
- frame was received with it, not just an end frame. (issue #10084
- reported by rushowr)
-
-2007-07-12 16:53 +0000 [r74839-74866] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: It helps if I actually add this stuff for
- the 7921 too - otherwise it won't actually do much of anything.
-
- * channels/chan_skinny.c: Add device ID for 7921 wireless skinny
- phone
-
- * channels/chan_skinny.c: Fix dialing in skinny that was broken in
- some cases. Issue 10136, fix provided by DEA.
-
-2007-07-12 15:53 +0000 [r74815] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul
- 2007) | 2 lines Only print out a warning for situations where it
- is actually helpful. (issue #10187 reported by denke) ........
-
-2007-07-11 22:57 +0000 [r74767] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) |
- 5 lines The function make_trunk() can fail and return -1 instead
- of a valid new call number. Fix the uses of this function to
- handle this instead of treating it as the new call number. This
- would cause a deadlock and memory corruption. (possible cause of
- issue #9614 and others, patch by me) ........
-
-2007-07-11 21:14 +0000 [r74722] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 74719 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11
- Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft"
- did not work...at all. Now it does. (closes issue #10178,
- reported and patched by makoto, with slight modification for 1.4
- and trunk by me) ........
-
-2007-07-11 18:34 +0000 [r74657] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11
- Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows
- the condition that uses LIKE. This fixes realtime extensions with
- ODBC. (closes issue #10175, reported by stuarth, patch by me)
- ........
-
-2007-07-11 18:18 +0000 [r74628-74642] Steve Murphy <murf@digium.com>
-
- * Makefile: This fixes 10172, where the entire man8 dir gets
- removed during an uninstall of asterisk
-
- * utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt:
- further reversion of previously applied floating point stuff for
- expr2
-
-2007-07-11 17:16 +0000 [r74515-74590] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_phone.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Instead of
- figuring out kernel versions that have compiler.h and not...
- let's just use autoconf to check for it's presence. (issue #10174
- reported by francesco_r)
-
- * channels/chan_phone.c: Only check if we need to do a SIGMA based
- tone generation if we have a card. (issue #10179 reported by
- mikowhy)
-
-2007-07-10 23:32 +0000 [r74476] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Forwarding a message with IMAP storage was
- storing the message in the sender's box instead of the forwarded
- mailbox. (closes issue #10138, reported and patched by jaroth)
-
-2007-07-10 19:58 +0000 [r74374-74428] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 74427 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6
- lines Fix an issue where it was possible to have a service level
- of over 100% Between the time recalc_holdtime and update_queue
- was called, it was possible that the call could have been hungup.
- Move both additions to the same place, so this won't happen.
- Issue 10158, initial patch by makoto, modified by me. ........
-
- * main/dns.c: Don't use #if to check if something is defined - use
- #ifdef instead. Pointed out by kpfleming
-
- * /, channels/chan_agent.c: Merged revisions 74376 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul
- 2007) | 4 lines Fix an issue with wrapuptime not working when
- using AgentLogin. Issue 10169, patch by makoto, with a minor mod
- by me to not re-break issue 9618 ........
-
- * main/dns.c, /, configure, include/asterisk/autoconfig.h.in,
- configure.ac: Merged revisions 74373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5
- lines Use res_ndestroy on systems that have it. Otherwise, use
- res_nclose. This prevents a memleak on NetBSD - and possibly
- others. Issue 10133, patch by me, reported and tested by scw
- ........
-
-2007-07-10 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.7.1 released.
-
-2007-07-10 16:00 +0000 [r74323] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: fix an uninitialized variable
-
-2007-07-10 15:38 +0000 [r74317] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4
- lines Fix a small typo in description in of Voicemail()
- application. Issue 10170, patch by casper. ........
-
-2007-07-10 15:31 +0000 [r74314] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10
- Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue
- #10075, this part reported by jmls on IRC, patch by me) ........
-
-2007-07-10 14:50 +0000 [r74262-74265] Joshua Colp <jcolp@digium.com>
-
- * /, main/app.c: Merged revisions 74264 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2
- lines Ensure the group information category exists before trying
- to do a string comparison with it. (issue #10171 reported by
- mlegas) ........
-
- * channels/chan_sip.c: Only spit out an inringing warning message
- when it is applicable. Since call limits are already toast in
- realtime let's not scare the user if they are using it. (issue
- #10166 reported by bcnit)
-
-2007-07-09 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.7 released.
-
-2007-07-09 21:31 +0000 [r74162-74211] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Update the configure script to check for
- a required function that is not present in the 1.2 version of
- libpri. This will prevent the configure script from thinking that
- it has compatible libpri support for Asterisk 1.4, when it
- actually does not because the installed version is from 1.2.
-
- * res/res_musiconhold.c: (closes issue #10123) Reported by:
- blitzrage Patches submitted by: juggie, qwell, me Tested by:
- blitzrage When trying to find a music on hold class to use, try
- all of the options, instead of only the first one that is set.
- Also, change the MusicOnHold applications to not hang up on the
- channel when a class can not be found.
-
-2007-07-09 20:19 +0000 [r74159] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8
- lines Several chan_zap options were not working on reload because
- they were arbitrarily disallowed when reloading some/most PRI
- options (such as signalling) was disallowed. Options such as
- polarityonanswerdelay and answeronpolarityswitch can safely be
- changed on a reload. This corrects that behavior. Issue 9186,
- patch by tzafrir. ........
-
-2007-07-09 18:38 +0000 [r74120-74122] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Forgot to get rid of an extraneous debug
- message.
-
- * apps/app_queue.c: The n option for Queue should make the queue
- exit immediately after failure to reach any members and should
- not be dependent on the timeout value passed to Queue (closes
- issue #10127, reported by bcnit, repaired by me)
-
-2007-07-09 15:32 +0000 [r74082] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_skinny.c: Only destroy the scheduler context if it
- was allocated. (issue #10124 reported by gzero)
-
-2007-07-09 14:57 +0000 [r74047] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixed a logic error in leave_voicemail.
- Pass the mailbox instead of the context to inbox_count when the
- context is "default." (closes issue #10135, reported by yannj,
- repaired by me)
-
-2007-07-09 14:49 +0000 [r74043-74045] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread
- synchronization tweaks. (issue #10124 reported by gzero)
-
- * configure, acinclude.m4: Use AC_CHECK_HEADER to check for
- ptlib/openh323 to allow for cross compiling. (issue #9675
- reported by zandbelt)
-
-2007-07-09 04:03 +0000 [r73985] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while
- 'make progdocs'. (Closes issue #10104)
-
-2007-07-09 03:13 +0000 [r73930-73980] Joshua Colp <jcolp@digium.com>
-
- * main/cdr.c: Give Agent channel names priority when doing CDR
- merging. (issue #10011 reported by krtorio)
-
- * pbx/pbx_config.c: Add a few sanity checks when writing out the
- dialplan. (issue #10157 reported by dome)
-
-2007-07-08 09:47 +0000 [r73849] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: While tracking down a bug, I need some more
- history. Dumphistory is very useful, indeed.
-
-2007-07-06 23:02 +0000 [r73769] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) |
- 4 lines If a sip_pvt struct has already registered an extension
- state callback, remove the old one before adding a new one. If
- this isn't done, Asterisk will crash. (issue #10120) ........
-
-2007-07-06 16:36 +0000 [r73727] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixing a rare case which causes voicemail
- to crash when compiled with IMAP storage. inboxcount has the
- possibility of finding an "interactive" vm_state when no
- persistent "non-interactive" vm_state exists for that mailbox. If
- this should happen when someone attempts to leave a message, it
- results in a crash. This patch, along with my commit in revision
- 72670 fix issue 10053, reported by jaroth. closes issue #10053
-
-2007-07-06 16:12 +0000 [r73679-73696] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06
- Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras
- Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
- with MSSQL 2005 by explicitly stating that '\' is being used as
- an escape character. ........
-
- * /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) |
- 7 lines (closes issue #10125) Reported by: makoto Patches
- submitted by: makoto This fixes a crash in chan_sip that happens
- when the bindaddr setting is not valid on Asterisk startup, gets
- fixed, and then a reload gets issued. ........
-
-2007-07-06 15:27 +0000 [r73675] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 73674 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06
- Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy.
- (issue 9618, reported by jiddings, patched by moi) closes issue
- #9618 ........
-
-2007-07-06 03:34 +0000 [r73551-73629] Russell Bryant <russell@digium.com>
-
- * BUGS: fix a little spelling error
-
- * channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop
- the monitor thread if it was never started. (closes issue #10124,
- reported by gzero, fixed by me)
-
- * channels/chan_iax2.c: copy from the correct buffer when deferring
- a full frame (related to issue #9937)
-
- * channels/chan_iax2.c: * Store the call number that a thread is
- processing without the full frame bit set to ease debugging *
- When deferring a full frame for processing, stick it into the
- queue for the thread that is processing frames for that call, not
- the one that read the current frame and is about to go back into
- the idle list (related to issue #9937)
-
-2007-07-05 22:20 +0000 [r73548] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007)
- | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just
- like we don't support it for G.729 ........
-
-2007-07-05 20:50 +0000 [r73512] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Pass HOLD and UNHOLD frames to the other
- channel when they are returned from a native bridge function.
- This fixes a problem where when two zap channels are natively
- bridged and one does a flash hook, the other channel did not
- receive music on hold. (Reported to me directly by Doug Bailey at
- Digium)
-
-2007-07-05 19:18 +0000 [r73467] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2
- lines Copy language information to the dialog structure when
- calling a peer for situations where a PBX may be started on the
- dialed channel. (issue #10121 reported by clegall_proformatique)
- ........
-
-2007-07-05 15:59 +0000 [r73400] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Correcting a minor CLI bug I found. When
- issuing the queue show command, if you type queue show and then
- press tab, you can continue pressing tab and it will keep
- auto-completing queue names even though only 1 queue can be used
- as an argument.
-
-2007-07-05 15:28 +0000 [r73398] Russell Bryant <russell@digium.com>
-
- * channels/chan_vpb.cc, channels/Makefile: Make this module build
- for me in dev-mode
-
-2007-07-05 14:21 +0000 [r73316-73355] Joshua Colp <jcolp@digium.com>
-
- * apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2
- lines Tweak spy locking. (issue #9951 reported by welles)
- ........
-
- * channels/chan_local.c, /: Merged revisions 73318 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul
- 2007) | 2 lines Actually check to make sure a PBX was started on
- one of the Local channels instead of blindly assuming it was.
- (issue #10112 reported by makoto) ........
-
- * /, apps/app_queue.c: Merged revisions 73315 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2
- lines Reset ServicelevelPerf variable back to 0 if we are unable
- to calculate it each time... otherwise we will get previous
- values. (issue #10117 reported by noriyuki) ........
-
-2007-07-04 14:53 +0000 [r73208-73253] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04
- Jul 2007) | 1 line bchannel configurations like echocancel and
- volume control, need to be setuped on inbound calls too. ........
-
- * channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04
- Jul 2007) | 1 line bad bug in overlapdial case, we called
- start_pbx multiple times, because the state wasn't changed..
- ........
-
-2007-07-03 20:17 +0000 [r73143] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2.c, main/Makefile,
- main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing
- expr floating patch from 1.4; too much of a behavior change. If
- you want this fix, try trunk instead. bug 9508.
-
-2007-07-03 15:42 +0000 [r73104-73106] Jason Parker <jparker@digium.com>
-
- * /: What the heck. This should not have happened.
-
- * /: use autotagged externals
-
-2007-07-03 12:38 +0000 [r73053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_dial.c, /: Merged revisions 73052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007)
- | 2 lines RetryDial should accept a 0 argument, but it does not,
- because atoi does not distinguish between 0 and error (closes
- issue #10106) ........
-
-2007-07-03 08:17 +0000 [r73005] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03
- Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only
- be called from mISDN Source channels.. #9449 ........
-
-2007-07-02 20:16 +0000 [r72933] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput,
- main/Makefile, main/ast_expr2.h, main/ast_expr2.y,
- main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support
- for floating point numbers added to ast_expr2 $\[...\] exprs.
- Fixes bug 9508, where the expr code fails with fp numbers. The
- MATH function returns fp numbers by default, so this fix is
- considered necessary.
-
-2007-07-02 18:18 +0000 [r72926] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Remove a bogus comment and add proper locking to
- the handler function for the CLI command to show information on
- manager actions.
-
-2007-07-02 14:32 +0000 [r72888] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Added additional DTMF debug messages for when
- emulation occurs.
-
-2007-07-02 08:41 +0000 [r72850-72852] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 72585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) |
- 1 line check if the bchannel stack id is already used, if so
- don't use it a second time. Also added a release_chan lock, so
- that the same chan_list object cannot be freed twice. chan_misdn
- does not crash anymore on heavy load with these changes. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
- Merged revisions 72099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) |
- 1 line simplified generation for dummy bchannels, also we mark
- them as dummies, so they are not used later as real-bchannels,
- optimized the RESTART mechanisms, we block a channel now on
- cause:44, and send out a RESTART automatically, then on reception
- of RESTART_ACKNOWLEDGE we unblock the channel again. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged
- revisions 72087 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) |
- 1 line simplified channel finding and locking a lot. removed
- unnecessary #ifdefed areas. ........
-
-2007-07-01 23:52 +0000 [r72806] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) |
- 5 lines When appending lines to call files to keep track of
- retries, write a leading newline just in case the original call
- file did not have a newline at the end. This fix is in response
- to a problem I saw reported on the asterisk-users mailing list.
- ........
-
-2007-06-30 16:50 +0000 [r72705-72766] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Tweak the configure script so that error
- output isn't spewed to the console when searching for GTK2 libs,
- and they aren't found.
-
- * formats/format_pcm.c: give format_pcm a more concise destription
-
-2007-06-29 19:07 +0000 [r72665] Luigi Rizzo <rizzo@icir.org>
-
- * main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for
- absence of the function. This was already done in trunk.
-
-2007-06-29 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.6 released.
-
-2007-06-29 14:26 +0000 [r72597-72599] Joshua Colp <jcolp@digium.com>
-
- * main/cdr.c: Minor change for older GCC versions.
-
- * Makefile, configure, configure.ac, makeopts.in: Backport fix for
- GCC versions without support for declaration-after-statement.
-
-2007-06-29 04:47 +0000 [r72554-72556] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/manager.c: Issue 10055 - Change memory allocation to use the
- heap for a command, since the output has the potential to
- overflow the stack (as it did here)
-
- * res/res_jabber.c: Fix 1.4 breakage
-
-2007-06-28 19:44 +0000 [r72493] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in: regenerate the
- configure script for rizzo
-
-2007-06-28 19:29 +0000 [r72453-72489] Luigi Rizzo <rizzo@icir.org>
-
- * configure.ac: add a check for gethostbyname_r so we can simplify
- the handling e.g. in utils.c Also add comments on a couple of
- features which are not working on FreeBSD. All the above has been
- already done in trunk so the merge must be blocked. Can someone
- please regenerate ./configure ?
-
- * Makefile, channels/chan_zap.c, main/say.c: Add
- -Wdeclaration-after-statement to AST_DEVMODE flags to catch
- variable declarations in the middle of a block. Fix the few
- instances of the above spotted out by the compiler. All of this
- has been already done or is not applicable in trunk, so the merge
- of this change will be blocked.
-
- * apps/app_meetme.c: cast a time_t so that it does not conflict
- with the print format. This change was already done on trunk so
- this change needs to be blocked from merging.
-
-2007-06-27 23:29 +0000 [r72383] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 72373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) |
- 3 lines Reinstating patch. This actually fixes the problem,
- however I was running a development branch without it and
- mistakenly thought it wasn't fixed. Fixes issue #10010, and
- #9654: 100% CPU usage caused by an asterisk console losing it's
- controlling terminal. ........
-
-2007-06-27 23:25 +0000 [r72381] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun
- 2007) | 2 lines Update documentation to clarify variable usage
- with MixMonitor. (issue #9494 reported by netoguy) ........
-
-2007-06-27 23:03 +0000 [r72335] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 72333 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) |
- 2 lines Reverted changes for earlier revisions 72259 to 72261.
- Issue #9654, #10010 ........
-
-2007-06-27 22:58 +0000 [r72328-72331] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to
- our list. Since these are dynamic payloads the other side
- shouldn't care. (issue #9426 reported by irroot)
-
- * /, apps/app_queue.c: Merged revisions 72327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2
- lines Fix issue where queue log events might be missing. (issue
- #7765 reported by mtryfoss) ........
-
-2007-06-27 21:08 +0000 [r72272] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) |
- 5 lines Fix a minor issue with parsing the priority number. You
- could have as much whitespace as you want around a numeric
- priority, but you couldn't have any whitespace around a special
- priority like "n" or "hint". (issue #10039, reported by mitheloc,
- fixed by me) ........
-
-2007-06-27 20:46 +0000 [r72260] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 72259 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) |
- 4 lines Fixes 100% load when controlling terminal disappears.
- Issue #9654, #10010 ........
-
-2007-06-27 20:25 +0000 [r72257] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 72256 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2
- lines I may possibly get shot for doing this... but... defer CDR
- processing until after the channel has been dealt with. This
- should eliminate all of the issues with channels going funky
- (SIP/PRI) when you are posting CDRs to a database that is either
- slow or unavailable and do not want to enable batching. ........
-
-2007-06-27 19:13 +0000 [r72205] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: use the proper type for storing group number
- bits so that if someone specifies 'group=42' it will actually
- work instead of being silently ignored
-
-2007-06-27 18:40 +0000 [r72182-72185] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Fix another problem in voicemail with
- missing symbols. Issue 10074, patch by kryptolus, extended to
- include #if 0'd blocks (just in case)
-
-2007-06-27 17:31 +0000 [r72148] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Make the ast_read_noaudio API call behave better
- under circumstances where DTMF emulation was happening and a
- generator was setup. (issue #10065 reported by stevefeinstein)
-
-2007-06-27 17:10 +0000 [r72125] Jason Parker <jparker@digium.com>
-
- * channels/chan_gtalk.c: Don't modify a variable that we don't want
- modified. Make a copy of it instead. Issue 10029, patch by
- phsultan with slight modifications by me (to remove needless
- casts).
-
-2007-06-27 16:34 +0000 [r72112] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: Only output debug information related to RTCP
- timestamps when RTCP debug is turned on (issue #10066, patch by
- me)
-
-2007-06-27 07:58 +0000 [r72042] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) |
- 1 line for inbound TE calls, we setup the bchannel when we get
- the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready.
- removed some #if 0 areas which weren't used anymore. ........
- r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) |
- 1 line isdn_lib.c didn't compile ........
-
-2007-06-27 00:58 +0000 [r72006] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work.
-
-2007-06-26 23:02 +0000 [r71953] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing a pointless line. This variable
- was already set earlier and between then and this line, there is
- no way that the values on the right side of the assignment could
- have changed.
-
-2007-06-26 20:36 +0000 [r71915] Jason Parker <jparker@digium.com>
-
- * main/rtp.c: Don't dereference a pointer that may be NULL here.
- Issue 10017.
-
-2007-06-26 19:00 +0000 [r71877] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: A few changes, the ultimate goal of which
- is to keep better track of the number of messages that a mailbox
- currently has. A description of the changes: 1. Changed the
- "updated" field of the vm_state struct to act more as a binary
- semaphore than a counting semaphore, since its current
- implementation made the inboxcount function not work properly.
- This change falls in line with a change made by UPenn with their
- IMAP setup and helps to sync our changes with theirs. 2.
- Eliminated some redundant calls to get_vm_state_by_mailbox inside
- leave_voicemail 3. Use the play_folder variable to keep track of
- the number of old and new messages in a mailbox as the messages
- are deleted 4. Added an increment to the number of new messages
- that was not there previously in the leave_voicemail function
-
-2007-06-26 15:47 +0000 [r71796] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixing bug where the authuser was
- mistakenly pulled from the mailbox string instead of the IMAP
- user. (closes issue 10054, reported and patched by jaroth)
-
-2007-06-26 12:27 +0000 [r71657-71751] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007)
- | 2 lines Issue 10062 - Trying to move a message without
- selecting one first results in memory corruption ........
-
- * /, res/res_agi.c: Merged revisions 71656 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007)
- | 2 lines Issue 10035 - handle_exec returns a result inconsistent
- with all of the other AGI commands ........
-
-2007-06-25 14:13 +0000 [r71522-71576] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_h323.c: Build a peer as well when hash323 is
- enabled in users.conf (issue #9599 reported by asagage)
-
- * channels/chan_agent.c: Minor tweak for queueing up the unhold
- frame... this will teach me to do bugs while half asleep. (issue
- #10046 reported by dimas)
-
-2007-06-25 12:40 +0000 [r71519] Russell Bryant <russell@digium.com>
-
- * doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue
- #10048, Matti)
-
-2007-06-25 01:10 +0000 [r71412-71430] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2
- lines Ignore other URIs after the first in a 300 Multiple Choice
- response. (issue #10041 reported by homesick) ........
-
- * main/cdr.c: Fix it so 1.4 actually compiles on my box.
-
- * channels/chan_agent.c: Check to make sure the channel pointer is
- present before queueing up an unhold frame on it. (issue #10046
- reported by dimas)
-
-2007-06-24 20:16 +0000 [r71362-71371] Russell Bryant <russell@digium.com>
-
- * build_tools/prep_tarball: Include the menuselect-tree file in
- tarballs to make builds from tarballs a little bit faster
-
- * main/asterisk.c, /: Merged revisions 71358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) |
- 2 lines Revert the patch from issue 9654 due to an unexpected
- side effect ........
-
-2007-06-24 17:50 +0000 [r71289-71291] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_features.c: Issue 10044 - chan->cdr is NULL here, so
- peer->cdr is what we really wanted to use
-
- * main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24
- Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to
- be able to set variables to the empty string. ........
-
-2007-06-23 03:29 +0000 [r71230] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, res/res_features.c: This patch is meant to fix 8433;
- where clid and src are lost via bridging.
-
-2007-06-22 22:44 +0000 [r71214] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20
- Jun 2007) | 1 line fixed a bug that was introduced by copy and
- paste in the last commit ..bchannels weren't cleaned properly.
- ........
-
-2007-06-22 15:38 +0000 [r71096-71123] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 70672 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) |
- 1 line we activate the bchannels in TE mode on incoming calls
- only when we want to connect the call. ........
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20
- Jun 2007) | 1 line forgot one place .. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20
- Jun 2007) | 1 line on receiption of cause:44 we mark the channel
- as in use and inform the user about the situation, we need to
- test the RESTART stuff then. Also shuffled the
- empty_chan_in_stack function after the bchannel cleaning
- functions, to avoid race conditions. ........
-
- * channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19
- Jun 2007) | 1 line when we send out a SETUP, but get no response,
- we should cleanup everything after reception of a hangup.
- ........
-
- * /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) |
- 1 line restart indicator 0x80 is correct, at least that's what
- libpri does. ........
-
- * channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12
- Jun 2007) | 1 line if the bridged partner is mISDN too we should
- not send dtmf tones, they are transmitted inband always ........
-
- * channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12
- Jun 2007) | 1 line if we have already some digits, we just stop
- the tones. ........
-
-2007-06-22 15:00 +0000 [r71068] Jason Parker <jparker@digium.com>
-
- * apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged
- revisions 71065 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4
- lines Fix a few silly usages of ast_playstream() - it only ever
- returns 0... Issue 10035 ........
-
-2007-06-22 14:53 +0000 [r71066] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 71064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) |
- 10 lines Fixed infinite loop when controlling terminal was lost
- and return value of input function wasn't checked for errors.
- This would cause 100% cpu to be taken up. (closes issue #9654,
- issue #10010) Reported by: mnicholson, and eserra Idea for the
- patch from mnicholson, patched by me ........
-
-2007-06-22 14:10 +0000 [r71063] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: My conditions for merging amaflags info was naive;
- DOCUMENTATION is the default, although null is possible; theft of
- user-settable fields is not good. Just copy them, leave them
- alone.
-
-2007-06-22 03:14 +0000 [r71003] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a small typo which ... well ...
- completely broke chan_iax2. oops! (issue #9937, patch by me)
-
-2007-06-21 22:34 +0000 [r70949] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 70948 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1
- line This little fix is in response to bug 10016, but may not
- cure it. The code is wrong, clearly. In a situation where you set
- the CDR's amaflags, and then ForkCDR, and then set the new CDR's
- amaflags to some other value, you will see that all CDRs have had
- their amaflags changed. This is not good. So I fixed it. ........
-
-2007-06-21 21:40 +0000 [r70899] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2
- lines Don't explode if the gain option is specified without a
- value. (issue #9274 reported by mfarver) ........
-
-2007-06-21 21:14 +0000 [r70866-70883] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Put the thread reading from the socket back
- in the idle list if it deferred the processing of a full frame to
- another thread
-
- * channels/chan_iax2.c: If a full frame is received while one of
- the iax2 threads is in the middle of handling a full frame for
- the same call, queue it up for processing by that same thread
- later instead of dropping it. (issue #9937, patch by me)
-
-2007-06-21 20:19 +0000 [r70841] Steve Murphy <murf@digium.com>
-
- * cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1
- line it was pointed out that the cdr_custom config load could get
- a lock, and under certain circumstances, would never release it.
- I also noted that the situation where more than one mapping spec
- was warned about, but did not ignore further mappings as it had
- promised. I think I have fixed both situations. ........
-
-2007-06-21 19:49 +0000 [r70808] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: When volgain is used don't leave a
- temporary file behind. (Closes Issue 8514, Reported and patched
- by ulogic, code reviewed by Jason Parker)
-
-2007-06-21 15:22 +0000 [r70727] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Do not Packet2Packet bridge if packetization settings
- do not allow it. (issue #9117 reported by phsultan)
-
-2007-06-21 15:21 +0000 [r70726] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Remove a couple of duplicate unlocks
-
-2007-06-21 13:58 +0000 [r70677] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Fix building with ODBC storage enabled.
- (issue #10025 reported by denisgalvao)
-
-2007-06-21 13:00 +0000 [r70656] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: Via complaints aired in asterisk-users, I submit
- these changes, which allow cdr updates to see macro
- context/exten, whether hung up or not
-
-2007-06-20 23:32 +0000 [r70554-70612] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql.
- Issue 10020, patch by my, with credit to prashant_jois for
- pointing out the problem.
-
- * cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race
- condition fix
-
- * cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur
- when reloading the module. Issue 10022, patch by me, with credit
- to prashant_jois for finding the bug.
-
-2007-06-20 22:22 +0000 [r70552] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2
- lines Don't overwrite the configured username setting upon a
- REGISTER. (issue #8565 reported by jsmith) ........
-
-2007-06-20 20:53 +0000 [r70494] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Make sure we clear the previously dialed
- number if it did not exist. Issue 9958.
-
-2007-06-20 19:29 +0000 [r70445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_dial.c, /: Merged revisions 70444 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007)
- | 2 lines Issue 9997 - Timelimit times out the wrong channel
- ........
-
-2007-06-20 18:46 +0000 [r70397] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) |
- 5 lines Fix a problem where an established call would not be
- properly disconnected when a PRI disconnect is received depending
- on which cause code was received. (issue #9588, original patch by
- softins, updated patch from jtexter3, and some additional
- feedback from mhardeman) ........
-
-2007-06-20 17:52 +0000 [r70198-70360] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, main/frame.c: Put the speex packetization values back
- in but disable it when setting up the smoother.
-
- * main/frame.c: Don't do packetization/smoother stuff with speex,
- it doesn't work.
-
-2007-06-20 00:03 +0000 [r70084-70164] Russell Bryant <russell@digium.com>
-
- * contrib/scripts/ast_grab_core: don't delete the backtrace in
- ast_grab_core
-
- * channels/chan_gtalk.c: Only attempt to queue a hangup on the
- owner channel if it actually exists. (issue #9795, patch from
- zandbelt)
-
-2007-06-19 18:23 +0000 [r70062] Steve Murphy <murf@digium.com>
-
- * main/channel.c, /: Merged revisions 70053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1
- line This fixes 9246, where channel variables are not available
- in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
- consolidate the channel variables during a masquerade, and then
- copy the merged variables back onto the clone, so the zombie has
- the same vars that the 'original' has. ........
-
-2007-06-19 17:07 +0000 [r70003] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 69992 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2
- lines Handle the CC field in the RTP header. (issue #9384
- reported by DoodleHu) ........
-
-2007-06-19 16:24 +0000 [r69987] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 69986 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2
- lines Update BRIDGEPEER variable if set to the new channel name
- when a masquerade happens. (issue #9699 reported by dimas)
- ........
-
-2007-06-19 15:22 +0000 [r69944] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Fix a crash that could occur when handing
- device state changes. When the state of a device changes, the
- device state thread tells the extension state handling code that
- it changed. Then, the extension state code calls the callback in
- chan_sip so that it can update subscriptions to that extension. A
- pointer to a sip_pvt structure is passed to this function as the
- call which needs a NOTIFY sent. However, there was no locking
- done to ensure that the pvt struct didn't disappear during this
- process. (issue #9946, reported by tdonahue, patch by me, patch
- updated to trunk to use the sip_pvt lock wrappers by eliel)
-
-2007-06-19 13:55 +0000 [r69805-69895] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2
- lines Perform an extra hangup check just in case. (issue #9589
- reported by bcnit) ........
-
- * /, res/res_features.c: Merged revisions 69846 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2
- lines Add parked call extension AFTER the parking slot has been
- announced, otherwise two threads will try to handle the same
- channel and it will go kaboom. (issue #9191 reported by japple)
- ........
-
- * main/callerid.c: Fix for building on PowerPC under Linux.
-
-2007-06-18 19:48 +0000 [r69796] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * channels/chan_sip.c: Issue 10005 - Segfault with missing
- arguments, plus fix a missing define for SIP INFO channels
-
-2007-06-18 19:00 +0000 [r69775-69794] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't count RTP timeout when involved in a
- T38 fax session. (issue #9222 reported by ivoc)
-
- * /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2
- lines Set the peer name on the dialog to the one configured in
- sip.conf and NOT the username to be used for authentication
- attempts. (issue #9967 reported by achauvin) ........
-
-2007-06-18 17:46 +0000 [r69744] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * contrib/scripts/safe_asterisk, /: Merged revisions 69743 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007)
- | 2 lines Issue 9998 - Remove SIG prefix, since it's not
- supported by ksh ........
-
-2007-06-18 16:51 +0000 [r69708] Joshua Colp <jcolp@digium.com>
-
- * main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called
- for the first time so that it does not needlessly spit out
- changed messages when the host really didn't change.
-
-2007-06-18 16:35 +0000 [r69689-69702] Russell Bryant <russell@digium.com>
-
- * res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c,
- build_tools/menuselect-deps.in, configure, funcs/func_odbc.c,
- include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c:
- To prevent 92138749238754 more reports of "I have unixodbc
- installed, but still can't build *_odbc.so!", check for ltdl
- directly, instead of just listing it as another library to
- include in the unixodbc check in the configure script. This also
- makes ltdl show up as a dependency in menuselect so people know
- what to go install. (related to issue #9989, patch by me)
-
- * build_tools/prep_moduledeps: Change the use of "echo -e" to
- "printf". On systems where /bin/sh is not bash, most of the lines
- in menuselect-tree were getting a "-e" at the beginning of every
- line. I'm surprised nobody noticed this, but I think the XML
- parser was being very nice and ignoring them.
-
-2007-06-18 16:04 +0000 [r69661-69668] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't defer the BYE till later on a transfer
- when the transfer itself goes kaboom and has no hope of working.
-
- * channels/chan_sip.c: Few minor transfer tweaks. We can't unlock
- something we never locked, and better handle a specific scenario
- with doing an attended transfer between two non-bridged calls.
-
-2007-06-18 15:46 +0000 [r69660] Russell Bryant <russell@digium.com>
-
- * Makefile: Tweak paths for BSD systems (issue #10001, stuarth)
-
-2007-06-18 13:55 +0000 [r69625] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix issue where it would be possible for the
- negotiated codecs to get set back to nothing. (issue #9992
- reported by yehavi)
-
-2007-06-15 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.5 released.
-
-2007-06-15 20:18 +0000 [r69579] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Fix a silly deadlock in res_features that I
- found while debugging on one of blitzrage's test machines. It was
- one of the situations where he was seeing hung channels, and may
- be the cause of some of the reports from other people. (related
- to issue #9235)
-
-2007-06-15 19:23 +0000 [r69558] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Add support for setting the maximum
- length of acceptable DTMF in SpeechBackground.
-
-2007-06-15 15:27 +0000 [r69518] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: The SLATRUNK_STATUS variable indicated
- "SUCCESS" for both an answer of the incoming call on the trunk,
- or if the trunk reached its ring timeout. This patch changes the
- variable to say "RINGTIMEOUT" in that case. (issue #9973,
- reported by n00dle, patch by me)
-
-2007-06-14 23:22 +0000 [r69434-69470] Jason Parker <jparker@digium.com>
-
- * main/config.c, /: Merged revisions 69469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4
- lines Fix an issue where the line number in an unterminated
- comment block error message would show the wrong line number.
- "Reported" to me on #asterisk (somebody posted an error message,
- and I happened to catch it) ........
-
- * sounds/Makefile: Update to latest versions of sound files.
-
-2007-06-14 21:50 +0000 [r69392] Kevin P. Fleming <kpfleming@digium.com>
-
- * cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c,
- cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c,
- main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c,
- apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c,
- main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
- channels/chan_iax2.c: use ast_localtime() in every place
- localtime_r() was being used
-
-2007-06-14 21:08 +0000 [r69358] Russell Bryant <russell@digium.com>
-
- * main/say.c: Fix some problems with saying dates and times for the
- "tw" langauge (issue #9964, ljmid)
-
-2007-06-14 15:21 +0000 [r69259] Jason Parker <jparker@digium.com>
-
- * funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun
- 2007) | 4 lines Change a quite broken while loop to a for loop,
- so "continue;" works as expected instead of eating 99% CPU...
- Issue 9966, patch by me. ........
-
-2007-06-13 21:19 +0000 [r69184-69222] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Whoops...
-
- * channels/chan_iax2.c: Let's make chan_iax2 media only native
- transfers actually work. (issue #9376 reported by simone
- cittadini)
-
- * channels/iax2-parser.c: Add TXMEDIA to list so that it is
- properly displayed during iax2 packet output.
-
-2007-06-13 19:57 +0000 [r69183] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Move the logic for destroying a call when no
- response is received to a BYE outside of the block that checks
- for FLAG_FATAL to be set. This flag is only set when the packet
- is transmitted with the reliability set to XMIT_CRITICAL when the
- original packet is transmitted. A BYE is always sent with it set
- to XMIT_RELIABLE, meaning this code could never be encountered.
- This resulted in seeing some SIP channels that would never go
- away with the last packet sent being a BYE. (part of issue #9235,
- patch from jcmoore)
-
-2007-06-13 19:41 +0000 [r69181] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Contains a patch for fixing an encoding
- problem when using Outlook to view voicemail emails and
- attachments. This fix has also been tested on Thunderbird,
- Evolution, Pine, and Mutt. (Issue 9336, reported by marwick,
- patched by mutterc)
-
-2007-06-13 19:08 +0000 [r69128-69144] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Really ignore NULL frames and check whether
- the channel hungup or not. (issue #9912 reported by junky)
-
- * /, main/app.c: Merged revisions 69127 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2
- lines Return group counting to previous behavior where you could
- only have one group per category. (issue #9711 reported by
- irroot) ........
-
-2007-06-13 16:56 +0000 [r69016-69071] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Clarify a bit of logic. This doesn't change
- behavior in any way, but it is helpful when following the logic
- to debug problems like 9235.
-
- * channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct
- was accessed without the lock held. This issue was reported to me
- via email by Dmitry Mishchenko. Thanks!
-
- * cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois
- in #asterisk-bugs. PQclear() was not called on the result
- structure after doing a PQexec(). Also, fix up some formatting in
- passing.
-
-2007-06-12 19:36 +0000 [r69012-69014] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Change the full frame dropping log message
- to debug to avoid future bug reports.
-
- * channels/chan_iax2.c: Schedule the sending of a PING packet a
- second later than previously so that it does not collide with the
- LAGRQ.
-
-2007-06-12 19:13 +0000 [r69010] Russell Bryant <russell@digium.com>
-
- * main/channel.c: In ast_channel_make_compatible(), just return if
- the channels' read and write formats already match up. There are
- code paths that call this function on a pair of channels multiple
- times. This made calls fail that were using g729 in some cases.
- The reason is that codec_g729a will unregister itself from the
- list of available translators will all licenses are in use. So,
- the first time the function got called, the right translation
- path was allocated. However, the second time it got called, the
- code would not find a translation path to/from g729 and make the
- call fail, even if the channel actually already had a g729
- translation path allocated. (SPD-32)
-
-2007-06-12 14:23 +0000 [r68922] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 68921 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2
- lines Bring RTP back to Asterisk at the end of a native bridge no
- matter what. ........
-
-2007-06-11 21:20 +0000 [r68814] Jason Parker <jparker@digium.com>
-
- * include/asterisk/time.h: Solaris 10 sometimes (?) needs this
- include in order to have NULL defined.
-
-2007-06-11 20:45 +0000 [r68781] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly
- used
-
-2007-06-11 16:57 +0000 [r68733] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
- Merged revisions 68732 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) |
- 1 line added check for NULL Pointer when calling misdn_new.
- Asterisk does not allow us to create channels anymore when stop
- gracefully is used :). also modified the restart_indicator to 0
- ........
-
-2007-06-11 14:33 +0000 [r68683] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 68682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2
- lines Improve deadlock handling of the channel list. (issue #8376
- reported by one47) ........
-
-2007-06-11 10:29 +0000 [r68644] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /, channels/misdn/ie.c,
- channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) |
- 1 line fixed problem that the dummybc chanels had no lock,
- checking for the lock now. Also fixed the channel restart stuff,
- we can now specify and restart particular channels too. ........
-
-2007-06-11 04:21 +0000 [r68595] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * pbx/pbx_config.c: "dialplan save" produced garbage in the config
- file
-
-2007-06-08 22:23 +0000 [r68527] Russell Bryant <russell@digium.com>
-
- * /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) |
- 4 lines Don't automatically hang up after running Dictate so that
- callers can exit cleanly using '#' (closes issue #9577, patch
- from Thomas Andrews) ........
-
-2007-06-08 15:52 +0000 [r68450] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: actually remember the type/subclass of full
- frames that are in process
-
-2007-06-08 00:17 +0000 [r68370-68401] Joshua Colp <jcolp@digium.com>
-
- * /, main/say.c: Merged revisions 68397 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2
- lines Don't call ast_waitstream_full when the control file
- descriptor and audio file descriptor are not set, simply call
- ast_waitstream! (issue #8530 reported by rickead2000) ........
-
- * main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2
- lines Do a DNS lookup immediately upon calling the dnsmgr
- function, don't wait until a refresh happens. (issue #9097
- reported by plack) ........
-
-2007-06-07 23:14 +0000 [r68354] Russell Bryant <russell@digium.com>
-
- * /, main/say.c: Merged revisions 68351 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) |
- 3 lines Fix a problem where saying a character wouldn't properly
- break out when the caller pressed '#' (issue #8113, reported by
- patbaker82, patch from jamesgolovich (hey, long time no see!) and
- patbaker82) ........
-
-2007-06-07 23:00 +0000 [r68326] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Fix incorrect French syntax of "old
- messages". Request for feedback was sent to asterisk-dev mailing
- list, with little response. Issue 9118, patch by junky.
-
-2007-06-07 22:14 +0000 [r68313] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: some improvements to the IAX2 full frame
- dropping logic recently added: - use inaddrcmp(), since we have
- it - output the type of frame and subclass being dropped, and the
- type/subclass that is already being processed (which caused the
- drop)
-
-2007-06-07 21:16 +0000 [r68280] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c, apps/app_queue.c: Fix loading persistent
- queue members when using realtime configuration for queues. Also,
- remove an unneeded leading slash for the astdb family. (issue
- #9911, patch by atis)
-
-2007-06-07 20:25 +0000 [r68211-68249] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix an issue with newer phones which
- require packets be padded out to the correct length. Issue 9887,
- patch by DEA.
-
- * apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4
- lines Don't try to save voicemail greetings unless the user
- presses '1' to accept/save. Issue 9904, patch by me. ........
-
-2007-06-07 19:47 +0000 [r68198] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a
- check to make sure that greetings get stored properly. (Issue
- 8016, reported by edhorton, patched by alamantia with
- modification by me. Thanks to Jason Parker for the advice on
- this).
-
-2007-06-07 19:46 +0000 [r68196] Olle Johansson <oej@edvina.net>
-
- * channels/chan_features.c: Disable chan_features by default in
- menuselect
-
-2007-06-07 19:30 +0000 [r68192] Russell Bryant <russell@digium.com>
-
- * main/strcompat.c: Include stdarg.h for build issues on Solaris
- (issue #9381)
-
-2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Fix logic when doing a name based channel search
- for a structure when you want to start from a specific point in
- the channel list. (issue #9324 reported by slavon)
-
- * apps/app_dial.c, /: Merged revisions 68070 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2
- lines Allow the 'g' option to work if used with the 'S' option.
- (issue #9888 reported by gasparz) ........
-
-2007-06-07 10:00 +0000 [r67993-68030] Olle Johansson <oej@edvina.net>
-
- * res/res_jabber.c: Adding a few Todo's to res_jabber so we don't
- forget.
-
- * res/res_jabber.c: Ok, we found out that this is not about if you
- have any *active* clients using TLS, but if you have initialized
- TLS at all during the lifetime of the module. So if you reload to
- disable TLS, it won't help.
-
- * res/res_jabber.c: If you have a jabber client that uses TLS,
- refuse unload. Bad fix, but will prevent crashes while we are
- trying to find a workaround. Iksemel development seems to have
- stalled and we might have to stop using the TCP/TLS connections
- in that library and use our own, which would scale better from a
- poll/select perspective I guess. It would also make it easier to
- migrate to OpenSSL and stop Asterisk from depending on both
- OpenSSL and GnuTLS.
-
- * include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make
- sure we can unload res_jabber. Patch by phsultan - thanks! Due to
- a bug in the iksemel library, this will not work if you are using
- GTLS in the connection. That's being investigated. If you figure
- out a way to handle that without us having to patch iksemel, let
- us know in the bug report. Thanks.
-
-2007-06-07 00:10 +0000 [r67924-67941] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2
- lines Only notify the devicestate system of a peer state change
- when the peer is built from the config file. (issue #9900
- reported by arkadia) ........
-
- * main/file.c: Properly handle cases where a stream can't be
- written to. (issue #9757 reported by junky)
-
-2007-06-06 22:08 +0000 [r67862-67872] Russell Bryant <russell@digium.com>
-
- * res/res_snmp.c: Disable reload functionality in res_snmp. It is
- not possible to initialize the snmp library more than once
- without completely unloading the module and loading it again.
- (issue #9571, reported by hristo, additional helpful debug
- information from festr, patch from me)
-
- * channels/chan_sip.c: Fix a crash when doing call pickups with SIP
- phones. The code unlocked the channel when it should not have.
- (issue #9652, reported by corruptor, fixed by me)
-
-2007-06-06 19:26 +0000 [r67804] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix for Issue 9810. There was a segfault
- under a specific set of circumstances: 1. VoiceMailMain was
- configured in the dialplan with an extension as its argument 2. A
- message was left for this mailbox 3. Tried to call VoiceMailMain
- but hung up before entering password. This was fixed by checking
- that a pointer was non-null prior to trying to dereference it.
- (Issue 9810, reported by xmarksthespot, patched by Corydon76 with
- modifications by me).
-
-2007-06-06 16:55 +0000 [r67716] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /, include/asterisk/linkedlists.h: Merged
- revisions 67715 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) |
- 5 lines We have some bug reports showing crashes due to a double
- free of a channel. Add a sanity check to ast_channel_free() to
- make sure we don't go on trying to free a channel that wasn't
- found in the channel list. (issue #8850, and others...) ........
-
-2007-06-06 13:30 +0000 [r67594-67650] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 67649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2
- lines Reinvite the RTP back to the Asterisk machine when the
- timeout happens. (issue #9888 reported by gasparz) ........
-
- * main/translate.c: Fix plc_samples warning when registering a
- translator. (issue #9897 reported by xylome)
-
- * apps/app_directed_pickup.c: Include macroexten while searching
- for a channel to pick up in case they are in a macro. (issue
- #9491 reported by jamesb63)
-
- * res/res_agi.c: Make the new "agi debug off" CLI command work.
- (issue #9890 reported by eliel)
-
- * /, main/devicestate.c: Merged revisions 67593 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2
- lines Revert channel name splitting fix for Zap. The moral of the
- story is don't use - in your user/peer names. (issue #9668
- reported by stevedavies) ........
-
-2007-06-05 23:01 +0000 [r67558] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Fix some crashes related to the use of the
- "meetme" CLI command. The code for this command was not locking
- the conference list at all. (issue #9351, reported by and patch
- submitted by Junk-Y, committed patch is different and by me)
-
-2007-06-05 21:30 +0000 [r67526] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug
- 9883, wherein macros were not allowing the includes construct.
- fixed and tested, looks OK. Now includes can serve as an adjunct
- to catch.
-
-2007-06-05 20:53 +0000 [r67457-67492] Russell Bryant <russell@digium.com>
-
- * include/asterisk/linkedlists.h: This bug has been hanging over my
- head ever since I wrote this SLA code. Every time I tried to go
- debug it by adding some debug output, the behavior would change.
- It turns out I wasn't crazy. I had the following piece of code:
- if (remove) AST_LIST_REMOVE_CURRENT(...); Well,
- AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my
- conditional statement didn't do much good at all. It always ran
- at least all of the macro minus the first statement, so I was
- seeing list entries magically disappear when they weren't
- supposed to. After many hours of debugging, I have come to this
- extremely irritating fix. :) (issues #9581, #9497)
-
- * channels/chan_zap.c: Suppress a bunch of debug output unless
- option_debug is on
-
-2007-06-05 18:32 +0000 [r67424] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails
- saved to IMAP storage using extensions other than gsm were unable
- to be played over the phone. (Issue 9786, reporter:
- xmarksthespot, Patched by xmarksthe spot with revisions by me,
- reviewed by Russell Bryant).
-
-2007-06-05 18:18 +0000 [r67421] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Correctly update date/time on devices
- throughout the life of the device, instead of just at
- registration. Issue 9152, yet another patch by DEA.
-
-2007-06-05 18:17 +0000 [r67420] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: Added code to automatically add a default case to
- switches that don't have one. In some cases, rather than fall
- thru, it results in a goto with -1 result, which terminates the
- extension; a sort of dialplan seqfault, sort of. This was
- required to fix bug reported in 9881
-
-2007-06-05 17:07 +0000 [r67360-67372] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Handle a failure in malloc() in
- ast_safe_string_alloc()
-
- * main/channel.c: Fix a problem that showed itself by causing Zap
- channel names to be completely bogus on my machine.
- ast_safe_string_alloc() was broken. It called vsnprintf() on a
- va_args list twice without re-initializing it. After the first
- usage, va_end() and va_start() must be called again.
-
-2007-06-05 16:14 +0000 [r67329-67334] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) |
- 1 line briding is a bool, fixed copy and paste issue. ........
-
- * channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05
- Jun 2007) | 1 line simplified the EVENT_SETUP handling in the
- cb_events function a lot. Commented the different possibilities a
- bit and made functions of shared code. When the dialed extension
- does not exist in the extensions.conf we'll jump into the 'i'
- extension if this does exist, else we disconnect the call with
- the cause:1 = No Route to Destination. ........
-
-2007-06-05 15:51 +0000 [r67308] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, main/loader.c, include/asterisk/module.h: When
- shutting down "gracefully", go through and run the unload()
- callbacks for all of the modules. "stop now" is considered a
- non-graceful shutdown and will not go through this process.
- (issue #9804, reported by chrisost, patch by me)
-
-2007-06-05 15:22 +0000 [r67304] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Only muck with the thread structure if an
- idle one was found/created.
-
-2007-06-05 14:35 +0000 [r67270] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: ensure that a burst of full frames
- (AST_FRAME_DTMF being the prime example) will not be processed
- out of order... this is a brute force fix, but seems to be the
- safest fix for now (thanks to the Digium PQ department for
- finding this bug)
-
-2007-06-05 10:25 +0000 [r67210] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn_config.c, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h: Merged revisions 67209 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) |
- 1 line added possibility to deactivate bridging per port ........
-
-2007-06-04 23:43 +0000 [r67162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, funcs/func_math.c: Merged revisions 67161 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007)
- | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops.
- ........
-
-2007-06-04 23:31 +0000 [r67158] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt
- structure can disappear and the code did not account for it and
- crashes. (issues #9642, #9569, #9666, probably others ... based
- on the work by stevedavies and mihai, with additional changes
- from me)
-
-2007-06-04 23:26 +0000 [r67121-67156] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix for skinny keepalives. If there is no
- traffic from the phone for (keep_alive * 1100) ms (arbitrarily
- adding 10% for network issues, etc), unregister the device. Issue
- 8394, patch by DEA.
-
- * channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar
- to the recent fix for chan_skinny) Issue 9855, patch by DEA.
-
-2007-06-04 22:28 +0000 [r67119] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Add comments for two functions that get
- called with the appropriate call locked, but perform operations
- that could result in the pvt structure getting destroyed before
- returning again, causing numerous seg faults all over the module.
- (inspired by issues #9642, #9569, and #9666, and the work done by
- stevedavies and mihai)
-
-2007-06-04 21:59 +0000 [r67073] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: This typo has been here since 1.4 forked. It has been
- the source of heartburn to many a dialplan/CDR programmer.
-
-2007-06-04 21:47 +0000 [r67071] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC)
-
-2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Better handle SIP devices that say they have
- SDP content... but really don't. (issue #9398 reported by
- mthomasslo)
-
- * apps/app_dial.c: Initialize cidname variable to nothing since it
- may be used without having been touched. (issue #9661 reported by
- dimas)
-
- * res/res_features.c: Returning a value that indicates the parking
- of a call was a success when it really wasn't (because the
- parking slot selected was in use) is the wrong thing to do.
- (issue #9723 reported by mdu113)
-
-2007-06-04 17:11 +0000 [r67061] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk, /,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.mandrake.zaptel,
- contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007)
- | 2 lines Add revision Id tags (by request of tzafrir) ........
-
-2007-06-04 16:02 +0000 [r67026] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Change the configure script to build a
- test program against libcurl to make sure the results from
- curl-config can be used to compile successfully. This is intended
- to help prevent a situation where you are cross compiling, and
- the configure script finds the curl library installed on the
- host. (issue #9865, reported and patched by zandbelt)
-
-2007-06-04 15:50 +0000 [r67021] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash
-
-2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When
- handling an implicit ACK to a frame that was marked as the final
- transmission for a call, don't call iax2_destroy() for that call
- while the global frame queue is still locked. There is a very
- nice explanation of the deadlock in the report. (issue #9663,
- thorough report and patch from stevedavies, additional positive
- test reports from mihai and joff_oconnell)
-
- * include/asterisk/stringfields.h: Fix some compiler warnings in
- C++ modules. (issue #9866, reported by osk, patch by Corydon76)
-
-2007-06-01 21:45 +0000 [r66919] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_odbc.c: On some drivers, deallocating the statement
- handle isn't enough. We also have to clear the cursor (nice,
- Oracle)
-
-2007-06-01 21:31 +0000 [r66897-66917] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing extraneous debugging lines from
- revision 66897. Sorry :)
-
- * apps/app_voicemail.c: Submitting a fix for voicemail with IMAP
- storage. Attachments with format specified as gsm were duplicated
- (i.e. two attachments) were left. Thank you very much to
- xmarksthespot for submitting the patch that fixed this. (Issues
- 9787 and 8873, Reported by xmarksthespot and jerjer, patched by
- xmarksthespot)
-
-2007-06-01 19:41 +0000 [r66879-66881] Russell Bryant <russell@digium.com>
-
- * channels/chan_skinny.c: Changes to the way DTMF is handled in the
- core broke dialing in chan_skinny. This patch makes chan_skinny
- usable again. I did not end up testing this, but there are
- multiple positive test reports listed in the bug report. (issue
- #9596, reported by pj, testing by pj and mvanbaak, and the fix
- was written by DEA)
-
- * apps/app_page.c: List app_meetme as a module that app_page
- depends on.
-
-2007-05-31 23:03 +0000 [r66821] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * doc/asterisk.8: Issue 9850 - update preferred command line syntax
-
-2007-05-31 18:41 +0000 [r66775] Russell Bryant <russell@digium.com>
-
- * res/res_speech.c, include/asterisk/app.h,
- include/asterisk/speech.h: Change a couple of header files to not
- use "new", which is a reserved keyword in C++. (issue #9830,
- reported by osk)
-
-2007-05-31 17:15 +0000 [r66770] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_macro.c: Merged revisions 66744 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007)
- | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime.
- Issue 8329 will remain unfixed for pbx_realtime, but only because
- we lack core API to do it. ........
-
-2007-05-31 16:14 +0000 [r66768] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2
- lines It is now possible for this path of execution to have the
- frame pointer be NULL, therefore we need to check for it before
- trying to access it. (issue #9836 reported by barthpbx) ........
-
-2007-05-30 23:26 +0000 [r66671] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixed seg-faults when recording greetings
- in voicemail with IMAP enabled. (Issue No. 9735, reported by
- xmarksthespot, patched by me)
-
-2007-05-30 17:28 +0000 [r66602-66639] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Silly me for having out of date source! Oh
- well... I'm still leaving my comment.
-
- * channels/chan_sip.c: When calling some peer/host that may not
- exist/reply back... don't keep the dialog in memory for all of
- eternity.
-
- * channels/chan_zap.c, channels/chan_features.c: Change how channel
- names are generated a bit. (issue #9825 reported by eldadran)
-
-2007-05-29 21:56 +0000 [r66538] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007)
- | 2 lines If the value of a variable passed to FIELDQTY is blank,
- then FIELDQTY should return 0, not 1. ........
-
-2007-05-29 19:32 +0000 [r66474-66503] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Properly handle 408 request timeout -
- according to the RFC, the dialog dies if a request in a dialog
- gets this response.
-
- * channels/chan_sip.c: Don't issue hangup on hangup on hangup on
- hangup (for jcmoore)
-
-2007-05-29 16:44 +0000 [r66437] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Handle cases where a frame may have no data. (issue
- #9519 reported by dmb)
-
-2007-05-29 16:07 +0000 [r66404-66414] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't reset hangupcause if we already have
- one
-
- * channels/chan_sip.c: Tracking down hanging channels, killing them
- one by one. Issue #9235 and related
-
-2007-05-29 15:43 +0000 [r66398] Joshua Colp <jcolp@digium.com>
-
- * doc/datastores.txt: Update datastores documentation. (issue #9801
- reported by mnicholson)
-
-2007-05-29 09:41 +0000 [r66363] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2
- lines Issue #9802 - Change inuse counter on CANCEL ........
-
-2007-05-28 23:16 +0000 [r66312] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c: Make the usedistinctiveringdetection option
- work again. (issue #9823 reported by premeau)
-
-2007-05-27 04:12 +0000 [r66244] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: I don't know what this was trying to do, but
- it's clearly incorrect. Issues 9808 and 9809.
-
-2007-05-25 14:43 +0000 [r66160] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac: have to check for OSP toolkit _after_
- checking for OpenSSL
-
-2007-05-25 14:41 +0000 [r66159] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, main/say.c: Merged revisions 66127 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007)
- | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch
- ........
-
-2007-05-25 14:28 +0000 [r66157] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac, channels/chan_gtalk.c, makeopts.in,
- res/res_jabber.c: handle the GNUTLS library properly in the
- configure script and build system don't build in OSP support
- unless we have found and are allowed to use SSL support
-
-2007-05-24 22:23 +0000 [r66076] Russell Bryant <russell@digium.com>
-
- * main/channel.c: if the string field init fails, clean up the
- stuff that was allocated already
-
-2007-05-24 22:16 +0000 [r66074] Joshua Colp <jcolp@digium.com>
-
- * main/slinfactory.c: Fix slinfactory logic when dealing with
- frames coming in that may already be in the signed linear format.
-
-2007-05-24 22:07 +0000 [r66068-66070] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Check the result of ast_string_field_init() in
- ast_channel_alloc()
-
- * main/rtp.c: Make 1.4 build on my machine, too..
-
-2007-05-24 20:54 +0000 [r66029-66030] Jason Parker <jparker@digium.com>
-
- * configure: Rebuild configure script for previous ar fix.
-
- * configure.ac: Following moving strip to AC_PATH_TOOL, we need to
- do something similar for ar.
-
-2007-05-24 20:42 +0000 [r65978-66026] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- Checking for the strip application needs to be done with
- AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross
- compilation environments.
-
- * Makefile: Clear CFLAGS before running make for menuselect. (issue
- #9784, reported by ovi, patch by me)
-
-2007-05-24 18:28 +0000 [r65965-65967] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_gtalk.c: oops, use #ifdef instead of #if
-
- * channels/chan_gtalk.c: don't reference GnuTLS headers and
- functions unless the configure script found it
-
- * main/rtp.c: don't use uninitialized variables
-
-2007-05-24 15:27 +0000 [r65902] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Add the ability to blacklist certain commands
- from being executed using the Command AMI action. (issue #9240
- reported by junky)
-
-2007-05-24 15:26 +0000 [r65892-65901] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk
- core dump since the GnuTLS interface did not support
- multithreading correctly.
-
- * channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN.
- Patch by phsultan. Thanks!
-
-2007-05-24 15:16 +0000 [r65877-65883] Jason Parker <jparker@digium.com>
-
- * .cleancount: Update cleancount for that last commit - just for
- good measure.
-
- * include/asterisk/translate.h, codecs/codec_speex.c,
- main/translate.c, codecs/codec_ilbc.c: Fix handling of
- zero-length frames when a codec is capable of native PLC. Issue
- 9183, patch by Mihai.
-
-2007-05-24 15:08 +0000 [r65866] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * funcs/func_math.c: merged qwell's func_math patch for issue 9507
-
-2007-05-24 15:08 +0000 [r65863] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: I like it when the RTP stack compiles myself...
-
-2007-05-24 15:05 +0000 [r65857] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues
- when calling from gtalk to SIP over nat.
-
-2007-05-24 15:04 +0000 [r65842-65853] Russell Bryant <russell@digium.com>
-
- * apps/app_festival.c: Ensure that frames are fully initialized.
- This will probably fix getting weird timestamp log messages in
- logs when using the Festival app. (issue #9781, patch by me)
-
- * main/rtp.c: Fix the calculation of the RTT for RTCP. The previous
- code would result in oscillating and incorrect data.
- Additionally, the RTT would sometimes report negative values due
- to incorrect calculations. (issue #9601, patch from davetroy)
-
-2007-05-24 14:48 +0000 [r65841] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for
- jingle
-
-2007-05-24 14:42 +0000 [r65839] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2
- lines Allow RFC2833 to be negotiated when an INVITE comes in
- without SDP and is not matched to a user or peer. (issue #9546
- reported by mcrawford) ........
-
-2007-05-24 14:38 +0000 [r65836] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan -
- Fix "login" as component to jabber server. ...and, by accident,
- fix a bug in chan_sip for stopping a loop on retransmits of BYE
- requests.
-
-2007-05-24 09:37 +0000 [r65768] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24
- Mai 2007) | 1 line we should only activate the generator in
- chan_misdn, when asterisk hask not yet taken the call
- (WAITING4DIGS state). Alerting audio will be generated fomr
- asterisk for example. ........
-
-2007-05-23 20:59 +0000 [r65677-65685] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: start the delayed PBX when receive voice or
- video full frames as well, and comment this delayed-PBX activity
-
- * /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007)
- | 2 lines ensure that variables are set on a newly created
- channel before we start a PBX on it ........
-
- * channels/chan_iax2.c: clear the 'delay PBX' flag when we are
- ready to start the PBX
-
- * channels/chan_iax2.c: don't start a PBX on a new incoming IAX2
- channel until we have some sort of response to our ACCEPT (ACK or
- anything else)
-
- * /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007)
- | 2 lines if we are going to set variables on a newly created
- channel, it should be done *before* we start the PBX on it
- ........
-
-2007-05-23 13:07 +0000 [r65589] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) |
- 3 lines Revert revision 62417 as someone reported problems with
- it to Mark. This was related to issue #9588. ........
-
-2007-05-22 20:25 +0000 [r65541] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/make_version: when building a version string for a
- developer branch, include the base branch in the version string
-
-2007-05-22 18:40 +0000 [r65501] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a
- dependency for app_voicemail and chan_zap (Thanks to mnicholson
- for pointing it out)
-
-2007-05-22 15:04 +0000 [r65452] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Remove a double const.
-
-2007-05-22 14:02 +0000 [r65408] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_followme.c: Fix a problem with flag recognition.
-
-2007-05-22 13:09 +0000 [r65394] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 65389 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) |
- 4 lines Fix a memory leak that I just noticed in the device state
- handling in app_queue. On most device state changes, it would
- leak roughly 8 to 64 bytes (the length of the name of the
- device). ........
-
-2007-05-22 08:12 +0000 [r65342] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22
- Mai 2007) | 1 line we stop the tones only when we're in the
- pre-call phase, otherwise e.g. when in CONNECTED state we should
- not stop tones when we receive an Information Message ........
-
-2007-05-20 17:59 +0000 [r65250] Joshua Colp <jcolp@digium.com>
-
- * res/res_agi.c: res_agi needs to export two symbols
- (ast_agi_register and ast_agi_unregister) for usage by others.
- (issue #9755 reported by mnicholson)
-
-2007-05-18 22:26 +0000 [r65200-65201] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c
- to main/cdr.c, and neither did I. This is the remainder of the
- 9717 patch, the fix for the run-away FAIL status for a call
-
- * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions
- 65172 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1
- line This update will fix the situation that occurs as described
- by 9717, where when several targets are specified for a dial, if
- any one them reports FAIL, the whole call gets FAIL, even though
- others were ringing OK. I rearranged the priorities, so that a
- new disposition, NULL, is at the lowest level, and the
- disposition get init'd to NULL. Then, next up is FAIL, and next
- up is BUSY, then NOANSWER, then ANSWERED. All the related set
- routines will only do so if the disposition value to be set to is
- greater than what's already there. This gives the intended
- effect. So, if all the targets are busy, you'd get BUSY for the
- call disposition. If all get BUSY, but one, and that one rings is
- not answered, you get NOANSWER. If by some freak of nature, the
- NULL value doesn't get overridden, then the disp2str routine will
- report NOANSWER as before. ........
-
-2007-05-18 18:16 +0000 [r65041-65123] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2
- lines Not getting an ACK to a 200 OK in the initial invite is
- critical to the call. ........
-
- * /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5
- lines Issue 9235 - part of the problem, maybe not all. Please
- retry with this patch (and no other patch) if you have problems
- with hanging SIP channels. Thank you. A special Thank You to
- WeBRainstorm that gave me access to his system. ........
-
- * channels/chan_sip.c: - Adding support for putting calls OFF hold
- with a re-invite with blank SDP. This was a bug found while doing
- tests at SIPit in Antwerp. - In order to not duplicate code, I
- restructured some of the code for putting calls on/off hold.
- Thanks DEA for reminding me. This fix has been asleep in the
- videocaps branch until now.
-
-2007-05-18 12:40 +0000 [r65039] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
- revisions 65007 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) |
- 1 line fixed a warning regarding Keypad encoding. encode the IE
- sending_complete at the right position. ........
-
-2007-05-18 10:37 +0000 [r64974] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue 9487 - stop media flows at hangup of
- call
-
-2007-05-18 08:58 +0000 [r64904] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18
- Mai 2007) | 1 line we *need* to send a PROCEEDING when
- sending_complete is set, even if need_more_infos is requested.
- ........
-
-2007-05-18 02:48 +0000 [r64868] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix a small bug I noticed while working on
- something else. app_queue did not unregister its device state
- monitoring callback in unload_module(). So, this would make
- Asterisk crash on the first device state change after you unload
- the module.
-
-2007-05-17 21:19 +0000 [r64820] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, include/asterisk/linkedlists.h: Merged revisions 64819 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007)
- | 2 lines How is it that we never caught that this is returning
- the opposite of our documentation, until now? ........
-
-2007-05-17 16:53 +0000 [r64761] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4
- lines If we have a negative current message, we shouldn't go back
- even further... Issue 9727. ........
-
-2007-05-17 16:52 +0000 [r64756-64759] Russell Bryant <russell@digium.com>
-
- * contrib/scripts/astxs (removed): Remove script that is no longer
- functional since the build system was redone. (issue #9340,
- reported by junky)
-
- * apps/app_dial.c: Increase the size of a buffer to support longer
- dial strings for channels. (issue #9291, reported and fix
- suggested by meni)
-
-2007-05-17 16:10 +0000 [r64720-64754] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Even more direct RTP setup fixes! Don't
- allow a codec that isn't supported to creep into the SDP of
- either side. (issue #9446 reported by marcelbarbulescu)
-
- * apps/app_voicemail.c: Fix authuser support. (issue #9740 reported
- by xmarksthespot)
-
-2007-05-17 06:13 +0000 [r64686] Russell Bryant <russell@digium.com>
-
- * README: Update the main README to reflect the new build process
- for 1.4 and above. (issue #9725, patch by eliel)
-
-2007-05-16 11:01 +0000 [r64516-64609] Olle Johansson <oej@edvina.net>
-
- * /: Blocking patch already in this code
-
- * channels/chan_sip.c: Fix auth on BYE. (Different patch than for
- 1.2)
-
- * channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE
-
- * channels/chan_sip.c: Final part of issue #9483 - fixing
- transfer() of sip calls in the dial plan (twilson)
-
- * channels/chan_sip.c: Issue #9439 - properly handle username
- parameters in SIP uri.
-
- * /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2
- lines Support SIP uri's starting with SIP: and sip: (reported by
- Tony Mountfield on the mailing list. Thanks!) ........
-
- * /, channels/chan_sip.c: Merged following patch with a lot of
- changes for 1.4 ------ Merged revisions 64514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6
- lines Issue #9726 - rlister - Better logging for ACL denials
- While at it, also added better logging and handling of peers that
- are not supposed to register. My patch, stole the issue report
- from Russell. My apologies, Russell :-) ........
-
-2007-05-16 08:44 +0000 [r64515] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16
- Mai 2007) | 1 line in the case immediate=yes, we directly jump
- into the dialplan, where people can use PlayTones to indicate a
- Dialtone, so we don't need to to that by ourself. also we should
- not do a dialtone_indicate for incoming calls on a TE port in
- overlapdialmode. ........
-
-2007-05-15 19:52 +0000 [r64353-64426] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Properly fix a problem that occurs when you
- set PARKINGEXTEN to an exten where a call is already parked.
- (issue #9723, patch by me)
-
- * res/res_features.c: When someone requests a specific parking
- space using the PARKINGEXTEN variable, ensure that no other
- caller is already there. (issue #9723, reported by mdu113, patch
- by me)
-
-2007-05-14 19:26 +0000 [r64324] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit
- more
-
-2007-05-14 19:13 +0000 [r64306] Russell Bryant <russell@digium.com>
-
- * channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by
- just ignoring it. An unknown indication will trigger an error and
- cause sounds to stop, which in this case, is ringing.
-
-2007-05-14 18:52 +0000 [r64280] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Handle network errors, like host or network
- unreachable, in a better way. This means that calls to hosts or
- qualify (OPTION) messages will fail quicker if the TCP/IP stack
- tells us that there is an issue. Since this is an unconnected UDP
- socket, we will not get error messages directly in most cases,
- but maybe on the second and third try. This is already
- implemented in trunk.
-
-2007-05-14 18:48 +0000 [r64240-64278] Joshua Colp <jcolp@digium.com>
-
- * codecs/codec_speex.c: Properly set datalen field when doing PLC
- in codec_speex. (issue #9722 reported by mihai)
-
- * /, main/devicestate.c: Merged revisions 64275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2
- lines Only perform stripping of - strings from the channel name
- for Zap channels. Anywhere else we might remove a legitimate part
- of a device name. (issue #9668 reported by stevedavies) ........
-
- * main/channel.c: Fix scenario where if a phone that simply called
- Echo() put itself on hold it could never get off hold.
-
-2007-05-14 13:58 +0000 [r64193] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570,
- worrisome CDR warnings have been removed, that are either not
- helpful, or not relevant.
-
-2007-05-14 10:39 +0000 [r64157] Olle Johansson <oej@edvina.net>
-
- * main/channel.c: Add hangupcause when we lack codecs for
- transcoding
-
-2007-05-12 22:27 +0000 [r64044-64114] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: This concludes my final adventure with
- bitmasks and the onhold flag. Would anyone care for some peanuts?
-
- * channels/chan_sip.c: Tweak hold flags some more. They can be of
- three states when active: active, inactive, one direction.
-
- * channels/chan_sip.c: Ensure the onhold flag is set no matter what
- when being put on hold.
-
-2007-05-11 20:16 +0000 [r63982] Jason Parker <jparker@digium.com>
-
- * main/manager.c: Hide manager password from "manager show user
- foo". I realize that there are other ways to get this, but we
- really don't need to just show it in plain text so easily. Issue
- 9273, patch by junky
-
-2007-05-11 16:35 +0000 [r63905] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * contrib/scripts/safe_asterisk, Makefile, /: Merged revisions
- 63903 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007)
- | 2 lines Issue 9121 - fixups for safe_asterisk script ........
-
-2007-05-11 16:05 +0000 [r63886] Russell Bryant <russell@digium.com>
-
- * main/manager.c: When MD5 authentication is not possible because
- there is no challenge present, either because the Challenge
- action was never issued, or some other reason, give a proper
- error message and return an error instead of claiming that the
- user wasn't found. (reported by jsmith on IRC)
-
-2007-05-11 15:43 +0000 [r63872] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: Make the PARKINGEXTEN feature of parking
- actually work. (issue #9708 reported by mdu113)
-
-2007-05-10 23:15 +0000 [r63830] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4
- lines Fix an issue with trying to kill a thread before it gets
- created. Issue 9709, patch by nic_bellamy. ........
-
-2007-05-10 22:23 +0000 [r63804] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Strip terminal escape sequences from CLI command
- output that is going to be sent out over the manager interface.
- (issue #9659, reported by pari, fixed by me)
-
-2007-05-10 20:48 +0000 [r63750] Doug Bailey <dbailey@digium.com>
-
- * main/callerid.c: Add test for negative offsets in cid data to
- prevent infinite loops.
-
-2007-05-10 20:46 +0000 [r63749] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4
- lines Do not allocate SIP pvt's for PEERs we can not reach. This
- was seen as a lot of dialogs being created then immediately
- destroyed at reload/restart of the SIP channel. ........
-
-2007-05-09 19:22 +0000 [r63656-63698] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Use the DTMF frame on the channel when returning
- a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE.
-
- * channels/chan_sip.c: Do not prematurely go on hold if sendonly
- was not actually set.
-
-2007-05-09 17:25 +0000 [r63654] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2
- lines Make sure we only create a DSP if it's requested on
- SUB_REAL ........
-
-2007-05-09 16:55 +0000 [r63612] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Modify ast_senddigit_begin() to use the same
- assumptions used elsewhere in the code in that if a channel does
- not have a send_digit_begin() callback, it only cares about DTMF
- END events. (pointed out by Michael Neuhauser on the asterisk-dev
- list)
-
-2007-05-09 16:54 +0000 [r63611] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2
- lines Properly handle hints that point to multiple devices in
- chan_sip. Why chan_sip is even doing this I have no idea but I
- would rather not go into a rant. (issue #9536 reported by
- rlister) ........
-
-2007-05-09 16:43 +0000 [r63608] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Only call ast_senddigit_begin() in
- ast_senddigit() if the channel has a send_digit_begin() callback.
- Checking the END_DTMF_ONLY flag was the wrong thing to do,
- because that flag indicates that a *bridged* channel only wants
- DTMF END events coming from this channel.
-
-2007-05-09 14:50 +0000 [r63566] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_directory.c: Merged revisions 63565 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007)
- | 2 lines Replicate fix from 51158 (app_voicemail) to
- app_directory (Issue 9224) ........
-
-2007-05-09 13:24 +0000 [r63535] Russell Bryant <russell@digium.com>
-
- * Makefile: I have seen multiple people post questions trying to
- figure out what the message "The configure script must be
- executed before running 'make'" means. So, add another like that
- says to specifically run ./configure. If this isn't obvious
- enough, then they should be using something like AsteriskNOW and
- not installing from source.
-
-2007-05-09 13:17 +0000 [r63534] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
- channels/misdn/isdn_msg_parser.c: Merged revisions
- 62945,63402,63519 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) |
- 1 line when we're in state WAITING4DIGS, we use the asterisk
- tone-generator which prods us, so we can't just return -1 in
- misdn_write in this case. Added a MISDN_KEYPAD channel variable,
- and fixed the sending of keypad. this enables us to modify the
- call forward parameters in the switch. ........ r63402 | crichter
- | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added
- application misdn_check_l2l1 which tries to pull up the L1/L2 on
- all ports that have the layers down in a group. It waits then for
- a timeout. This helps for scenarios where multiple PMP BRIs are
- grouped together, or where a provider has a faulty PTP
- Implementation, that looses the L2 after a while. ........ r63519
- | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
- release_chan frees ch, so we should never touch ch after
- release_chan, this may cause segfaults. ........
-
-2007-05-09 13:04 +0000 [r63532] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't retransmit 200 OK's on ignore status.
- (Reported on asterisk-users)
-
-2007-05-08 22:38 +0000 [r63478] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_macro.c: Merged revisions 63477 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007)
- | 2 lines Issue 9602 - segfault in app_macro ........
-
-2007-05-08 16:53 +0000 [r63403-63448] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: I mixed up the use of the find_feature()
- function, so I renamed it find_dynamic_feature, and changed the
- code to use the correct lock when using it.
-
- * res/res_features.c: Use a read/write lock when accessing the
- built-in features.
-
- * contrib/scripts/realtime_pgsql.sql (added),
- contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to
- contrib/scripts to be with the rest of the sql examples. (issue
- #9676, suretec)
-
-2007-05-08 06:22 +0000 [r63360] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007)
- | 2 lines Issue 9527 - upon entering a folder, no message is
- selected (curmsg == -1), so deleting causes memory corruption
- (beyond bounds) ........
-
-2007-05-07 22:28 +0000 [r63329] Russell Bryant <russell@digium.com>
-
- * configs/res_pgsql.conf.sample (added),
- configs/extconfig.conf.sample, contrib/realtime_pgsql.sql
- (added): Add a sample configuration file and example tables for
- use with res_config_pgsql. (issue #9676, suretec)
-
-2007-05-07 21:45 +0000 [r63283-63286] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged
- revisions 63285 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2
- lines Properly handle what happens during a masquerade in
- relation to group counting. (issue #9657 reported by ramonpeek)
- ........
-
- * channels/chan_sip.c: Minor backport of revision 59083 in trunk.
- Don't queue an unhold frame up if the call was never on hold to
- begin with.
-
-2007-05-07 20:05 +0000 [r63196-63254] Olle Johansson <oej@edvina.net>
-
- * main/config.c: Don't remove configuration from memory just
- because one section failed.
-
- * /: Guess svnmerge doesn't handle files that move around. Blocking
- patch to ./config.c
-
-2007-05-06 12:28 +0000 [r63152] Olle Johansson <oej@edvina.net>
-
- * main/file.c: Stop the video stream when you stop playback of all
- streams for a call
-
-2007-05-04 20:03 +0000 [r63099] Jason Parker <jparker@digium.com>
-
- * res/res_jabber.c: Fix a crash when checking version attribute in
- an incoming XML caps element. Issue 9667, patch by phsultan.
-
-2007-05-04 16:45 +0000 [r63047] Pari Nannapaneni <paripurnachand@digium.com>
-
- * configs/manager.conf.sample: explanation for httptimeout in
- manager.conf
-
-2007-05-03 16:44 +0000 [r62989] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2
- lines When a peer is seeded or built tell the devicestate core to
- update it's status. This is easier then having chan_sip load
- before pbx_config. (issue #9658 reported by dlynes) ........
-
-2007-05-03 16:38 +0000 [r62986] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/loader.c: improve loader a bit, by avoiding trying to
- initialize embedded modules twice and avoiding trying to load
- modules from disk when they have been loaded already during the
- 'preload' pass (reported by blitzrage on IRC, patch by me)
-
-2007-05-03 15:23 +0000 [r62942] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell
- Bryant, 2007, TM, Patent Pending). This set of changes came from
- a debugging session I had with Dwayne Hubbard. When he called
- into his home FXO, ran the Echo application, and pressed a digit,
- the digit would be echoed back and would never end. This is
- fixed, along with a couple other little improvements. * When
- chan_zap is in the middle of playing a digit to a channel, it
- feeds back null frames, not voice frames. So, I have modified
- ast_read to check the timing on emulated DTMF when it receives
- null frames, in addition to where it was doing this on voice
- frames. * Make a tweak to setting the duration on emulated DTMF
- digits. If there was no duration specified, it set it to be the
- minimum, instead of the default. * Instead of timing the emulated
- digits off of the number of samples in audio frames that pass
- through, just use time values. Now there is no code in this
- section that assumes 8kHz audio.
-
-2007-05-03 14:41 +0000 [r62913] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19
- (added), pbx/ael/ael-test/ael-test18/extensions.ael,
- pbx/ael/ael-test/ael-test19/extensions.ael (added),
- pbx/ael/ael-test/ael-test19 (added),
- pbx/ael/ael-test/ref.ael-test20 (added),
- pbx/ael/ael-test/ael-test20/extensions.ael (added),
- pbx/ael/ael-test/ael-test20 (added): updated the ael regressions
- to match what's in trunk
-
-2007-05-03 14:36 +0000 [r62912] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
- channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
- revisions 61357,61770,62885 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) |
- 1 line some fixes for PMP Hold/Retrieve, it should work now, when
- briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200
- (Di, 24 Apr 2007) | 1 line added lock for sending messages to
- avoid double sending. shuffled some empty_chans after the
- cb_event calls, this avoids that a release_complete from a quite
- different call releases a fresh created setup by accident.
- ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03
- Mai 2007) | 1 line fixed the problem that misdn_write did not
- return -1 when called with 0 samples in a frame this resultet in
- a deadlock in some circumstances, when the call ended because of
- a busy extension. added encoding of keypad. ........
-
-2007-05-03 13:54 +0000 [r62883] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test18 (added),
- pbx/ael/ael-test/ref.ael-vtest13,
- pbx/ael/ael-test/ael-test18/extensions.ael (added),
- pbx/ael/ael-test/ael-test18 (added),
- pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c,
- pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7:
- These mods fix bug 9623, where an '@' in the eswitch contents
- causes a syntax error. I also updated the regressions.
-
-2007-05-03 00:23 +0000 [r62797-62842] Kevin P. Fleming <kpfleming@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02
- May 2007) | 2 lines doh... initializing the pointer variable will
- work just a bit better ........
-
- * res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02
- May 2007) | 7 lines increase reliability and efficiency of static
- Realtime config loading via ODBC: don't request fields we aren't
- going to use don't request sorting on fields that are pointless
- to sort on explicitly request the fields we want, because we
- can't expect the database to always return them in the order they
- were created (reported by blitzrage in person (!), patch by me)
- ........
-
- * res/res_config_pgsql.c: improve static Realtime config loading
- from PostgreSQL: don't request sorting on fields that are
- pointless to sort on use ast_build_string() instead of snprintf()
- don't request the list of fieldnames that resulted from the query
- when we both knew what they were before we ran the query _AND_ we
- aren't going to do anything with them anyway (patch by me,
- inspired by blitzrage's bug report about res_config_odbc)
-
-2007-05-02 22:59 +0000 [r62739-62789] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Merge changes from team/russell/inband_dtmf ...
- Fix some issues related to generating inband DTMF. There are two
- changes here: 1) The list of DTMF tones in the senddigit_begin()
- function explicitly specified 100ms of the tone followed by 100ms
- of silence. This really broke things with the way that Asterisk
- now wants complete control over when the digit begins and ends.
- So, regardless of what Asterisk really wanted to do, this was
- going to play out the tone at the length it wanted to. This
- caused various problems like DTMF translation to inband to be
- extremely unreliable. The list of tones has been changed so that
- the correct DTMF tone is played indefinitely until Asterisk tells
- it to stop. 2) ast_write() had to be modified to let a DTMF_END
- frame get processed even when a generator is present. This is how
- the tone will finally get stopped. (issues #8944, #9250, #9348,
- maybe others. Thanks to mdu113 from #8944 for the testing and
- feedback!)
-
- * main/manager.c: Backport the change that only went in to trunk
- that fixes the command manager action over http. (reported
- internally by pari and bkruse)
-
-2007-05-02 20:46 +0000 [r62738] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May
- 2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being
- in 'h' extension louses up the dst field ........
-
-2007-05-02 17:43 +0000 [r62692] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007)
- | 4 lines Issue 9638 - if a text frame is sent with no
- terminating NULL through a bridged IAX connection, the remote end
- will receive garbage characters tacked onto the end. ........
-
-2007-05-02 17:10 +0000 [r62689] Steve Murphy <murf@digium.com>
-
- * configs/extensions.conf.sample, main/channel.c, main/pbx.c,
- channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the
- clid, src fields in channel_alloc call. b)in the channel_alloc
- func, set the cid_num and name fields from the arglist[blush]. c)
- don't update the channel app & app data fields if you are in the
- 'h' extension. d)the load_module func in cdr_radius needs to
- return DECLINE, SUCCESS.
-
-2007-05-02 06:15 +0000 [r62624] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't unlock a channel that we already know
- does not exist (propably isue 8228)
-
-2007-05-01 21:57 +0000 [r62548] Russell Bryant <russell@digium.com>
-
- * /, res/res_features.c: Merged revisions 62547 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) |
- 4 lines Remove an unnecessary check that makes it so if you hang
- up after doing an attended transfer before the target extension
- answers the channel, the transfer is not successful. (issue
- #9338, patch by svanlund) ........
-
-2007-05-01 21:34 +0000 [r62545] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user()
- (found by rayjay, different fixes by me)
-
-2007-05-01 16:26 +0000 [r62497] Russell Bryant <russell@digium.com>
-
- * /, configs/indications.conf.sample: Merged revisions 62496 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) |
- 3 lines Add indications.conf information for the Philippines.
- (issue #9525, reported and patched by loloski) ........
-
-2007-04-30 15:58 +0000 [r62414-62419] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) |
- 4 lines This patch fixes an issue where depending on the cause
- code, when the network sends a PRI disconnect, the call may not
- be properly hung up. (issue #9588, reported and patched by
- softins) ........
-
- * include/asterisk/http.h, main/http.c: When serving dynamic
- content, include a Cache-Control header to instruct the browsers
- to not store the resulting content. (issue #9621, reported by
- Pari, patch by me)
-
-2007-04-30 14:52 +0000 [r62371] Jason Parker <jparker@digium.com>
-
- * configs/iax.conf.sample: Remove unused (and potentially
- confusing) jitterbuffer options from sample config.
-
-2007-04-30 14:36 +0000 [r62369] Joshua Colp <jcolp@digium.com>
-
- * main/asterisk.c, /: Merged revisions 62368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2
- lines Update copyright notice. It's now the year 2007! ........
-
-2007-04-29 05:50 +0000 [r62299-62331] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Fix a bug that made the "language" setting
- in zapata.conf not functional. (issue #9626, reported and fixed
- by sergee)
-
- * apps/app_meetme.c: Note that the "talker optimization" option
- will be enabled by default in 1.6
-
-2007-04-27 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.4 released.
-
-2007-04-27 21:10 +0000 [r62218] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c: Fix a weird problem where when a caller
- talking to someone sitting behind an agent channel sent a digit,
- the digit would be played to the agent for forever. This is
- because chan_agent always returned -1 from its send_digit_begin
- and _end callbacks. This non-zero return value indicates to the
- Asterisk core that it would like an inband DTMF generator put on
- the channel. However, this is the wrong thing to do. It should
- *always* return 0, instead. When the digit begin and end
- functions are called on the proxied channel, the underlying
- channel will indicate whether inband DTMF is needed or not, and
- the generator will be put on that one, and not the Agent channel.
- (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed
- by me)
-
-2007-04-27 16:17 +0000 [r62174] Jason Parker <jparker@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3
- lines This transcoder message needn't be a NOTICE. I've seen it
- cause confusion more than a few times. ........
-
-2007-04-27 16:14 +0000 [r62171] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: If no variables were passed into
- pbx_substitute_variables_helper_full(), then don't even bother
- creating a temporary bogus channel, since that is only for
- allowing certain functions to operate on the variables as if they
- were on a channel. Most importantly, this fixes a crash. (issue
- #9613, reported by callguy, fixed by me)
-
-2007-04-27 14:04 +0000 [r62095-62137] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4
- lines Issue #7351 - SIP Cancel fails due to the wrong contact
- uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka
- - THANKS!!!! THis was a hard one to catch. ........
-
- * channels/chan_zap.c, main/manager.c: Issue #9608 - fix some
- annoying DEBUG messages not controlled by option_debug (DEA).
- Thanks!
-
-2007-04-26 16:33 +0000 [r61959-62038] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2
- lines Revert previous fix for when the IAX2 channel goes funky
- (that's the technical term). This is causing legit calls to be
- prematurely hung up. (issue #9600 reported by justdave) ........
-
- * main/channel.c: Missed an ast_app_group_discard during merge.
- Thanks blitzrage!
-
- * res/res_monitor.c: Don't always say that the channel is being
- paused if it is actually being unpaused in the Manager ack
- message. (reported by jsmith in #asterisk-bugs)
-
- * main/config.c, /: Merged revisions 61958 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2
- lines Don't count failed include attempts against the
- configuration include level. (issue #9593 reported by mostyn)
- ........
-
-2007-04-25 22:29 +0000 [r61914] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007)
- | 2 lines handle a very bizarre race condition with channels
- being redirected before a simple switch can be started on them
- (issue #9286) ........
-
-2007-04-25 21:59 +0000 [r61863-61870] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) |
- 2 lines If the callerid= option is specified, but empty, clear
- any previous data. ........
-
- * /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) |
- 2 lines Ensure that callerid settings are reset on a reload.
- ........
-
-2007-04-25 19:21 +0000 [r61805] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c, main/channel.c, include/asterisk/app.h,
- funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2
- lines Merge rewritten group counting support. No more storing
- data on the variable list of the channels. That was bad, mmmk?
- (issue #7497 reported by sabbathbh) ........
-
-2007-04-25 16:22 +0000 [r61799] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) |
- 3 lines Fix a typo where cid_num got copied instead of cid_ani.
- (issue #9587, reported and patched by xrg) ........
-
-2007-04-24 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.3 released.
-
-2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 61786 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) |
- 4 lines Don't crash if a manager connection provides a username
- that exists in manager.conf but does not have a password, and
- also requests MD5 authentication. (ASA-2007-012) ........
-
- * main/channel.c, include/asterisk/channel.h: Improve DTMF handling
- in ast_read() even more in response to a discussion on the
- asterisk-dev mailing list. I changed the enforced minimum length
- of a digit from 100ms to 80ms. Furthermore, I made it now enforce
- a gap of 45ms in between digits. These values are not
- configurable in a configuration file right now, but they can be
- easily changed near the top of main/channel.c.
-
-2007-04-24 18:43 +0000 [r61779] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007)
- | 1 line removed #if 0 block from chan_phone, chan_zap, and
- chan_modem restart_monitor() ........
-
-2007-04-24 16:16 +0000 [r61774] Russell Bryant <russell@digium.com>
-
- * main/dial.c: Add a few more state changes in
- handle_frame_ownerless() so that the SLA code will get notified
- of these changes even when an owner channel is not provided. This
- isn't from a specific bug report, it's just something I noticed
- while poking around.
-
-2007-04-24 16:07 +0000 [r61772] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2
- lines Allow RFC2833 to be sent in the response SDP when an INVITE
- comes in without SDP. (issue #9546 reported by mcrawford)
- ........
-
-2007-04-23 18:17 +0000 [r61763-61765] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Some dialplan functions, such as CUT(), expect to
- operate on variables on a channel. So, this little hack lets them
- work in places where a channel doesn't exist, such as within
- DUNDi configuration. (issue #9465, reported and patched by
- Corydon76, testing by blitzrage)
-
- * main/channel.c: Ensure that digits passing through Asterisk have
- a reasonable minimum length. It is currently 100 ms. If someone
- thinks this should be different, feel free to speak up. (related
- to issues #8944, #9250, and #9348)
-
-2007-04-20 21:35 +0000 [r61705-61707] Jason Parker <jparker@digium.com>
-
- * main/rtp.c: Avoid invalid seqno cycling detection. Per comment
- from Dave Troy: This adds back in some simple typecasting I had
- in an earlier version which I realize now may be breaking things.
- Issue #9554.
-
- * main/loader.c, /: Merged revisions 61704 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4
- lines Fix an issue that I noticed while looking over issue 9571.
- The reload timestamp was getting set after reloading the built-in
- stuff, and before the modules. ........
-
-2007-04-20 20:42 +0000 [r61697] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: Remove a stray debug message introduced by a recent
- commit.
-
-2007-04-20 19:51 +0000 [r61694] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 61692 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5
- lines If the '* to hangup' option is not enabled, we don't need
- to disable * as a valid exit key. If it was enabled, this
- statement would've never been checked in the first place. Issue
- #9552 ........
-
-2007-04-20 18:19 +0000 [r61690] Russell Bryant <russell@digium.com>
-
- * main/config.c, apps/app_voicemail.c, main/manager.c,
- include/asterisk/config.h: Fix the UpdateConfig manager action to
- properly treat "variables" and "objects" differently (a=b versus
- a=>b). (issue #9568, reported by pari, patch by me)
-
-2007-04-19 08:37 +0000 [r61686] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3
- lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by
- Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........
-
-2007-04-19 04:36 +0000 [r61681-61683] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/manager.c: Bug 9557 - simple reason why reading a function
- always returned NULL
-
- * funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c,
- funcs/func_groupcount.c, /, funcs/func_timeout.c,
- funcs/func_cdr.c: Merged revisions 61680 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007)
- | 5 lines Bug 9557 - Specifying the GetVar AMI action without a
- Channel parameter can cause Asterisk to crash. The reason this
- needs to be fixed in the functions instead of in AMI is because
- Channel can legitimately be NULL, such as when retrieving global
- variables. ........
-
-2007-04-18 22:10 +0000 [r61678] Kevin P. Fleming <kpfleming@digium.com>
-
- * sounds/Makefile: allow external build systems to extract the
- required sound file versions
-
-2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson <oej@edvina.net>
-
- * main/rtp.c: Clean upp formatting, add some doxygen stuff while
- we're in cleaning mode... Thanks Kevin!
-
- * main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy)
-
-2007-04-16 14:47 +0000 [r61664-61666] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: #9483, half of patch by twilson to solve 302
- redirect issues
-
- * /: Blocking AstHoloPatch from 1.2
-
-2007-04-13 21:17 +0000 [r61658] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: This is a fix to the way CDR merge handles the data
- that results from ForkCDR.
-
-2007-04-13 19:17 +0000 [r61648-61656] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 61655 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2
- lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves
- the same as OUTBOUND_GROUP except it will get unset after use so
- it won't get accidentally inherited. (issue #BE-140) ........
-
- * apps/app_speech_utils.c: Do not bother looking for a result if
- none are present.
-
- * channels/chan_sip.c: For those very verbose SIP implementations
- that attach tons of info to the Contact header... let's increase
- our variable sizes. (issue #9535 reported by jeffg)
-
-2007-04-13 17:10 +0000 [r61645] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Eliminate a compiler warning with
- ODBC_STORAGE enabled so that it will build under dev-mode.
-
-2007-04-13 17:01 +0000 [r61644] Steve Murphy <murf@digium.com>
-
- * channels/chan_oss.c: A fix for chan_oss that resulted from the
- CDR changes; it helps to use the right info.
-
-2007-04-13 16:32 +0000 [r61641] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't assume the callid of a dialog will be
- set, as in some circumstances it may not. (issue #9534 reported
- by tecnoxarxa)
-
-2007-04-11 16:05 +0000 [r61477] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) |
- 5 lines If someone sets the "useragent" option in sip.conf to be
- empty, then don't add the User-Agent header at all. It is an
- optional header, anyway. Also, the bug report says that some of
- Japan's SIP providers don't allow it for some weird reason.
- (issue #9488, reported by makoto, fixed by me) ........
-
-2007-04-11 15:39 +0000 [r61443] Nadi Sarrar <ns@beronet.com>
-
- * channels/chan_misdn.c: Don't export AOCD variables on
- misdn_hangup anymore, this was mainly a fix for trunk..
-
-2007-04-11 15:09 +0000 [r61377-61427] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) |
- 6 lines Fix a bug with switching between host=dynamic and using
- specific hosts for peers. The code would only reset the peer's
- address when it is dynamic if it was a new peer structure. Now,
- it will also reset the address if it was already in the peer
- list, but before the reload, it was not dynamic. (issue #9515,
- reported by caio1982, fixed by me) ........
-
- * main/http.c: Add "svgz" to the mimetypes table. (issue #9510,
- bkruse) In passing, constify the elements of the mimetypes table.
-
- * /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) |
- 5 lines Remove the attempt at reporting configuration errors in
- sip.conf. This can cause a bunch of improper messages when using
- realtime. I give up. As oej tried to convince me when I put this
- in, there is just no easy way to do it. (inspired by a message on
- the -dev list) ........
-
-2007-04-11 13:40 +0000 [r61342-61373] Nadi Sarrar <ns@beronet.com>
-
- * channels/chan_misdn.c: Export AOCD variables on misdn_hangup.
-
- * channels/chan_misdn.c: Ignore facility messages in case we don't
- have a corresponding channel object.
-
- * channels/chan_misdn.c: AOCD's are now exported to asterisk
- channel variables.
-
-2007-04-10 16:05 +0000 [r61220] Russell Bryant <russell@digium.com>
-
- * main/Makefile, main/http.c, main/minimime (removed): File upload
- support was added to solve some needs for the Asterisk GUI.
- However, after much discussion, it has been decided that adding
- this to 1.4 is not in the best interests of the project. It has
- been removed here, but will remain in trunk.
-
-2007-04-10 12:43 +0000 [r61183] Nadi Sarrar <ns@beronet.com>
-
- * channels/misdn_config.c, /: Merged revisions 61170 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr
- 2007) | 2 lines msns config parameter defaults to '*' ........
-
-2007-04-10 05:18 +0000 [r61136] Steve Murphy <murf@digium.com>
-
- * apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a
- previous fix to overcome a compiler warning; the app NoCDR() has
- been updated to mark the channel CDR as POST_DISABLED instead of
- destroying the CDR; this way its flags are propagated thru a
- bridge and the CDR is actually dropped. The cases where only one
- channel in a bridge has a CDR was cleaned up.
-
-2007-04-09 19:58 +0000 [r61072] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3
- lines - Don't send ActionID before Response: header. - Don't use
- a blank in an AMI header ........
-
-2007-04-09 19:55 +0000 [r61062-61070] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/minimime/mm_envelope.c, res/res_features.c: fix up some
- warnings found using --enable-dev-mode
-
- * main/minimime/Doxyfile (removed),
- main/minimime/tests/messages/CVS (removed),
- main/minimime/tests/CVS (removed): remove some more stuff we
- don't need
-
-2007-04-09 19:41 +0000 [r61042-61044] Russell Bryant <russell@digium.com>
-
- * main/minimime/test (removed): Remove another directory that
- should no longer be there
-
- * main/minimime/Make.conf (removed), main/minimime/mytest_files
- (removed), main/minimime/.cvsignore (removed), main/minimime/sys
- (removed), main/minimime/mm-docs (removed): Remove various files
- that I thought I already removed.
-
-2007-04-09 19:05 +0000 [r61022] Jason Parker <jparker@digium.com>
-
- * apps/app_queue.c: Use the appropriate interface name with
- COMPLETECALLER. Issue 9395.
-
-2007-04-09 18:32 +0000 [r60989] Steve Murphy <murf@digium.com>
-
- * channels/chan_oss.c, main/channel.c, main/cdr.c,
- channels/chan_phone.c, channels/chan_misdn.c,
- channels/chan_skinny.c, channels/chan_features.c,
- channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c,
- channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
- channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c,
- channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
- include/asterisk/channel.h, channels/chan_gtalk.c,
- channels/chan_iax2.c: This is a big improvement over the current
- CDR fixes. It may still need refinement, but this won't have as
- many folks bothered.
-
-2007-04-09 18:02 +0000 [r60984] Olle Johansson <oej@edvina.net>
-
- * res/res_jabber.c: Add final new line after JabberEvent
-
-2007-04-09 17:22 +0000 [r60936] Jason Parker <jparker@digium.com>
-
- * /, apps/app_directory.c: Merged revisions 60935 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5
- lines Allow matching on names shorter than 3 chars. This also
- fixes the case where somebody wants to match on less then 3
- chars. Issue 9071 ........
-
-2007-04-09 03:01 +0000 [r60847-60850] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/asterisk.c, include/asterisk.h, /: Merged revisions 60849
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007)
- | 2 lines Don't check for error when lowering priority (according
- to the manpage, it should never happen anyway). It might could
- happen, though, if another thread messed with the priority, so
- safeguard against that (reported via -dev list). ........
-
- * channels/chan_local.c, /: Merged revisions 60846 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08
- Apr 2007) | 2 lines Bug 9505 - If the return value for
- local_queue_frame is set, then p->lock is no longer valid.
- ........
-
-2007-04-09 01:03 +0000 [r60762-60798] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 60797 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2
- lines When calling a device that then forwards us elsewhere... we
- have to make our channels compatible if it is the only channel
- being dialed. (issue #9445 reported by marcelbarbulescu) ........
-
- * apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if
- MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy)
-
-2007-04-08 14:14 +0000 [r60661-60713] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_macro.c: Merged revisions 60711 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007)
- | 2 lines Gosub called within a Macro resets the arguments
- improperly and causes general weirdness. (Issue 8329) ........
-
- * main/http.c: Fix --enable-dev-mode
-
- * channels/chan_oss.c: Off by one error, resulting in a crash
- (Issue 9500)
-
- * /, main/file.c: Merged revisions 60660 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007)
- | 2 lines Bug 9486 - memory leak when opening a filestream
- ........
-
-2007-04-06 20:58 +0000 [r60603] Russell Bryant <russell@digium.com>
-
- * main/minimime/sys/mm_queue.h, main/minimime/Doxyfile,
- main/minimime/mimeparser.yy.c, main/minimime/minimime.c,
- main/manager.c, main/minimime/mm_mimepart.c,
- main/minimime/test.sh, configure, include/asterisk/compat.h,
- main/strcompat.c, main/minimime/mm_internal.h, main/http.c,
- main/minimime/tests/parse.c, main/minimime/mm_base64.c,
- main/minimime/mm_mimeutil.c, main/minimime/mm.h,
- main/minimime/tests, main/minimime/mm_header.c,
- main/minimime/mm_error.c, main/Makefile,
- main/minimime/mm_codecs.c, main/minimime/mm_param.c,
- configure.ac, main/minimime/Makefile, main/minimime/mm_init.c,
- include/asterisk/manager.h, main/minimime/strlcpy.c,
- configs/http.conf.sample, main/minimime/mm_parse.c,
- main/minimime/tests/create.c, main/minimime/mm_contenttype.c,
- main/minimime/mm_util.c, main/minimime/mm_envelope.c,
- main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c,
- main/minimime/tests/messages/test2.txt,
- main/minimime/tests/messages/test3.txt,
- main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c,
- main/minimime/tests/messages/test4.txt,
- main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h,
- main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c,
- main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt,
- main/minimime/mimeparser.l, main/minimime/mm_context.c,
- main/minimime/mimeparser.tab.h, main/minimime (added),
- main/minimime/mm_warnings.c, main/minimime/mm_queue.h,
- main/minimime/tests/messages, include/asterisk/autoconfig.h.in,
- main/minimime/mimeparser.y, Makefile.moddir_rules,
- main/minimime/sys, main/minimime/tests/Makefile: To be able to
- achieve the things that we would like to achieve with the
- Asterisk GUI project, we need a fully functional HTTP interface
- with access to the Asterisk manager interface. One of the things
- that was intended to be a part of this system, but was never
- actually implemented, was the ability for the GUI to be able to
- upload files to Asterisk. So, this commit adds this in the most
- minimally invasive way that we could come up with. A lot of work
- on minimime was done by Steve Murphy. He fixed a lot of bugs in
- the parser, and updated it to be thread-safe. The ability to
- check permissions of active manager sessions was added by Dwayne
- Hubbard. Then, hacking this all together and do doing the
- modifications necessary to the HTTP interface was done by me.
-
-2007-04-06 20:32 +0000 [r60568-60572] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * UPGRADE.txt: clarified a sentence in the format_wav section
-
- * UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and
- plan to remove GAIN code from trunk
-
-2007-04-06 19:50 +0000 [r60521-60565] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: When a station picks up a trunk that was on
- hold, make the hints reflect that nobody has the trunk on hold
- anymore.
-
- * apps/app_meetme.c: Fix a few problems with SLA. (issue #9459,
- reported by francesco_r, fixed by me) * The original behavior was
- that if one station put a call on hold, another one picked it up,
- and then hung up, the code would still consider the call on hold
- by the first station, so the trunk would not be hung up. However,
- to better comply with what most people seem to expect it to
- behave, it will now hang up the trunk. * Fix a problem with
- "barge=no". This was only intended to prevent people from joining
- calls that are in progress. However, it also prevented other
- people from picking up a call that was on hold. This has been
- fixed. * When there are no active stations on a trunk and it is
- on hold, the code now indicates the HOLD and UNHOLD conditions to
- the trunk channel. This allows music on hold to be played to the
- trunk when it is on hold.
-
-2007-04-06 18:21 +0000 [r60459-60485] Matt Frederickson <creslin@digium.com>
-
- * channels/chan_zap.c: Make sure we check the faxdetect option
- before doing fax processing
-
- * channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2
- lines There should only be one code path for doing DTMF
- conditionals on channels. This fixes it. ........
-
-2007-04-06 14:49 +0000 [r60399] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007)
- | 2 lines remove undocumented 'cardsmode' parameter and stop
- searching for transcoders during reload() ........
-
-2007-04-06 01:14 +0000 [r60361] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c, apps/app_speech_utils.c,
- include/asterisk/speech.h: Add support for returning different
- types of results (ie: NBest).
-
-2007-04-05 22:58 +0000 [r60325] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * formats/format_wav.c: modified default GAIN for issue 5823,
- thanks jrwalliker
-
-2007-04-05 22:35 +0000 [r60323] Steve Murphy <murf@digium.com>
-
- * configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added
- some clarification to the example configs for CDRs, on how to
- select a backend. Also, made cdr-csv the default if you 'make
- samples', and no other changes.
-
-2007-04-05 16:10 +0000 [r60268] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5
- lines Just because we can't find the voicemail configuration
- file, doesn't mean that the module failed to load. The user could
- be using realtime. Issue #9473 ........
-
-2007-04-05 15:47 +0000 [r60265] Russell Bryant <russell@digium.com>
-
- * main/http.c: Add the MIME type for gif by request from Pari
-
-2007-04-05 12:55 +0000 [r60214] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2
- lines Only unlock our pvt and net locks if we are actually going
- to try to lock the owner again. (issue #9472 reported by zoa)
- ........
-
-2007-04-04 17:40 +0000 [r60013-60137] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 60134 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) |
- 6 lines It is valid to redirect channels via the manager
- interface that are not in the UP state. Instead of checking for
- that to prevent to ensure a dead channel doesn't get redirected,
- just use the ast_check_hangup() API call. (issue #9457, reported
- by Callmewind, patch by me) (related to issue #8977) ........
-
- * channels/chan_sip.c: Add a Content-Length of 0 to the response
- built by transmit_response_with_unsupported(). (issue #9454,
- reported by makoto, fixed by me)
-
- * /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) |
- 4 lines Fix the return value of handle_common_options() so that
- it always properly indicates whether it handled the option or
- not. (issue #9455, reported by Netview, fixed by me) ........
-
- * apps/app_meetme.c: Fix a problem where if a trunk was hung up
- while it was on hold, all of the hints would reflect the line
- still on hold, even though it should reflect that it is back to
- not in use. (issue #9459, reported by francesco_r, fixed by me)
-
-2007-04-03 19:40 +0000 [r59963] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Don't clash when a person both speaks
- and uses DTMF.
-
-2007-04-03 19:16 +0000 [r59853-59939] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) |
- 4 lines Don't attempt to report configuration errors in
- build_user(). oej pointed out that for a "friend" entry, this
- won't work, because all user options are valid for peers, but not
- the other way around. ........
-
- * /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) |
- 3 lines Make chan_sip report when it encounters an unknown
- option. (issue #9440, reported by nightcrawler) ........
-
- * /, main/app.c: Merged revisions 59886 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) |
- 5 lines When doing a built-in blind or attended transfer, restore
- the ability to use '#' to terminate the number and immediately do
- the transfer instead of having to dial the number and just wait
- for the feature digit timeout. (issue #8366, xueliangliang)
- ........
-
- * Makefile: Ensure that menuselect gets executed in dependency
- check mode every time you run make.
-
-2007-04-03 11:02 +0000 [r59804] Nadi Sarrar <ns@beronet.com>
-
- * channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h:
- Merged revisions 59788,59803 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2
- lines Use the new sysfs way of mISDN 1.2 to check if a port is NT
- or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di,
- 03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........
-
-2007-04-03 07:20 +0000 [r59774] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h:
- Merged revisions 59623-59624,59639 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) |
- 1 line we can now make 30 channels on a PRI (before we forgot
- chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200
- (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........
- r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) |
- 1 line added option which allows us to accept incoming SETUP
- Messages without automatically sending Proceeding or Setup
- Acknowledge, this is useful with some broken switches and if you
- want to Release incoming calls without previously having
- acknowledged them. The new option is
- noautorespond_on_setup=yes|no default is no, so we don't break
- the existing behaviour ........
-
-2007-04-02 18:58 +0000 [r59724] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2
- lines Increase the maximum size for a string of mailboxes to
- 1024. (issue #9270 reported by rtucker) ........
-
-2007-04-02 17:31 +0000 [r59688] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: continue in for-loop should go to the incrementer,
- not the test. As per 9435, thanks to marcelbarbulescu
-
-2007-04-02 15:39 +0000 [r59654] Russell Bryant <russell@digium.com>
-
- * main/netsock.c, /: Merged revisions 59608 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) |
- 6 lines Add the SO_REUSEADDR flag to sockets handled by netsock.
- This is needed by the patch that went in for issue 7874.
- chan_iax2 needs to be able to create socket that is lisetning on
- INADDR_ANY, but also be able to bind sockets to specific
- addresses. (Thanks to Stevenson on the asterisk-dev mailing list
- for explaining why this flag was needed.) ........
-
-2007-03-30 22:50 +0000 [r59573] Jason Parker <jparker@digium.com>
-
- * configure, main/Makefile, acinclude.m4: Add linux-uclibc host
- arch..."thingy". Sorry, I don't know what it's called...
-
-2007-03-30 17:51 +0000 [r59452-59522] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
- include/asterisk/cdr.h: several changes via kpflemings review
-
- * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
- include/asterisk/cdr.h: These mods fix CDR issues from 8221,
- 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated
- from transfer situations.
-
- * configs/extensions.conf.sample: A small clarification to keep
- bugs from being filed, and confusion from rising, if
- clearglobalvars is set, and globals are set in the AEL file.
- (9419)
-
-2007-03-29 17:43 +0000 [r59363] Russell Bryant <russell@digium.com>
-
- * res/res_jabber.c: When building a response to a subscription, the
- "from" must be the full Jabber ID. This fixes some problems where
- jabber users are not able to add their Asterisk account to their
- user list, since they are unable to get Asterisk to approve their
- subscription. (issue #8210, reported by caspy, and verified by
- bradtem)
-
-2007-03-29 17:38 +0000 [r59361] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2
- lines Keep a global array of variables indicating whether certain
- conference rooms are in use. This ensures that two people going
- into a new dynamic conference when the 'e' option is set don't go
- into the same conference room. (issue #8835 reported by eliel)
- ........
-
-2007-03-29 17:17 +0000 [r59304-59358] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, /: Merged revisions 59357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) |
- 5 lines If an error occurs when reading from an RTP socket, and
- the error code does not indicate that we should try again, then
- return NULL instead of a "null frame". This will prevent Asterisk
- from trying over and over again, and eventually causing the
- system to crash. (issue #8285, john) ........
-
- * channels/chan_iax2.c: When the IAX2 read callback gets called,
- return NULL instead of a "null frame". This will cause Asterisk
- to hangup the call instead of keep trying whatever it was doing.
- Under normal conditions, this function would *never* be called.
- However, the author of this patch says an error will occur that
- will cause it to get called every 100 thousand calls or so. When
- this does happen, it puts the channel in a loop that eventually
- brings down the system. So, hangup up the call is certainly a
- better alternative. (issue #8286, john)
-
- * Makefile: Export the GTK2 library and include information to sub
- Makefiles.
-
-2007-03-29 16:07 +0000 [r59300-59302] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007)
- | 3 lines Issue 9415 - No point to getting a diagnostic field if
- we aren't doing anything with the information. (Plus, it tends to
- crash the Postgres ODBC driver.) ........
-
-2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Another crash that I thought we had fixed already
- - Issue 9396
-
- * apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007)
- | 2 lines Oops ........
-
- * apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007)
- | 2 lines Fix a few remaining bad mmap(2) return values ........
-
-2007-03-27 23:20 +0000 [r59262-59278] Russell Bryant <russell@digium.com>
-
- * /, apps/app_directory.c: Merged revisions 59277 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) |
- 3 lines Fix the check of the return value from mmap(). Thanks to
- Corydon for catching this one. ........
-
- * apps/app_directory.c: Fix app_directory to actually compile with
- ODBC_STORAGE, and update the code to the latest res_odbc API.
-
- * apps/Makefile: Fix app_directory when ODBC_STORAGE is being used.
- The Makefile did not properly ensure that this information got
- copied from what was selected for app_voicemail. (issue #9224)
-
- * channels/chan_sip.c: Fix the check that ensures that the CHANNEL
- function's first argument is "rtpqos". Thanks, Corydon. :)
-
-2007-03-27 18:16 +0000 [r59261] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes
- asterisk), kpfleming pointed on asterisk-dev, that DECLINE in
- this case the proper thing to do. This change now has it doing
- the proper thing.
-
-2007-03-27 18:05 +0000 [r59256-59259] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) |
- 4 lines Fix the use of the "sourceaddress" option when "bindaddr"
- is set to 0.0.0.0 instead of having each interface explicitly
- listed. (issue #7874, patch by stevens) ........
-
- * channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS
- function to just be additional parameter of the CHANNEL function.
- This way, it will be possible for other RTP based channel drivers
- to expose this information in the future.
-
-2007-03-27 15:00 +0000 [r59254] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27
- Mär 2007) | 1 line fixed #9355 ........
-
-2007-03-26 21:45 +0000 [r59230] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * channels/chan_sip.c: Oops, this should be case insensitive
-
-2007-03-26 21:41 +0000 [r59228] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes
- asterisk). I turned a duplicate context from a WARNING to an
- ERROR. Now you get a module load failure, and asterisk just
- exits. That's better than a crash, right\?
-
-2007-03-26 21:37 +0000 [r59227] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * channels/chan_sip.c: Change this to a single dp function to make
- oej happy.
-
-2007-03-26 20:06 +0000 [r59225] Steve Murphy <murf@digium.com>
-
- * main/config.c: Fix for 9257; by eliminating the globals in
- main/config.c, we make it thread-safe, which is a minimum
- requirement.
-
-2007-03-26 19:34 +0000 [r59223] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Add ability to specify no timeout. This
- means as soon as the prompt is done playing it moves on to the
- next priority.
-
-2007-03-26 18:33 +0000 [r59215-59217] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Somehow the code for building the email for
- voicemail got out of sync. This change makes a few tweaks to get
- 1.4 in sync with trunk. (issue #9301)
-
- * apps/app_meetme.c: Fix some codec negotiation problems when
- CallerID support is not enabled in SLA. (issue #9308, reported by
- twilson)
-
-2007-03-26 18:13 +0000 [r59213] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Make SpeechBackground obey the digit
- timeout value.
-
-2007-03-26 17:53 +0000 [r59207-59209] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Rename the new dialplan functions to match
- the variable name
-
- * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The
- AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in
- some because they get set in sip_hangup. So, there are common
- situations where the variables will not be available in the
- dialplan at all. So, this patch provides an alternate method for
- getting to this information by introducing AUDIORTPQOS and
- VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76,
- with some testing by blitzrage)
-
-2007-03-26 17:38 +0000 [r59206] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
- pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE,
- and STANDALONE_AEL
-
-2007-03-26 15:25 +0000 [r59202] Nadi Sarrar <ns@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure,
- include/asterisk/autoconfig.h.in, channels/misdn/Makefile,
- channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2
- provides a dsp pipeline for i.e. echo cancellation modules, make
- chan_misdn use it. * add a check for linux/mISDNdsp.h to
- configure.ac and update the autogenerated files: 'configure',
- 'autoconfig.h.in' (the 'configure' script was not in sync with
- the latest configure.ac, so the diff is a bit bigger than
- expected).
-
-2007-03-26 15:16 +0000 [r59200] Joshua Colp <jcolp@digium.com>
-
- * pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the
- aelparse binary! DONT_OPTIMIZE should now work once again.
-
-2007-03-24 01:39 +0000 [r59195] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2
- lines Only try to handle a response if it has a response code.
- (ASA-2007-011) ........
-
-2007-03-23 16:11 +0000 [r59188-59189] Steve Murphy <murf@digium.com>
-
- * /: blocking out the fix in 59187... already incorporated here
-
- * /, apps/app_macro.c: Merged revisions 59186 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1
- line Added a few words in the Macro doc strings about the
- behavior of macros with hangups (et al.), as per 9337 ........
-
-2007-03-22 23:40 +0000 [r59180-59182] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: don't allow string input to overrun the
- buffer to hold it (ASA-2007-010)
-
- * channels/chan_misdn.c: remove variables that are no longer used
- (--enable-dev-mode is good, developers should be using it)
-
-2007-03-22 14:40 +0000 [r59145] Steve Murphy <murf@digium.com>
-
- * utils/Makefile: The stuff in utils was compiling with -O6 even if
- DONT_OPTIMIZE is set in menuconfig. Added the include to fix that
-
-2007-03-21 18:08 +0000 [r59081-59089] Joshua Colp <jcolp@digium.com>
-
- * main/http.c: Add svg mimetype for pari.
-
- * res/res_monitor.c, /: Merged revisions 59086 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2
- lines Indicate the filename changed when it is changed. (issue
- #9311 reported by jsmith) ........
-
- * channels/chan_sip.c: Until we can do media level parsing for
- sendrecv/etc just use the first value found. This crept up when a
- phone was offered audio+video and returned an inactive video
- stream. chan_sip thought the phone said to put the person on hold
- but that was totally wrong. (issue #9319 reported by benbrown)
-
-2007-03-20 21:04 +0000 [r59078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/logger.c: Fix defines for inline stack backtraces (only used
- by developers anyway)
-
-2007-03-20 20:42 +0000 [r59076] Joshua Colp <jcolp@digium.com>
-
- * channels/iax2-parser.c: Copy len variable as well, should fix
- remaining IAX2 DTMF issues.
-
-2007-03-20 17:48 +0000 [r59069-59070] Steve Murphy <murf@digium.com>
-
- * apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should
- return it to its previous, untouched, state.
-
- * apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h:
- The fix for the AEL <<security hole>> (bug 9316) is here...
-
-2007-03-20 13:16 +0000 [r59064] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h: Merged revisions
- 58849-58850,59062-59063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) |
- 1 line added method standard_dec for dialing out on groups, to
- avoid conflicts, which caused issues with some ISDN providers
- ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13
- Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 |
- crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
- avoid sending a disconnect when we already received one. ........
- r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) |
- 1 line modified a loglevel ........
-
-2007-03-19 Jason Parker <jparker@digium.com>
-
- * Asterisk 1.4.2 released.
-
-2007-03-19 22:29 +0000 [r59049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_strings.c: Oops, this should have been a %d all along
-
-2007-03-19 15:52 +0000 [r59042] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295
- reported by ajohnson)
-
-2007-03-19 15:42 +0000 [r59040] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * configs/sip_notify.conf.sample: Fix unescaped semicolon (reported
- via -dev list)
-
-2007-03-18 20:37 +0000 [r59037] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return
- code 0 (reported by qwerty1979)
-
-2007-03-18 16:36 +0000 [r59035] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_followme.c: Don't return a non-zero return code if the
- profile doesn't exist, to match what the documentation says it
- already does. (#9307 Reported by kkiely)
-
-2007-03-16 16:12 +0000 [r58992] Joshua Colp <jcolp@digium.com>
-
- * apps/app_page.c: Wait for the async thread to exit when hanging
- up all of the paged phones under all circumstances. (issue #9181
- reported by PhilSmith)
-
-2007-03-16 01:42 +0000 [r58947-58957] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample: fix a couple SLA documentation
- references
-
- * doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex
- (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added),
- doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added),
- doc/channelvariables.txt (added), doc/ael.txt (added),
- doc/billing.tex (removed), build_tools/prep_tarball,
- doc/callingpres.txt (added), doc/enum.txt (added),
- doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added),
- doc/cdrdriver.tex (removed), build_tools/make_buildopts_h,
- doc/security.txt (added), doc/imapstorage.txt (added),
- doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed),
- doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac,
- doc/iax.txt (added), doc/ael.tex (removed),
- doc/channelvariables.tex (removed), doc/enum.tex (removed),
- doc/security.tex (removed), doc/math.txt (added), Makefile,
- doc/imapstorage.tex (removed), doc/privacy.tex (removed),
- doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt
- (added), apps/app_voicemail.c, doc/cliprompt.txt (added),
- doc/chaniax.txt (added), doc/app-sms.txt (added),
- doc/ast_appdocs.tex (removed), doc/realtime.tex (removed),
- doc/ices.txt (added), doc/dundi.tex (removed),
- doc/linkedlists.txt (added), doc/queuelog.txt (added),
- doc/extconfig.txt (added), doc/radius.txt (added),
- doc/cliprompt.tex (removed), doc/chaniax.tex (removed),
- doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex
- (removed), doc/ices.tex (removed), doc/asterisk.tex (removed),
- doc/queuelog.tex (removed), doc/configuration.txt (added),
- doc/asterisk-conf.txt (added), doc/sla.pdf (added),
- doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt
- (added), doc/mp3.tex (removed), doc/configuration.tex (removed),
- doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added),
- doc/channels.txt (added), doc/ip-tos.tex (removed),
- doc/extensions.txt (added), doc/queues-with-callback-members.txt
- (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added),
- doc/misdn.txt (added), doc/manager.txt (added),
- doc/jitterbuffer.tex (removed), doc/extensions.tex (removed),
- doc/billing.txt (added), doc/localchannel.txt (added),
- doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt
- (added), doc/00README.1st (added): Making these documentation
- changes in the 1.4 branch upset various people, so these chanes
- will only be done in the trunk.
-
- * build_tools/prep_tarball: Add the --pdf option to the usage of
- rubber in prep_tarball
-
- * Makefile, build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
- configure script checking for GTK2 and some additional Makefile
- targets to support gmenuselect
-
-2007-03-15 23:52 +0000 [r58946] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match
- common syntax and update the resulting appdocs TeX file
-
-2007-03-15 23:24 +0000 [r58941] Russell Bryant <russell@digium.com>
-
- * doc/asterisk.tex: add a link to the rubber homepage
-
-2007-03-15 23:11 +0000 [r58939] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_setcdruserfield.c, main/pbx.c,
- apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c:
- Expand deprecation warnings from simply warning on use to the
- builtin documentation.
-
-2007-03-15 22:51 +0000 [r58935-58937] Russell Bryant <russell@digium.com>
-
- * doc/asterisk.tex, Makefile: Add Asterisk version information to
- the generated PDF
-
- * build_tools/prep_tarball: have prep_tarball attempt to build
- asterisk.pdf
-
-2007-03-15 22:32 +0000 [r58933] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_realtime.c: Function works fine, but the documentation
- is backwards.
-
-2007-03-15 22:25 +0000 [r58931] Russell Bryant <russell@digium.com>
-
- * doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex
- (added), doc/freetds.txt (removed), doc/odbcstorage.txt
- (removed), configure, doc/sla.tex, doc/cygwin.txt (removed),
- doc/model.txt (removed), doc/channelvariables.txt (removed),
- doc/ael.txt (removed), doc/billing.tex (added),
- doc/callingpres.txt (removed), doc/enum.txt (removed),
- doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed),
- doc/cdrdriver.tex (added), build_tools/make_buildopts_h,
- doc/security.txt (removed), doc/imapstorage.txt (removed),
- doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added),
- doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac,
- doc/iax.txt (removed), doc/ael.tex (added),
- doc/channelvariables.tex (added), doc/enum.tex (added),
- doc/security.tex (added), doc/math.txt (removed), Makefile,
- doc/imapstorage.tex (added), doc/privacy.tex (added),
- doc/realtime.txt (removed), doc/dundi.txt (removed),
- doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt
- (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed),
- doc/ast_appdocs.tex (added), doc/realtime.tex (added),
- doc/ices.txt (removed), doc/dundi.tex (added),
- doc/linkedlists.txt (removed), doc/queuelog.txt (removed),
- doc/extconfig.txt (removed), doc/radius.txt (removed),
- doc/cliprompt.tex (added), doc/chaniax.tex (added),
- doc/hardware.txt (removed), doc/mp3.txt (removed),
- doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex
- (added), doc/queuelog.tex (added), doc/configuration.txt
- (removed), doc/asterisk-conf.txt (removed), doc/sla.pdf
- (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added),
- doc/h323.txt (removed), doc/mp3.tex (added),
- doc/configuration.tex (added), doc/asterisk-conf.tex (added),
- doc/jitterbuffer.txt (removed), doc/channels.txt (removed),
- doc/ip-tos.tex (added), doc/extensions.txt (removed),
- doc/queues-with-callback-members.txt (removed), doc/apps.txt
- (removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt
- (removed), doc/manager.txt (removed), doc/jitterbuffer.tex
- (added), doc/extensions.tex (added), doc/billing.txt (removed),
- doc/localchannel.txt (removed),
- doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt
- (removed), doc/00README.1st (removed): Merge changes from
- svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc
- directory into a single LaTeX formatted document so that we can
- generate a PDF, HTML, or other formats from this information. *
- Add a CLI command to dump the application documentation into
- LaTeX format which will only be include if the configure script
- is run with --enable-dev-mode. * The PDF turned out to be close
- to 1 MB, so it is not included. However, you can simply run "make
- asterisk.pdf" to generate it yourself. We may include it in
- release tarballs or have automatically generated ones on the web
- site, but that has yet to be decided.
-
-2007-03-15 18:13 +0000 [r58923] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Don't assume that the pvt structure will
- still exist after calling schedule_delivery as it may not. (issue
- #9278 reported by fmachado)
-
-2007-03-14 19:18 +0000 [r58894-58906] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Some people like to put "limitonpeer"
- instead of "limitonpeers" in their configuration. While we're at
- it, support "limitonpeerz" and "limitonpeerssssss". (inspired by
- issue #9172)
-
- * doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the
- examples section
-
- * doc/security.txt, /: Merged revisions 58896 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) |
- 3 lines Add a note to the security file that the Asterisk CLI and
- log files may contain sensitive information, and that people
- should keep this in mind. ........
-
- * configs/sla.conf.sample, apps/app_meetme.c: By default, don't
- attempt to do any CallerID handling at all with SLA because it is
- known to not work properly in some situations. However, add an
- option to enable it for those that would like to use it anyway.
- The short story behind this is that to properly handle CallerID
- with SLA, we need the ability to change the CallerID on an
- existing call, and we are not ready to handle that.
-
-2007-03-14 01:47 +0000 [r58880] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_strings.c: Issue 9162 -
- pbx_substitute_variables_helper assumes the buffer is initialized
- to all zeroes. This fixes a case where it wasn't.
-
-2007-03-13 23:19 +0000 [r58870-58872] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Ensure that the blinky lights show that the
- trunk stopped ringing when the trunk hangs up before a station
- has answered it. (issue #9234, reported by francesco_r)
-
- * configs/sla.conf.sample: fix the reference to the SLA
- documentation
-
-2007-03-13 11:49 +0000 [r58843-58848] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2
- lines Issue #9229 - No port in request URI on register to non
- default SIP ports (neelakantan) ........
-
- * channels/chan_sip.c: Don't hangup the call on OK or errors on
- MESSAGE and INFO inside of a dialog (like video update requests).
-
- * channels/chan_sip.c: Issue #9251 - Clear From URI from user
- attributes (tgrman)
-
-2007-03-12 13:08 +0000 [r58825-58826] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 57034,57523,57753,58558 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) |
- 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com
- bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02
- 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........
- r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) |
- 1 line fixed another place where the out_cause was hardcoded to
- 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09
- Mar 2007) | 1 line we can free channel 31 as well, since we can
- occupy it ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, channels/misdn/ie.c,
- channels/misdn/isdn_msg_parser.c: added UU transceiving and
- corect handling for rdnis
-
-2007-03-12 01:21 +0000 [r58779-58783] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Allow RFC2833 compensation to compensate for even
- stupider implementations by queueing up the end frame at the
- start, not the actual end. (issue #8963 reported by AndrewZ)
-
- * channels/chan_sip.c, configs/sip.conf.sample: Add
- matchexterniplocally setting which only substitutes your
- externip/externhost setting if it matches the localnet setting. I
- know of at least two people who need opposite settings, so I made
- it an option! (issue #8821 reported by kokoskarokoska)
-
-2007-03-10 18:11 +0000 [r58638-58705] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a few more places in chan_iax2 where
- the ast_frame used for receiving a frame was not properly
- initialized. - Interpolating a frame when the jitterbuffer is in
- use - decrypting a frame when IAX2 encryption is on - frames in
- an IAX2 trunk
-
- * apps/app_meetme.c: Make the compiler happy and initialize a
- variable.
-
- * doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added):
- Merge some updates to the SLA documentation. I plan to keep
- working on this to explain all of the expected behavior with call
- handling, configuration details for specific phones, and other
- things. However, I got tired of doing it in plain text, so I
- switched to using LaTeX. I have included the PDF version. I
- haven't been able to get a nice looking plain text version out of
- it yet, but I'm not terribly concerned since this is supposed to
- be more of the manual, while the plain text sample configuration
- file is the reference.
-
-2007-03-09 21:08 +0000 [r58584-58604] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Fix spelling of unavailable in voicemail
- documentation. (issue #9248 reported by tensai)
-
- * /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2
- lines If we are unable to lookup the host in a c line we have to
- abort, otherwise the previous data is gone and we will
- (potentially) have no data when all is said and done. ........
-
-2007-03-08 22:15 +0000 [r58510-58512] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Hang up the channel that put the call on hold
- in the event processing thread to avoid a race condition. Also,
- if the station originated the call that it is putting on hold,
- don't hang up the trunk if it was the only station on the call
- and it is hanging up due to hold and not a normal hangup.
-
- * channels/chan_zap.c: Add a missing break statement so that
- handling the above event does not incorrectly destroy the
- channel. (issue #9242, andrew)
-
-2007-03-08 21:33 +0000 [r58479] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Fix segfault (Issue 9236)
-
-2007-03-08 20:54 +0000 [r58474] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Refactor hold handling a bit so that it does
- not require keeping the call up when a call is put on hold.
-
-2007-03-08 18:01 +0000 [r58389-58436] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Make early SDP seeding even smarter! We have to check
- codecs in the make_compatible function too. (issue #9221 reported
- by marcelbarbulescu)
-
- * main/dsp.c, /: Merged revisions 58388 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2
- lines Only print out debug message if the definition that makes
- the variables shows up was actually defined. (issue #9233
- reported by serginuez) ........
-
-2007-03-08 13:23 +0000 [r58351-58354] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/http.c: this change was not needed; fclose() handles closing
- the file descriptor already
-
- * apps/app_meetme.c: fix a compiler warning, and overwriting 'res'
- value
-
- * main/http.c: fix two cases where HTTP session file descriptors
- would not be closed
-
-2007-03-08 01:01 +0000 [r58243-58320] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, configure, configure.ac: If we receive
- ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256,
- tzafrir) Also, update the configure script to make sure that we
- don't try to build chan_zap if the installed version of zaptel
- does not include ZT_EVENT_REMOVED.
-
- * /, channels/chan_iax2.c: (This bug was reported to me by Kinsey
- Moore) Merged revisions 58242 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) |
- 7 lines Fix a problem where the Asterisk channel name could be
- that of the wrong IAX2 user for a call. This is because the first
- step of choosing this name is to look for an IAX2 peer that
- happens to have the same IP/port number that this call is coming
- from and assuming that is it. However, this is not always
- correct. So, I have made it change this name after authentication
- happens since at that point, we have an exact match. ........
-
-2007-03-07 17:52 +0000 [r58240] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, channels/chan_sip.c: Ensure we have (or should have)
- at least one matching codec before attempting early bridge SDP
- seeding. (issue #9221 reported by marcelbarbulescu)
-
-2007-03-07 00:27 +0000 [r58165-58168] Russell Bryant <russell@digium.com>
-
- * main/manager.c, /: Merged revisions 58164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) |
- 4 lines If the channels acquired using the manager Redirect
- action are not up, then don't attempt to do anything with them.
- It could lead to weird behavior, including crashes. (issue #8977)
- ........
-
-2007-03-06 23:10 +0000 [r58121] Steve Murphy <murf@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1
- line Fix for 9220: Eyebeam cannot renew subscriptions for
- presence info. Reason: re-SUBSCRIBE requests don't include Accept
- headers, which the rfc says are optional (to put it tersely), (it
- uses MAY), and luckily, the sip_pvt struct has the format info
- stored, so we simply leave it if the format is set, and the
- accept header null. ........
-
-2007-03-06 23:00 +0000 [r58119] Russell Bryant <russell@digium.com>
-
- * configs/voicemail.conf.sample: Clarify the documentation of the
- dialout and sendvoicemail options. (issue #9000, caio1982 and
- serge-v)
-
-2007-03-06 20:37 +0000 [r58053] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2
- lines Change error message to proper message ........
-
-2007-03-06 18:01 +0000 [r58023] Russell Bryant <russell@digium.com>
-
- * channels/chan_skinny.c: Return an error of transmit_response is
- called without a session. (issue #9002)
-
-2007-03-05 19:19 +0000 [r57870-57914] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Since chan_iax2 does not support reception
- of DTMF with duration ensure that it is set to 0 on the frame.
- (issue #8521 reported by gdhgdh)
-
- * apps/app_meetme.c: Don't create a listen channel and record the
- conference unless the option is turned on. (issue #9204 reported
- by francesco_r)
-
- * apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2
- lines Make create_dirpath use our standard for return values. -1
- is failure, 0 is success. (issue #9205 reported by ballares)
- ........
-
-2007-03-05 15:20 +0000 [r57826] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 57825 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1
- line Fixed a typo introduced via 9156 (either the gotos or their
- doc strings are wrong) ........
-
-2007-03-05 04:19 +0000 [r57768-57798] Joshua Colp <jcolp@digium.com>
-
- * main/slinfactory.c: Don't allow a NULL pointer to reach
- ast_frdup. (issue #9155 reported by cmaj)
-
- * res/res_jabber.c: Don't reference a potentially NULL pointer.
- (issue #9199 reported by klolik)
-
- * main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198
- reported by edgreenberg)
-
-2007-03-03 15:31 +0000 [r57707] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2,
- pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7:
- Updated the regression tests
-
-2007-03-03 06:45 +0000 [r57649] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007)
- | 2 lines Memory leak of a list, if call recording was abandoned
- ........
-
-2007-03-03 00:59 +0000 [r57620] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/say.c: submitted patch for Georgian language, issue 9010,
- submitted by Alexander Shaduri
-
-2007-03-03 00:02 +0000 [r57591] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample: add missing configuration template.
- Thanks to Lacy Moore on asterisk-users for pointing this out\!
-
-2007-03-02 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.1 released.
-
-2007-03-02 23:03 +0000 [r57556] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Update the check that is used to
- determine whether zaptel transcoder support is present. The
- interface has changed.
-
-2007-03-02 17:06 +0000 [r57477] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2
- lines 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 16:55 +0000 [r57426-57473] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 57458 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1
- line further refinement in wording of goto documentation, as per
- 9156, goto not proceeding to next instruction ........
-
- * pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes
- right, but 9184 points out the problem-- the escape is removed by
- pbx_config, and pbx_ael should also, before sending it down into
- the pbx engine. Also, you have to insert it back in, if you are
- generating extensions.conf code from the AEL.
-
-2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant <russell@digium.com>
-
- * main/file.c: Return the correct digit that interrupted the
- stream. This fixes exiting the Background application when using
- the m option. (issue #9176, mjagdis)
-
- * configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt,
- include/asterisk/channel.h: Merge changes from
- svn/asterisk/team/russell/sla_updates * Originally, I put in the
- documentation that only Zap interfaces would be supported on the
- trunk side. However, after a discussion with Qwell, we came up
- with a way to make IP trunks work as well, using some things
- already in Asterisk. So, here it is, this now officially supports
- IP trunks. * Update the SLA documentation to reflect how to setup
- IP trunks. * Add a section in sla.txt that describes how to set
- up an SLA system with voicemail. * Simplify the way DTMF
- passthrough is handled in MeetMe. * Fix a bug that exposed itself
- when using a Local channel on the trunk side in SLA. The
- station's channel needs to be passed to the dial API when dialing
- the trunk. * Change a WARNING message to DEBUG in channel.h. This
- message is of no use to users.
-
-2007-03-01 22:21 +0000 [r57318] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 57317 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar
- 2007) | 2 lines 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) ........
-
-2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development <support@transnexus.com>
-
- * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick.
-
-2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla
- docs
-
- * configs/sla.conf.sample, apps/app_meetme.c: Merge more changes
- from svn/asterisk/team/russell/sla_updates * Add support for
- private hold. By setting "hold=private" for a trunk, only the
- station that put the call on hold will be able to retrieve it
- from hold. Also, by setting "hold=private" for a station, any
- call that station puts on hold can only be retrieved by that
- station.
-
- * apps/app_meetme.c: Minor formatting change
-
- * configs/sla.conf.sample, apps/app_meetme.c: Merge changes from
- svn/asterisk/team/russell/sla_updates * Add support for the
- "barge=no" option for trunks. If this option is set, then
- stations will not be able to join in on a call that is on
- progress on this trunk.
-
-2007-02-28 19:23 +0000 [r57139] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 57118 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1
- line 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 18:57 +0000 [r57093] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 57092 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb
- 2007) | 2 lines Fix a few more issues with the agent logoff CLI
- command. (issue #9123 reported by arbrandes) ........
-
-2007-02-28 18:20 +0000 [r57089] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample, apps/app_meetme.c: Merge current set of
- changes from svn/asterisk/team/russell/sla_updates * Add support
- for station ring delays. Ring delays can be set globally for a
- station or for specific trunks on the station. * Fix a few bugs
- in existing code. * Restructure and Reorganize code to improve
- readability and maintainability. * Improve formatting of the "sla
- show (trunks|stations)" CLI commands.
-
-2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Picky compiler...
-
- * apps/app_speech_utils.c: Better handle timeouts when the
- individual speaks after everything has been played but before the
- timeout ends.
-
-2007-02-28 17:15 +0000 [r57049] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: I was surprised that I had not yet downgraded
- missing goto targets and macro call defs to a warning, in case
- they are in extensions.conf; I rectified this problem. Also, A
- goto in a macro to a target in a catch block was not being found;
- I fixed this too; the cause was that I needed to treat catch
- statements like an extension in the find_match code.
-
-2007-02-27 17:36 +0000 [r56975] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Fix voicemail email attachments. I missed
- the conversion of one of the line endings and there was an extra
- one where it should not have been. (issue #9128)
-
-2007-02-26 22:01 +0000 [r56922] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky,
- picky... show deprecation warning in application help, too
- (reported via list)
-
-2007-02-26 20:42 +0000 [r56888] Russell Bryant <russell@digium.com>
-
- * channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where
- if a device was not specified in alsa.conf, then we just use the
- system default, instead of creating our own default of hw:0,0.
- (issue #9139)
-
-2007-02-26 20:07 +0000 [r56856] Joshua Colp <jcolp@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2
- lines Obey the clearglobalvars option in extensions reload (or
- dialplan reload depending on your version). (issue #9146 reported
- by ramonpeek) ........
-
-2007-02-26 20:04 +0000 [r56847] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a crash in my last change to
- iax2_indicate(). (issue #9150)
-
-2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp <jcolp@digium.com>
-
- * apps/app_record.c: Update app_record documentation to use new CLI
- command, core show file formats. (issue #9151 reported by junky)
-
- * main/pbx.c: Use ast_strlen_zero to see if the language and/or
- context argument is not present for Background instead of just
- checking if it is NULL. (issue #9141 reported by mjagdis)
-
-2007-02-26 16:51 +0000 [r56785] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Do more complete locking of the
- chan_iax2_pvt struct in the indicate callback. (Problem brought
- up by Ben Smithurst on the asterisk-dev list)
-
-2007-02-26 16:36 +0000 [r56783] Joshua Colp <jcolp@digium.com>
-
- * main/asterisk.c: Allow both of the show version files and core
- show file versions CLI commands to work. (issue #9135 reported by
- mvanbaak)
-
-2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Move a comment to be in the correct struct.
-
-2007-02-25 14:46 +0000 [r56685] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/channel.c, /: Merged revisions 56684 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007)
- | 3 lines 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. ........
-
-2007-02-24 02:02 +0000 [r56569] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Make sure to set a speeddials parent on
- creation. Don't crash if hold is pressed when no call is active.
- Don't return in places that we shouldn't..
-
-2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: update to match zaptel 1.4 API change that
- was committed a few minutes ago
-
-2007-02-23 23:24 +0000 [r56505] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, /: Merged revisions 56504 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) |
- 8 lines 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. ........
-
-2007-02-23 21:53 +0000 [r56457] Joshua Colp <jcolp@digium.com>
-
- * main/sched.c: Change log notice to debug. It is possible for a
- scheduled item to execute and be deleted at close to the same
- time and unavoidable. If this happens this message creeps up.
-
-2007-02-23 20:20 +0000 [r56407] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) |
- 4 lines 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). ........
-
-2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/make_version_h: build special version strings for
- AADK/S800i builds
-
-2007-02-23 17:58 +0000 [r56341] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: The IMAP storage code uses the same code to
- build the email that is used when voicemail is sent via email
- using something like sendmail. In the patch from bug 8033 to fix
- various IMAP storage problems, the line endings in the email file
- were changed in the code from "\n" to "\r\n". However, this
- breaks sending regular voicemail to email. So, this change
- conditionally sets line endings to "\r\n" only if IMAP_STORAGE is
- enabled. (issue #9128, patch by jarjarbinks, modified by me to
- not break IMAP storage)
-
-2007-02-22 23:08 +0000 [r56277] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample, main/dial.c, apps/app_meetme.c,
- doc/sla.txt: Merge changes from team/russell/sla_updates. This
- batch of changes to the SLA code does a few different things. * I
- made the SLA code event driven instead of having to act in a lot
- of busy loops while dialing things to wait for state changes.
- This makes the code more efficient and readable at the same time.
- * I have implemented a couple of new features. The first is
- inbound trunk ringing timeouts. This is an option that defines
- how long to let an incoming call on a trunk to ring. * I have
- also implemented ring timeouts for stations. They may be
- specified for the entire station, meaning it is how long to let
- the station ring before giving up. You can also specify a ring
- timeout for a specific trunk on a station. So, you can say that
- you only want a specific station to ring 5 seconds if it is line1
- ringing, but otherwise, there is no timeout.
-
-2007-02-22 18:49 +0000 [r56231] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /, channels/chan_sip.c: Merged revisions 56230
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2
- lines 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 14:06 +0000 [r56169] TransNexus OSP Development <support@transnexus.com>
-
- * doc/osp.txt: Update OSP documentation for v1.4.
-
-2007-02-22 10:33 +0000 [r56125] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Move message from verbose to debug
-
-2007-02-22 02:39 +0000 [r56094] Steve Murphy <murf@digium.com>
-
- * sounds/Makefile: updated the sound tarball versions in Makefile
-
-2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Restructure a little bit of code to reduce
- nesting. There is no functionality change here.
-
- * /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) |
- 3 lines If we receive a frame that is not in any of the
- negotiated formats, then drop it. (potentially issue #8781 and
- SPD-12) ........
-
-2007-02-22 00:35 +0000 [r56008] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c: Print out deprecation notice on usage output of CLI
- commands. (issue #8925 reported by blitzrage)
-
-2007-02-22 00:08 +0000 [r56006] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/loader.c: disable unloading of embedded modules... there is
- a fundamental problem with doing so that will not be fixed in
- this version of Asterisk due to its invasiveness
-
-2007-02-21 20:35 +0000 [r55957] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2
- lines 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 20:27 +0000 [r55954] Jason Parker <jparker@digium.com>
-
- * channels/chan_gtalk.c: Fix locking issue, and accept
- "transport-accept" as a valid accept message. This should solve
- issues 8970 and 8503.
-
-2007-02-21 20:22 +0000 [r55951] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Simplify the last change to app_meetme, and
- move the call to dispose_conf() up into the block where we know a
- conf exists.
-
-2007-02-21 20:16 +0000 [r55914-55949] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Only dispose of the conference if one was
- created.
-
- * apps/app_speech_utils.c: Only start playing the next file if we
- have not been quieted.
-
- * channels/chan_sip.c: Add a flag that indicates whether a SIP
- dialog is an outgoing call or not. SIP_OUTGOING originally did it
- but it was repurposed to the direction of the last transaction,
- which can cause update_call_counter to falsely decrease the wrong
- counters. (please don't hurt me oej) (issue #8943 reported by
- mdu113)
-
-2007-02-21 14:06 +0000 [r55869] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, build_tools/make_version: Merged revisions 55868 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21
- Feb 2007) | 2 lines use new tag version script ........
-
-2007-02-21 08:32 +0000 [r55834] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly
- after transfer (decrement inuse early on transferer's call leg)
-
-2007-02-21 02:01 +0000 [r55799] Jason Parker <jparker@digium.com>
-
- * channels/chan_gtalk.c: Fix segfault when buddy couldn't be found.
- Issue 7764, patch by sailer
-
-2007-02-21 01:03 +0000 [r55751-55758] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Improve the reference counting to fix bugs
- where people report seeing conferences listed that have no
- members. (issue #9073)
-
-2007-02-21 00:11 +0000 [r55670-55741] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Better handle dropped IMAP connections.
- (issue #9054 reported by bsmithurst)
-
- * channels/chan_sip.c: Return behavior I removed. I did not
- remember that you could just add a localnet entry to make it
- work.
-
- * channels/chan_sip.c: Don't test our own address against the
- localnet settings. At least one person has had issues as a result
- of this from #7051 so I'm reversing it. (issue #8821 reported by
- kokoskarokoska)
-
- * /, channels/chan_agent.c: Merged revisions 55669 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb
- 2007) | 2 lines 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 20:26 +0000 [r55589-55634] Russell Bryant <russell@digium.com>
-
- * main/http.c: Add the Asterisk version information to the Server
- header in HTTP responses. (requested by Pari)
-
- * include/asterisk/manager.h: Increase the maximum number of
- manager headers to 128, at the request of Pari.
-
-2007-02-20 16:53 +0000 [r55555] Jason Parker <jparker@digium.com>
-
- * channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free
- with strdupa (thanks file) 55555!
-
-2007-02-20 16:41 +0000 [r55553] Russell Bryant <russell@digium.com>
-
- * configs/sla.conf.sample: Change the formatting of sla.conf.sample
- to make it more readable. (issue #9112, blitzrage)
-
-2007-02-19 21:12 +0000 [r55483] Olle Johansson <oej@edvina.net>
-
- * res/res_jabber.c: - Not sending arguments to an application is
- not "out of memory" - Making error messages a bit more clear
-
-2007-02-19 18:11 +0000 [r55435] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007)
- | 2 lines forcename and forcegreetings options should check to
- see if the recording already exists ........
-
-2007-02-19 14:52 +0000 [r55397] Doug Bailey <dbailey@digium.com>
-
- * channels/chan_iax2.c: Changed iax2 process thread to detached to
- correct memory leak due to left over thread context on thread
- exit. Modified module unload process to avoid deadlocks on
- pthread cancels
-
-2007-02-18 12:35 +0000 [r55250-55278] Olle Johansson <oej@edvina.net>
-
- * /, apps/app_record.c: Merged revisions 55277 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2
- lines Documentation update (#9053, jsmith) ........
-
- * /: Block patch that was made only for 1.2 (already implemented in
- 1.4 and trunk)
-
-2007-02-17 17:39 +0000 [r55219] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Add missing membername option to AddQueueMember
- documentation. (issue #9088 reported by seanbright)
-
-2007-02-17 17:10 +0000 [r55217] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix an issue where callerid would not be
- displayed on some phones. Issue 8995, initial patch and research
- done by wedhorn
-
-2007-02-17 03:55 +0000 [r55086-55154] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 55153 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2
- lines Answer the channel before recording privacy information.
- (issue #8926 reported by lmamane) ........
-
- * apps/app_queue.c: Make the 'i' option of Queue actually work.
- (issue #8986 reported by utis)
-
- * /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2
- lines 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) ........
-
-2007-02-17 00:40 +0000 [r55006-55052] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: If the
- pg_config application is found, but there is probably executing
- it, then consider postgres unavailable. (issue #8637)
-
- * codecs/gsm/Makefile: Filter out yet another architecture that
- does not work with the optimizations in the built-in libgsm.
- (issue 8637, ovi)
-
- * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged
- revisions 55005 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) |
- 9 lines 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 22:18 +0000 [r55002] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 54999 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb
- 2007) | 2 lines 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 21:12 +0000 [r54969] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) |
- 5 lines 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 18:51 +0000 [r54924] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Need to check macro extension as well as macro
- context for directed pickup.
-
-2007-02-16 18:03 +0000 [r54888-54898] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_config.c: Fix setting "autofallthrough" to yes by
- default. It was set to enabled in pbx.c. However, if the option
- was not present in extensions.conf, then pbx_config.c would set
- it back to disabled.
-
- * res/res_features.c: Clean up a few coding guidelines issues -
- spaces to tabs, use sizeof() to pass the size of a static buffer,
- add spaces ...
-
-2007-02-16 17:25 +0000 [r54886] Jason Parker <jparker@digium.com>
-
- * main/asterisk.c: Clarify a restart message. It's silly, but the
- reporter had a very valid point. Issue 9079
-
-2007-02-16 17:02 +0000 [r54884] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Allow directed pickup to pick up the real
- context instead of the macro context if a Macro is used. (issue
- #8984 reported by jamesb63)
-
-2007-02-16 12:06 +0000 [r54772-54787] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #7541 - Handle multipart attachments
- to SIP messages - even if boundary is quoted.
-
- * /, res/res_agi.c: Merged revisions 54771 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2
- lines Issue #9069 - If we open with TH we should not close with
- /TD. (seanbright) ........
-
-2007-02-16 00:48 +0000 [r54481-54714] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Don't let dtmf leak over into the engine
- and let it skew the results... also give DTMF results priority.
- (issue #9014 reported by surftek)
-
- * apps/app_dial.c, /: Merged revisions 54622 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2
- lines Use a separate variable to indicate execution should
- continue instead of the return value. (issue #8842 reported by
- pluto70) ........
-
- * apps/app_dial.c: Forward begin DTMF frames as well as end. (issue
- #9068 reported by mhardeman)
-
-2007-02-14 18:44 +0000 [r54439] Olle Johansson <oej@edvina.net>
-
- * /: Block patch only needed in 1.2
-
-2007-02-14 16:56 +0000 [r54375] Matt Frederickson <creslin@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2
- lines When handling glare on a PRI, move the requested channel
- rather than hang up the old one. Fix for 8957 and 9011. ........
-
-2007-02-14 01:09 +0000 [r54290] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Add G722 to ast_best_codec. If anyone disagrees
- with it's placement, feel free to change it. (issue #9045
- reported by gork)
-
-2007-02-13 21:31 +0000 [r54204-54235] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Remove a couple of leftover debug messages
-
- * include/asterisk/devicestate.h: Fix the documentation on the
- return values from device state provider registration and
- deletion.
-
- * channels/chan_sip.c: If we fail to create the SIP socket, then
- return -1 from reload_config() so that load_module() will return
- AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get
- spammed with error messages every time chan_sip tries to send a
- message.
-
-2007-02-13 18:41 +0000 [r54180] Olle Johansson <oej@edvina.net>
-
- * /: Blocking patch for 1.2 only
-
-2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant <russell@digium.com>
-
- * main/dial.c, include/asterisk/dial.h: Change
- ast_set_state_callback() to ast_dial_set_state_callback()
-
- * main/dial.c, apps/app_meetme.c, apps/app_page.c,
- include/asterisk/dial.h: - Add the ability to register a callback
- to monitor state changes in an asynchronous dial operation. -
- Rename the various references to "status" to "state" in the dial
- API
-
-2007-02-12 16:34 +0000 [r54026] Joshua Colp <jcolp@digium.com>
-
- * configure, configure.ac: Make the --without-oss argument work.
- (issue #9026 reported by puzzled)
-
-2007-02-12 15:38 +0000 [r54002] Russell Bryant <russell@digium.com>
-
- * configs/users.conf.sample: Fix a typo where "vmpassword" should
- be "vmsecret"
-
-2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c: Fix VLDTMF reception
-
- * apps/app_echo.c: Much simpler than previous one ;-)
-
- * main/channel.c: Provide correct DTMF duration
-
- * main/cli.c: Bring deprecated 'debug channel <x|all>' command back
-
-2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac, acinclude.m4: don't display the
- --with-imap message unless --with-imap was specified without a
- path use '-n' instead of '! -z' for tests
-
-2007-02-10 01:02 +0000 [r53783-53821] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Add some output for "show application
- SLAStation/SLATrunk"
-
- * channels/chan_sip.c: Change some text to properly state "On
- Hold", which was already done in trunk.
-
- * configs/sla.conf.sample, include/asterisk/app.h,
- include/asterisk/utils.h, main/dial.c, apps/app_meetme.c,
- channels/chan_sip.c, doc/sla.txt (added),
- include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge
- team/russell/sla_rewrite This is a completely new implementation
- of the SLA functionality introduced in Asterisk 1.4. It is now
- functional and ready for testing. However, I will be adding some
- additional features over the next week, as well. For information
- on how to set this up, see configs/sla.conf.sample and
- doc/sla.txt. In addition to the changes in app_meetme.c for the
- SLA implementation itself, this merge brings in various other
- changes: chan_sip: - Add the ability to indicate HOLD state in
- NOTIFY messages. - Queue HOLD and UNHOLD control frames even if
- the channel is not bridged to another channel. linkedlists.h: -
- Add support for rwlock based linked lists. dial.c: - Add the
- ability to run ast_dial_start() without a reference channel to
- inherit information from.
-
- * apps/app_echo.c: When the Echo() application receives the digit
- '#', echo that back as well. Since we already sent the BEGIN
- frame for that digit, it makes sense to send the END as well.
-
-2007-02-09 23:52 +0000 [r53779-53781] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_gtalk.c: another dependency
-
- * apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c,
- funcs/func_odbc.c, res/res_adsi.c: add some inter-module
- dependencies
-
- * build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk
- scripts to work when both MODULEINFO and MAKEOPTS are present in
- a source file
-
-2007-02-09 19:33 +0000 [r53749] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Temporarily change musicclass on channel to one
- specified in Dial so that the 'm' option functions properly.
- (issue #8969 reported by christianbee)
-
-2007-02-09 16:42 +0000 [r53715] Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/imapstorage.txt, configure, configure.ac: clarify the fact
- that voicemail IMAP storage cannot be built against a distro's
- binary c-client library package (at least not at this time)
-
-2007-02-08 23:18 +0000 [r53672] Olle Johansson <oej@edvina.net>
-
- * main/acl.c: Don't output debug unless we asked for it
-
-2007-02-08 17:54 +0000 [r53601] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Fix timeout issue when utterance is
- longer then timeout itself.
-
-2007-02-08 13:47 +0000 [r53530-53532] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/loader.c: Issue 9007 - Mutex not released on early return
-
- * apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007)
- | 2 lines Issue 9003 - If fullname is empty, quote() passes back
- "\"" ........
-
-2007-02-07 23:52 +0000 [r53464-53497] Russell Bryant <russell@digium.com>
-
- * main/db1-ast/Makefile: When building libdb1.a, put the additional
- flags needed at the beginning of ASTCFLAGS, instead of at the
- end. This way, we ensure that we find the local headers first
- before accidentally trying to use headers that exist in locations
- specified in the ASTCFLAGS passed from the main Makefile. (issue
- #8637, ovi)
-
- * main/Makefile: The clean target actually needs to run "distclean"
- on editline. This is because we need to make sure that its
- configure script gets executed again, because the CFLAGS we want
- to pass to editline may have changed.
-
-2007-02-07 17:53 +0000 [r53434] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: We can not reliably do P2P bridging with DTMF passing
- back with compensation if we need to listen for DTMF frames.
- (issue #8962 reported by caio1982)
-
-2007-02-07 17:39 +0000 [r53429] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: When parsing the NTP timestamp in a sender report
- message, you are supposed to take the low 16 bits of the integer
- part, and the high 16 bits of the fractional part. However, the
- code here was erroneously taking the low 16 bits of the
- fractional part. It then shifted the result 16 bits down, so the
- result was always zero. This fix makes it grab the appropriate
- high 16 bits, instead. (issue #8991, pointed out by
- andre_abrantes)
-
-2007-02-07 17:04 +0000 [r53358-53399] Joshua Colp <jcolp@digium.com>
-
- * apps/app_playback.c: Directly load say.conf in load_module
- instead of calling the reload function. (issue #8946 reported by
- junky)
-
- * /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2
- lines Fix a few potential memory leaks with realtime users and
- peers. (issue #8999 reported by bsmithurst) ........
-
-2007-02-07 15:33 +0000 [r53355] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_macro.c: Merged revisions 53354 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007)
- | 2 lines Issue 7440 - Macro called from Macro from the h
- extension exits prematurely ........
-
-2007-02-07 09:22 +0000 [r53324] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 52843 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) |
- 1 line fixed some possible segfaults. also fixed an very
- important bug which occurs on high load (when calls are very fast
- generated) ........
-
-2007-02-07 05:24 +0000 [r53246-53294] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_jabber.c: Text fix for jabber reload command (reported by
- bkruse via IRC)
-
- * main/manager.c, /: Merged revisions 53245 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007)
- | 2 lines Issue 8987 - Status could return two responses
- (mnicholson) ........
-
-2007-02-05 23:43 +0000 [r53222] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Formatting
-
-2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp <jcolp@digium.com>
-
- * apps/app_playback.c: Ensure say_cfg is NULL when the module is
- loaded. (issue #8946 reported by junky)
-
- * apps/app_playback.c: Unregister Playback CLI commands as well as
- dialplan application. (issue #8946 reported by junky)
-
-2007-02-05 00:18 +0000 [r53143] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Add some comments on queue system behaviour
- and how it affects the SIP channel
-
-2007-02-03 21:05 +0000 [r53138] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make SIPDtmfMode application work with
- recent capability changes, and also fix an RTP stack issue when
- the auto option was used. (issue #8972 reported by mdu113)
-
-2007-02-03 20:44 +0000 [r53135-53136] Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 53133 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) |
- 4 lines 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) ........
-
-2007-02-03 10:02 +0000 [r53131] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string
- because due to compatibilities with CS1000 reported at
- www.voip-info.org
-
-2007-02-02 21:26 +0000 [r53129] BJ Weschke <bweschke@btwtech.com>
-
- * UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a
- warning to the console that things might possibly be
- misconfigured when queue member's states are still 'Not in Use'
- when we're about to bridge them with a caller from queue. Also,
- put some documentation quoted from oej's queues.txt efforts
- started in /trunk today. This commit puts #7433 into feedback
- state for 1.4, and pending no further negative feedback, it will
- finally be closed.
-
-2007-02-02 17:15 +0000 [r53114-53120] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Correct a copy/pasted error message line for RTCP.
-
- * main/config.c, /: Merged revisions 53117 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2
- lines 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) ........
-
- * Makefile: Add systemname to asterisk.conf generation per recent
- discussions about it. (issue #8968 reported by blitzrage)
-
-2007-02-02 00:24 +0000 [r53109] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, configs/sip.conf.sample: Disable the direct
- p2p RTP call setup in SIP. You can enable it in sip.conf, but it
- is now considered experimental until we solve the
- AST_CONTROL_ANSWER with payload and videocaps stuff.
-
-2007-02-01 22:24 +0000 [r53097-53104] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2
- lines Copy noncodeccapability over to the joint variable so that
- telephone-event will get transmitted in the sent INVITE. ........
-
- * main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile
- here as well, but it apparently required both dev mode and no
- optimizations to creep up.
-
- * /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2
- lines Don't negotiate RFC2833 when not configured to do so.
- (issue #8799 reported by mdu113) ........
-
-2007-02-01 21:24 +0000 [r53093] Russell Bryant <russell@digium.com>
-
- * funcs/func_strings.c: Fix the FIELDQTY function to not crash.
- (reported by blitzrage and Corydon on IRC)
-
-2007-02-01 21:15 +0000 [r53091] Olle Johansson <oej@edvina.net>
-
- * /: Going backwards, blame file.
-
-2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb
- 2007) | 2 lines Return previous behavior of having MOH pick up
- where it was left off. (issue #8672 reported by sinistermidget)
- ........
-
- * funcs/func_strings.c: Make func_strings build under dev mode.
- Didn't I do this today already in the berkeley DB?
-
-2007-02-01 21:05 +0000 [r53079-53085] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: - Clean INC_COUNT flag when we decrement
- call counter - If it's still set at time of dialog destruction,
- make sure we decrement the device call counter properly before we
- destroy the dialog
-
- * apps/app_queue.c: Change debug level for state change message
- that is not really informative when debugging app_queue
-
- * channels/chan_sip.c: Cleaning up the devicestate callback
- function
-
-2007-02-01 20:13 +0000 [r53075-53077] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_strings.c: Oops.
-
- * /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007)
- | 2 lines Bug 8965 ........
-
-2007-02-01 19:33 +0000 [r53072] Joshua Colp <jcolp@digium.com>
-
- * main/asterisk.c: Add missing 'F' letter to getopt so it magically
- becomes a valid option. (issue #8960 reported by tzafrir)
-
-2007-02-01 19:21 +0000 [r53070] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007)
- | 2 lines No wonder FIELDQTY doesn't work with functions... the
- documentation in pbx.c was wrong ........
-
-2007-02-01 17:37 +0000 [r53064] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix silly logic. We really want to write
- UDPTL frames out when the call is up.
-
-2007-02-01 16:35 +0000 [r53062] Olle Johansson <oej@edvina.net>
-
- * configs/sip.conf.sample: Add explanation of port= in combination
- with defaultip= (thanks jsmith)
-
-2007-02-01 13:17 +0000 [r53060] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: we update the name on any first reply of
- our setup
-
-2007-02-01 11:07 +0000 [r53057] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c: chan_h323 is very stable, so let it built
- by default
-
-2007-02-01 00:24 +0000 [r53050-53052] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: When going on hold have the side that was put on hold
- reinvite back to Asterisk. When going off hold have the side that
- was taken off hold reinvited back to the other party.
-
- * main/rtp.c: Add more frame types to forward in the RTP bridge
- loops.
-
-2007-01-31 21:32 +0000 [r52859-53046] Russell Bryant <russell@digium.com>
-
- * main/cdr.c, main/manager.c, pbx/pbx_spool.c,
- channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
- pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c,
- main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c,
- channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c:
- Merged revisions 53045 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) |
- 3 lines Fix a bunch of places where pthread_attr_init() was
- called, but pthread_attr_destroy() was not. ........
-
- * apps/app_userevent.c: Remove an extra \r\n from manager user
- events. (issue #8955, mnicholson)
-
- * main/rtp.c, /: Merged revisions 53039 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) |
- 3 lines Use the proper format string to print unsigned values in
- the rtp debug output. (issue #8954, wmis) ........
-
- * apps/app_queue.c: Only changed the paused status in an existing
- queue member if the paused column exists.
-
- * apps/app_queue.c: Instead of always creating a realtime queue
- member as unpaused, read the "paused" column and use that value
- for the paused status of the member. (issue #8949, jmls)
-
- * contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10.
- (issue #8363, johnlange)
-
- * doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue
- #8942, lters)
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- codecs/codec_gsm.c: When we are checking for a system installed
- version of libgsm, we need to check for gsm.h as well.
- Furthermore, when checking for this header, it may be located in
- a gsm/ sub directory, so check for that, as well. (issue #8773)
-
- * channels/chan_sip.c: Only set the DTMF flag on the rtp structure
- if the DTMF mode is actually RFC2833, not just that it is not
- INFO. This makes it get set for inband DTMF as well, which is not
- valid. (issue #8936)
-
- * main/asterisk.c, /: Merged revisions 52903 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) |
- 9 lines 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. ........
-
-2007-01-30 15:29 +0000 [r52856] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Drop the deprecated show commands since the
- original ones were changed back. (issue #8937 reported by
- PCadach)
-
-2007-01-30 08:46 +0000 [r52807-52809] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c: Revert reprecation of h.323 gk cycle
- command from pre-1.4 version instead of duplicated h323 cycle gk
-
- * res/res_odbc.c: Don't play with free()'d pointers
-
- * configure, acinclude.m4: Handle non-standard OpenH323/PWLib
- library names
-
-2007-01-30 00:15 +0000 [r52763] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) |
- 5 lines 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 23:43 +0000 [r52717] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan
- 2007) | 2 lines 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 22:58 +0000 [r52688-52695] Russell Bryant <russell@digium.com>
-
- * main/Makefile: Add a missing quotation mark. This was pointed out
- by jcmoore on #asterisk-dev.
-
- * main/manager.c: Remove a recursive lock of the manager session.
- This was pointed out by zandbelt in issue #8711.
-
-2007-01-29 22:12 +0000 [r52679] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * pbx/pbx_config.c: Argument number correction
-
-2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant <russell@digium.com>
-
- * main/Makefile: ASTLDFLAGS needs to be passed to the editline
- configure script as LDFLAGS. (issue #8928, zandbelt)
-
- * main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF
- mode translation. P2P bridging can only be used when the DTMF
- modes don't match if the core is monitoring DTMF in both
- directions. Then, the core will handle the translation.
- Otherwise, this bridging method can not be used. (issue #8936)
-
- * main/manager.c: The session lock can not be held while calling
- action callbacks. If so, then when the WaitEvent callback gets
- called, then no event can happen because the session can't be
- locked by another thread. Also, the session needs to be locked in
- the HTTP callback when it reads out the output string. This fixes
- the deadlock reported in both 8711 and 8934. Regarding issue
- 8711, there still may be an issue. If there is a second action
- requested before the processing of the first action is finished,
- there could still be some corruption of the output string buffer
- used to build the result. (issue #8711, #8934)
-
-2007-01-29 18:59 +0000 [r52572] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Use ast_calloc instead of malloc.
-
-2007-01-29 17:57 +0000 [r52535] Steve Murphy <murf@digium.com>
-
- * apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR
- backport to 1.4). It was committed to trunk via 7663. But it
- wasn't so much an enhancement as a fix for the bad language
- output for portuguese in Brazil, so, after a lot of prodding from
- patient Brazilians, here is the same fix for 1.4
-
-2007-01-29 17:33 +0000 [r52523] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Set quota information to 0 when creating a
- vm_state. (issue #8924 reported by neutrino88)
-
-2007-01-29 16:54 +0000 [r52506] Russell Bryant <russell@digium.com>
-
- * main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in
- the last commit to the adaptive jitterbuffer code. - Specifically
- indicate to the compiler that the "dropem" variable only needs
- one but. - Change formatting to conform to coding guidelines.
-
-2007-01-29 04:18 +0000 [r52494] Jim Dixon <telesistant@hotmail.com>
-
- * main/jitterbuf.c, include/jitterbuf.h: Fixed problem with
- jitterbuf, whereas it would not complain about, and would allow
- itself to be overfilled (per the max_jitterbuf parameter). Now it
- rejects any data over and above that size, and complains about
- it.
-
-2007-01-28 05:15 +0000 [r52462] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * configure, configure.ac: Suggested change to fix normal usage of
- --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing
- list)
-
-2007-01-27 02:13 +0000 [r52335-52416] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 52415 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2
- lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log
- follow documentation. (issue #7677 reported by amilcar) ........
-
- * main/manager.c: Have the manager interface send back an "Already
- logged in" message instead of "Invalid/Unknown Command" when the
- client authenticates for a second time. (issue #8509 reported by
- pari)
-
- * /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2
- lines Make the last context entry read in the dominant one.
- (issue #8918 reported by pj) ........
-
- * main/file.c: Fix core show file formats CLI command.
-
-2007-01-25 19:18 +0000 [r52163-52265] Joshua Colp <jcolp@digium.com>
-
- * /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2
- lines Allow dequeueing of frames with negative timestamp by
- moving jitterbuffer frames check to jb_next. (issue #8546
- reported by harmen) ........
-
- * channels/chan_sip.c: Drop out variables I accidentally put in.
-
- * channels/chan_sip.c: Decrement onHold count if we are hung up on
- and still on hold. (issue #8909 reported by alexh42)
-
- * apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan
- 2007) | 2 lines Add another note about audio files being played
- back to each bridged party. (issue #8718 reported by ppyy)
- ........
-
-2007-01-25 01:37 +0000 [r52107-52160] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, configs/users.conf.sample: By suggestion
- from kpfleming last week, change "vmpassword" to "vmsecret".
-
- * configure, configure.ac: Remove libnsl as a required lib for
- libiksemel to work. This change was already made in the trunk.
- (issue #8762)
-
- * include/asterisk/dial.h: Fix the formatting of doxygen comments
- to properly indicate that the comment documents the previous
- entity, as opposed to the next one.
-
-2007-01-24 18:26 +0000 [r52052] Steve Murphy <murf@digium.com>
-
- * utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1
- line 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 18:20 +0000 [r52016-52049] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c (added), apps/app_page.c, main/Makefile,
- include/asterisk/dial.h (added): Merge in dialing API and the
- app_page that uses it. (issue #BE-118)
-
- * channels/chan_sip.c: Fix changing channel formats when joint
- capability changes and there are no audio formats... I didn't
- break it originally! (issue #8535 reported by ivoc)
-
-2007-01-24 17:14 +0000 [r52000] Russell Bryant <russell@digium.com>
-
- * configure: rebuild configure script to reflect last chan_h323
- related changes.
-
-2007-01-24 12:57 +0000 [r51979-51989] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: added fix from #8899
-
- * channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24
- Jan 2007) | 1 line fixed the busy problem (dialstatus was not
- busy when we called a busy extension) ........
-
-2007-01-24 09:30 +0000 [r51931] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Show capabilities *and* preference in
- general settings in "sip show settings" (reported by Clona/Telio
- - Thanks!)
-
-2007-01-24 08:04 +0000 [r51895] Paul Cadach <paul@odt.east.telecom.kz>
-
- * acinclude.m4: Allow x64 builds of H.323 (please, rebuild
- configure)
-
-2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 51843 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) |
- 6 lines 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, /: Merged revisions 51828 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) |
- 4 lines 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 22:46 +0000 [r51788] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c,
- channels/chan_sip.c, channels/chan_skinny.c,
- channels/chan_features.c, channels/chan_alsa.c,
- channels/chan_gtalk.c, channels/chan_iax2.c: Update channel
- drivers to use module referencing so that unloading them while in
- use will not result in crashes. (issue #8897 reported by junky)
-
-2007-01-23 22:04 +0000 [r51750-51781] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Fix some bugs in process_message(). The manager
- session lock needs to be held when sending some sort of response,
- or calling one of the manager action callbacks. This resolves an
- issue where people using the GUI would get random crashes when
- they start clicking around a lot. (issue #8711, reported and
- debugged by zandbelt)
-
- * main/http.c: Fix setting the default port of 8088 on 64-bit or
- big-endian machines.
-
- * main/manager.c: When traversing the list of manager actions, the
- iterator needs to be initialized to the list head *after* locking
- the list. Also, lock the actions list in one place it is being
- accessed where it was not being done.
-
-2007-01-23 20:32 +0000 [r51683-51716] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: this mod from 8593 (dstchannel in cdr is
- empty when transfer call).
-
- * main/callerid.c: via 8748 (callerid.c loses name when returning
- PRIVATE_NUMBER flag), the user suggested this mod, saying it
- would allow 'WITHHELD' to appear in the name field, which would
- be useful
-
-2007-01-23 10:28 +0000 [r51648-51649] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
- channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) |
- 6 lines * 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) ........ r50506 | crichter |
- 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get
- L2 UP, the L1 is UP definitely too, so we set the L1 state up as
- well. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c: manually merged r49922 and r50335, because
- of conflicts. this commint includes addition of the ISDN RESTART
- Message
-
-2007-01-23 06:51 +0000 [r51615] Paul Cadach <paul@odt.east.telecom.kz>
-
- * channels/chan_h323.c, channels/Makefile: Do not abort Asterisk
- startup if h323 configuration file not found (reported by
- mithraen)
-
-2007-01-23 03:00 +0000 [r51513-51558] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Only change audio formats on the channel if
- we have an audio format to change to. (issue #8535 reported by
- ivoc)
-
- * /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan
- 2007) | 2 lines 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 19:28 +0000 [r51409] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: This fixes 8836, according to dnatural
-
-2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan
- 2007) | 2 lines 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) ........
-
-2007-01-20 06:53 +0000 [r51348-51350] Jason Parker <jparker@digium.com>
-
- * configs/say.conf.sample: Fix Italian numeral support in say.conf
- for "_[2-9]00" case. "2131" would've translated to something
- along the lines of (pardon my..Italian {or lack thereof})
- "duecentocentotrentuno", which makes no sense at all.
-
- * configs/say.conf.sample: Fix German language support in say.conf
- Properly support 21, 31, 41, 51, 61, 71, 81, and 91.
- einundzwanzig has the same format as zweiundzwanzig (as do all
- other "_ZX" spoken numerals) Fix support for numbers in the
- 10,000,000 to 99,999,999 range. Add support for numbers in the
- 100,000,000 to 999,999,999 range.
-
-2007-01-20 00:13 +0000 [r51302-51343] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Remove an unused instance of an unnamed enum.
-
- * apps/app_meetme.c: Remove another duplicated definition
-
- * apps/app_meetme.c: Remove a variable that was declared twice.
-
- * codecs/gsm/Makefile: Add a couple more processors that need
- optimizations excluded. (issue #8637)
-
- * channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk.
- AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same
- thing. So, a digit would have been interpreted incorrectly here.
- Since the channel driver will always have the begin and end
- callbacks called for a digit, only support the button-down and
- button-up messages.
-
- * .cleancount: Bump the cleancount since my last commit changed the
- channel structure.
-
- * channels/chan_oss.c, main/rtp.c, main/channel.c,
- channels/chan_phone.c, channels/chan_misdn.c,
- channels/chan_skinny.c, channels/chan_features.c,
- channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c,
- channels/chan_zap.c, channels/chan_local.c, main/frame.c,
- channels/chan_sip.c, channels/chan_agent.c,
- include/asterisk/channel.h, channels/chan_gtalk.c,
- channels/chan_iax2.c: Merge the changes from the
- /team/group/vldtmf_fixup branch. The main bug being addressed
- here is a problem introduced when two SIP channels using SIP INFO
- dtmf have their media directly bridged. So, when a DTMF END frame
- comes into Asterisk from an incoming INFO message, Asterisk would
- try to emulate a digit of some length by first sending a DTMF
- BEGIN frame and sending a DTMF END later timed off of incoming
- audio. However, since there was no audio coming in, the DTMF_END
- was never generated. This caused DTMF based features to no longer
- work. To fix this, the core now knows when a channel doesn't care
- about DTMF BEGIN frames (such as a SIP channel sending INFO
- dtmf). If this is the case, then Asterisk will not emulate a
- digit of some length, and will instead just pass through the
- single DTMF END event. Channel drivers also now get passed the
- length of the digit to their digit_end callback. This improves
- SIP INFO support even further by enabling us to put the real
- digit duration in the INFO message instead of a hard coded 250ms.
- Also, for an incoming INFO message, the duration is read from the
- frame and passed into the core instead of just getting ignored.
- (issue #8597, maybe others...)
-
- * main/asterisk.c: Merged revisions 51300 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) |
- 4 lines 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) ........
-
-2007-01-19 00:17 +0000 [r51272-51274] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/chan_zap.c: chan_zap compiles without libpri after
- committing 7877 patch
-
- * channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007)
- | 3 lines 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 23:36 +0000 [r51270] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: block this patch since it is already here
-
-2007-01-18 22:50 +0000 [r51265] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, main/channel.c, main/pbx.c,
- funcs/func_strings.c, main/app.c: Add some more checks for
- option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832,
- patch(es) by tgrman
-
-2007-01-18 21:54 +0000 [r51262] Russell Bryant <russell@digium.com>
-
- * Makefile, configure, main/Makefile, acinclude.m4, makeopts.in:
- Ensure that the locations given to the Asterisk configure script
- for ncurses, curses, termcap, or tinfo are further passed along
- to the editline configure script. This fixes some
- cross-compilation environments. (issue #8637, reported by ovi,
- patch by me)
-
-2007-01-18 21:14 +0000 [r51256] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18
- Jan 2007) | 2 lines If a timezone is not specified, assume
- localtime (instead of gmtime) (Issue #7748) ........
-
-2007-01-18 19:17 +0000 [r51251] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Only start timeout once we reach the end
- of the files to play back.
-
-2007-01-18 18:42 +0000 [r51245] Jason Parker <jparker@digium.com>
-
- * main/cli.c: Fix an issue with file name completion in "module
- load" and "load". Issue 8846
-
-2007-01-18 18:36 +0000 [r51243] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Copy MOH settings when calling a peer so
- that if they put someone on hold or get put on hold themselves
- they get the right music class. (issue #8840 reported by mdu113)
-
-2007-01-18 18:28 +0000 [r51241] Jason Parker <jparker@digium.com>
-
- * main/channel.c: Fix an issue with deprecated commands
-
-2007-01-18 17:49 +0000 [r51236] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18
- Jan 2007) | 2 lines Document all the fields, including the
- indication that "uniqueid" should not be renamed. ........
-
-2007-01-18 17:18 +0000 [r51233] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Make the "hasmanager" option in users.conf
- actually have an effect. (issue #8740, LnxPrgr3)
-
-2007-01-18 00:48 +0000 [r51211-51213] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Build the IMAP remote directory string
- better and properly. Fix an issue with encoding the GSM voicemail
- when attaching to the voicemail. (issue #8808 reported by
- akohlsmith)
-
- * main/rtp.c: Pass data as well for hold/unhold/vidupdate frames.
- (issue #8840 reported by mdu113)
-
-2007-01-17 23:31 +0000 [r51198-51205] Russell Bryant <russell@digium.com>
-
- * funcs/func_odbc.c: Fix some instances where when loading
- func_odbc, a double-free could occur. Also, remove an unneeded
- error message. If the failure condition is actually a memory
- allocation failure, a log message will already be generated
- automatically.
-
- * channels/chan_zap.c: Instead of dividing the offset by 2
- directly, make it more clear that the offset is being scaled by
- the size of the elements in the buffer. (Inspired by a discussing
- on the asterisk-dev list about this code)
-
- * /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) |
- 3 lines 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 20:56 +0000 [r51195] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, main/utils.c: Merged revisions 51194 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007)
- | 4 lines 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). ........
-
-2007-01-17 17:36 +0000 [r51186] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: re-add "password" for realtime voicemail
-
-2007-01-17 06:36 +0000 [r51182] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Return the correct result when directly writing out a
- packet so that the core doesn't then decide to handle it the
- regular way again. (issue #8833 reported by rcourtna)
-
-2007-01-17 01:29 +0000 [r51176] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_voicemail.c: a few more coding style cleanups and one
- bug fix (from AnthonyL)
-
-2007-01-17 00:46 +0000 [r51172] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Move rescheduling of lagrq/pings into the
- scheduler callback.
-
-2007-01-17 00:20 +0000 [r51165-51170] Jason Parker <jparker@digium.com>
-
- * main/rtp.c: Fix issue with dtmf continuation packets when the
- dtmf digit is 0... Issue 8831
-
- * apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with
- IMAP storage and realtime voicemail. Also update the vmdb sql
- script for IMAP specific options. Issue 8819, initial patches by
- bsmithurst (slightly modified by me)
-
- * doc/voicemail_odbc_postgresql.txt: change documentation to
- reflect new procedure in 1.4/trunk
-
-2007-01-16 21:51 +0000 [r51159-51162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions
- 51161 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007)
- | 2 lines Add documentation walkthrough on getting Postgres to
- work with voicemail (from Issue 8513) ........
-
- * apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007)
- | 2 lines Postgres driver doesn't like a NULL pointer when
- retrieving the length (Bug 8513) ........
-
-2007-01-16 17:46 +0000 [r51150] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c: minor things i missed before i get jumped
- on
-
-2007-01-16 17:39 +0000 [r51148] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_features.c: Merged revisions 51145 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2
- lines 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)
- ........
-
-2007-01-16 17:36 +0000 [r51146] Jason Parker <jparker@digium.com>
-
- * main/file.c: Display more useful output when streaming files.
- Include the channel name to which the file is being played. Issue
- 8828, patch by junky.
-
-2007-01-16 05:55 +0000 [r51087] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2
- lines 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) ........
-
-2007-01-16 01:15 +0000 [r51057] Russell Bryant <russell@digium.com>
-
- * main/config.c: It is possible for the config pointer to be NULL
- here, so it needs to be checked before dereferencing it.
-
-2007-01-16 00:22 +0000 [r51030] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c, configs/users.conf.sample: Patch allows for
- changing voicemail password in users.conf from voicemail main,
- written by AnthonyL bug #8436
-
-2007-01-15 23:49 +0000 [r50994] Russell Bryant <russell@digium.com>
-
- * Makefile.rules: Filter out a few CFLAGS that are not valid
- CXXFLAGS.
-
-2007-01-15 21:08 +0000 [r50957] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946
- | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4
- lines Solves issue with forwarding voicemails from folders other
- than inbox. patch by anthonyl. ........
-
-2007-01-15 18:23 +0000 [r50921] Jason Parker <jparker@digium.com>
-
- * main/asterisk.c: re-add deprecated "show version" CLI command.
-
-2007-01-15 16:36 +0000 [r50895] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Move event processing into do_message so that it
- gets executed again when events are tripped.
-
-2007-01-15 15:03 +0000 [r50867] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, main/Makefile,
- configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the
- ACX_PTHREAD macro from the Autoconf macro archive for setting up
- compiler pthreads support... should improve portability to
- platforms with unusual pthreads requirements
-
-2007-01-14 21:59 +0000 [r50820] Joshua Colp <jcolp@digium.com>
-
- * main/astmm.c: Add missing newlines for two memory CLI commands.
-
-2007-01-14 05:13 +0000 [r50782] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c,
- main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c,
- main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c,
- main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c,
- main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c,
- main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c,
- main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c,
- main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c,
- main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c,
- main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h,
- main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c,
- main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c,
- main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c,
- main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c,
- main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c,
- main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13
- Jan 2007) | 2 lines Bug 8814 - db should look for its header
- using a relative path, instead of the system path (Fixes FreeWRT)
- ........
-
-2007-01-13 16:45 +0000 [r50754] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, build_tools/make_sample_voicemail (added): when
- building the sample greetings for maibox 1234@default during
- 'make samples', build a greeting for each language and file
- format the user selected to install with menuselect (reported by
- Brian Capouch on asterisk-dev)
-
-2007-01-13 06:00 +0000 [r50674-50727] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Only write a frame out to the channel if one
- exists. There are cases where one may not and would therefore
- cause the channel driver to segfault. (issue #8434 reported by
- slimey)
-
- * res/res_snmp.c: Only join the snmp thread on an unload if the
- thread is actually running. (issue #8810 reported by junky)
-
-2007-01-12 19:24 +0000 [r50647] Jason Parker <jparker@digium.com>
-
- * configs/voicemail.conf.sample: Update documentation to state that
- you shouldn't use realtime static with voicemail.conf
-
-2007-01-12 16:42 +0000 [r50602] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: We need to check for res being 0 in do_message
- itself, otherwise our headers will get lost.
-
-2007-01-12 14:42 +0000 [r50562] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/pbx.c, /: Merged revisions 50561 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007)
- | 2 lines minor documentation clarification ........
-
-2007-01-11 05:53 +0000 [r50377-50468] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Remove check for channel state as it can
- definitely be something other then ring, and also clean up the
- code a bit. This should solve the parking issues and maybe some
- attended transfer issues people have been seeing.
-
- * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add
- support to see whether NAT was detected (yay symmetric RTP) and
- also add a check in chan_sip so that if NAT has been detected and
- the reinvite behind nat option has been turned off, then just do
- partial bridge. (issue #8655 reported by mnicholson)
-
- * apps/app_speech_utils.c: Merge speech-multi branch which adds
- support for joining multiple sound files together to be played
- one after another in SpeechBackground.
-
- * main/config.c: Fix parsing when using something like ldap
- settings. (done by anthonyl)
-
- * channels/chan_sip.c: Fix chan_sip not working issue. Let's not
- prematurely return 0. (issue #8783 reported by st41ker)
-
-2007-01-10 16:45 +0000 [r50346] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made
- it fail to load if the config file existed. Issue 8777
-
-2007-01-10 04:55 +0000 [r50266-50298] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 50295 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2
- lines 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) ........
-
- * main/pbx.c: Ensure data's existence before trying to access it.
- (issue #8774 reported by rcourtna)
-
-2007-01-10 02:17 +0000 [r50228] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 50227 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) |
- 6 lines 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 17:11 +0000 [r50186] Jason Parker <jparker@digium.com>
-
- * main/cli.c: Re-add CLI command that should have only been
- deprecated in 1.4. Thanks kshumard! (reported in person, so no
- associated issue #)
-
-2007-01-09 13:40 +0000 [r50151] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007)
- | 4 lines 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) ........
-
-2007-01-09 11:25 +0000 [r50124] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: - handle re-invites properly in sip_hangup()
- - Add some invitestate status changes just to be sure
-
-2007-01-08 23:39 +0000 [r50098] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Fix an issue with voicemail and users.conf,
- where it wouldn't ever parse a password, since it was using
- "secret" instead of "password" Issue 8761, reported by and patch
- suggestion from ssokol.
-
-2007-01-08 21:11 +0000 [r50073] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_senddtmf.c: we can't unlock a channel if we cant find
- it. - AnthonyL bug #8741
-
-2007-01-08 18:21 +0000 [r50032] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Disable the more intense packet2packet bridging until
- the bugs can be worked out.
-
-2007-01-08 14:26 +0000 [r49925-50006] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Issue #8677 - Handle failure of T.38
- re-invite This is not a fix, but adding an error message to tell
- the admin that we have a bad configuration. We should not send
- T.38 re-invites to devices that can't handle it (with the current
- architecture where you have to hard-code t.38 support per
- device). To really fix this, we need to figure out a way to tell
- the incoming call that the re-invite failed, so we can signal
- failure on that end and go back to the original call.
-
- * channels/chan_sip.c: Issue #8524, support multiple via header
- values (tardieu) Thanks!
-
- * channels/chan_sip.c: We only need one forward declaration
-
- * channels/chan_sip.c: Issue 8735: Terminate state when extension
- is unavailable for subscription
-
-2007-01-08 05:11 +0000 [r49890] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2
- lines Ensure we use the default refresh value of 60 if the remote
- server does not send one. (issue #8746 reported by maethor)
- ........
-
-2007-01-08 03:53 +0000 [r49866] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac: since we use AC_PATH_TOOL to find tools,
- we should use the results it provides for us (reported by Brian
- Capouch on the asterisk-dev list)
-
-2007-01-07 21:44 +0000 [r49831-49834] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007)
- | 2 lines If openstream fails, then we crash (Issue 8564)
- ........
-
- * channels/chan_sip.c: Second condition was a subset of the first,
- so hold was never decremented, thus hint stayed stuck (Issue
- 8747)
-
-2007-01-06 00:24 +0000 [r49742] Jason Parker <jparker@digium.com>
-
- * main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping
- byte of allocated memory! This looks like it may have been a
- chicken/egg scenario.. You had to call a cleanup func, because
- everything was allocated. Then since you had to call a cleanup
- func, you were forced to allocate - ie; strdup("").
-
-2007-01-05 23:51 +0000 [r49710-49715] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, acinclude.m4: one more time...
-
- * configure, acinclude.m4: proper fix for r49712
-
- * configure, acinclude.m4: if --with-foo=<path> is specific for a
- configure option, ensure that it is used for header file checking
- as well
-
- * main/manager.c: ast_func_read() needs a writable copy of the
- function name to be passed
-
-2007-01-05 23:16 +0000 [r49705] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and
- chan_zap also depend on zaptel. This fixes an issue (8727) with
- zaptel being in a different directory, using --with-zaptel.
-
-2007-01-05 22:52 +0000 [r49676-49680] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c: don't 'consume' the params list before we try to
- use it again
-
- * res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c,
- main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c,
- main/db.c, channels/chan_zap.c, channels/chan_sip.c,
- apps/app_meetme.c, res/res_features.c, channels/chan_agent.c,
- utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c,
- apps/app_queue.c, res/res_jabber.c: reduce stack consumption for
- AMI and AMI/HTTP requests by nearly 20K in most cases
-
-2007-01-05 22:14 +0000 [r49675] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Don't keep repeating the warning over and over
- when the end of the call is reached. (issue #8724 reported by
- xrg)
-
-2007-01-05 17:09 +0000 [r49581-49636] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_skinny.c,
- channels/chan_iax2.c: Merged revisions 49635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007)
- | 2 lines ensure that threads which are supposed to be detached
- (because we aren't going to wait on them) are created properly
- ........
-
- * channels/chan_iax2.c: revert the dynamic_list insertion change...
- that was not the right thing to do
-
- * channels/chan_iax2.c: create the IAX2 processing threads as
- background threads so they will use smaller stacks when we create
- a dynamic thread, put it on the dynamic_list right away so we
- don't lose track of it
-
-2007-01-04 23:00 +0000 [r49568] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: It's possible for the iax2 pvt to
- disappear, so if it has... don't bother looking for dpentries.
-
-2007-01-04 22:51 +0000 [r49553] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/threadstorage.h, main/asterisk.c,
- build_tools/cflags.xml, include/asterisk.h, main/Makefile,
- main/threadstorage.c (added), main/utils.c: add support for
- tracking thread-local-storage objects that exist via
- 'threadstorage' CLI commands
-
-2007-01-04 22:28 +0000 [r49551] Joshua Colp <jcolp@digium.com>
-
- * main/config.c: Only free comments and line buffer once we reach
- the first level. (issue #8678 reported by ssokol, fixed by
- anthonyl)
-
-2007-01-04 21:58 +0000 [r49460-49536] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/iax2-parser.c, main/frame.c: don't mark these
- allocations as 'cache' allocations when caching has been disabled
-
- * channels/iax2-parser.c: if we're going to decrement the frame
- count when we free a frame, we should inrement it when we create
- one :-)
-
- * channels/iax2-parser.c, channels/iax2-parser.h,
- channels/chan_iax2.c: only do IAX2 frame caching for voice and
- video frames
-
- * main/frame.c: don't do frame header caching in the core if
- LOW_MEMORY is defined
-
- * channels/iax2-parser.c: don't define this type either if
- LOW_MEMORY is enabled
-
-2007-01-04 18:11 +0000 [r49459] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447
- | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2
- lines converted a lot of 256 to PATH_MAX and some white space
- fixes. ........
-
-2007-01-04 18:06 +0000 [r49457-49458] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode
-
- * codecs/Makefile: make building of codec_gsm against the system
- GSM library actually work
-
-2007-01-04 16:50 +0000 [r49413] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412
- | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3
- lines good catch russell sorry i missed that. fix magic number
- with proper sizeof ........
-
-2007-01-04 04:33 +0000 [r49388] Russell Bryant <russell@digium.com>
-
- * funcs/func_realtime.c: Fix the REALTIME() dialplan function.
- ast_build_string() advances the string pointer to the position to
- begin the next write into the buffer. So, this pointer can not be
- used to copy the contents of the string later. The beginning of
- the buffer must be saved. Interestingly enough, this code could
- not have ever worked. (Pointed out by Sebb on IRC, thanks!)
-
-2007-01-03 23:32 +0000 [r49355] Matt O'Gorman <mogorman@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from
- https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354
- | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6
- lines 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 09:06 +0000 [r49313] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c,
- /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
- configs/misdn.conf.sample: Merged revisions
- 48319,48321,48467,48552,48576,49135,49303 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) |
- 1 line changed a few debugs to higher debug levels ........
- r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) |
- 1 line 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. ........ r48467 |
- crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
- 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.
- ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18
- Dez 2006) | 1 line 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. ........ r48576 | crichter |
- 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line 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
- ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02
- Jan 2007) | 1 line 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. ........ r49303 | crichter |
- 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * 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 ........
-
-2007-01-03 03:21 +0000 [r49282] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, Makefile.rules: various Makefile improvements to get
- chan_vpb (and any other C++ modules) to build properly
-
-2007-01-03 01:19 +0000 [r49259] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Check pvt structure presence before passing
- to send_command. This gets rid of the irritating message about a
- packet without pvt structure. This happens because the scheduled
- item is getting cancelled at almost the exact moment it is
- getting executed.
-
-2007-01-02 22:30 +0000 [r49237] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
- pbx/ael/ael.flex: This is a slight modification to Josh's edits
- for #8579; both files edited were the produced by flex