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Russell Bryant c333c2141d importing files for 1.4.12.1 release 
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.12.1@84780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-05 15:27:27 +00:00

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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>
* /: Blocked revisions 76802 via svnmerge ........ r76802 | qwell |
2007-07-24 11:32:04 -0500 (Tue, 24 Jul 2007) | 3 lines Don't
create the Asterisk channel until we are starting the PBX on it.
(ASA-2007-018) ........
* 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)
2007-07-22 21:42 +0000 [r76410] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /: Blocked revisions 76409 via svnmerge ........ r76409 |
tilghman | 2007-07-22 16:39:55 -0500 (Sun, 22 Jul 2007) | 2 lines
We should not use C++ reserved words in API headers (closes issue
#10266) ........
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>
* /: Blocked revisions 74587 via svnmerge ........ r74587 | file |
2007-07-11 14:15:11 -0300 (Wed, 11 Jul 2007) | 2 lines Use some
Makefile magic to determine if linux/compiler.h is present.
(issue #10174 reported by francesco_r) ........
* 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.
* /: Blocked revisions 74165 via svnmerge ........ r74165 | russell
| 2007-07-09 16:00:17 -0500 (Mon, 09 Jul 2007) | 4 lines When the
specified class isn't found, properly fall back to the channel's
music class or the default. (issue #10123, reported by blitzrage,
patches from juggie, qwell, and me) ........
* 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 17:59 +0000 [r72925] Jason Parker <jparker@digium.com>
* /: Blocked revisions 72924 via svnmerge ........ r72924 | qwell |
2007-07-02 12:58:25 -0500 (Mon, 02 Jul 2007) | 4 lines Fix an
issue with playing "oclock" multiple times in French with 24 hour
time format. Issue 10101 ........
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 16:31 +0000 [r72630] Russell Bryant <russell@digium.com>
* /: Blocked revisions 72629 via svnmerge ........ r72629 | russell
| 2007-06-29 11:30:56 -0500 (Fri, 29 Jun 2007) | 4 lines Backport
changes that make chan_iax2 not start the PBX on an incoming
channel until the three-way call setup is completed. These
changes are already in 1.4 and trunk. ........
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>
* /: Blocked revisions 72184 via svnmerge ........ r72184 | qwell |
2007-06-27 13:40:15 -0500 (Wed, 27 Jun 2007) | 4 lines Fix
another problem in voicemail with missing symbols. Issue 10074,
patch by kryptolus, extended to include #if 0'd blocks (just in
case) ........
* 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 17:49 +0000 [r71848] Jason Parker <jparker@digium.com>
* /: Blocked revisions 71847 via svnmerge ........ r71847 | qwell |
2007-06-26 12:49:14 -0500 (Tue, 26 Jun 2007) | 4 lines Don't try
to install an init script that doesn't exist. Reported to me on
#asterisk on Freenode IRC. ........
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 16:05 +0000 [r71128] Joshua Colp <jcolp@digium.com>
* /: Blocked revisions 71124 via svnmerge ........ r71124 | file |
2007-06-22 12:02:40 -0400 (Fri, 22 Jun 2007) | 2 lines Send an
unhold indication when going off hold. (issue #10036 reported by
speedy) ........
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 16:46 +0000 [r69991] Russell Bryant <russell@digium.com>
* /: Blocked revisions 69990 via svnmerge ........ r69990 | russell
| 2007-06-19 11:45:37 -0500 (Tue, 19 Jun 2007) | 12 lines
Backport fix for crashes related to subscriptions from 1.4 ...
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 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)
* /: Blocked revisions 60016 via svnmerge ........ r60016 | russell
| 2007-04-03 18:23:23 -0500 (Tue, 03 Apr 2007) | 3 lines Add a
missing "\r\n" in the body of the NOTIFY that is sent to indicate
the status of a transfer. (issue #9388, reported by rarritt)
........
* /: Blocked revisions 60014 via svnmerge ........ r60014 | russell
| 2007-04-03 18:00:10 -0500 (Tue, 03 Apr 2007) | 3 lines Use the
more generic check for "sed -r" support that was already present
in 1.4. (related to issue #9399) ........
* /: Blocked revisions 60012 via svnmerge ........ r60012 | russell
| 2007-04-03 17:54:49 -0500 (Tue, 03 Apr 2007) | 3 lines On
Darwin, the -r argument to sed is not valid. It has to be -E.
(issue #9399, reported by jcovert) ........
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) ........
* /: Blocked revisions 59355 via svnmerge ........ r59355 | russell
| 2007-03-29 12:10:28 -0500 (Thu, 29 Mar 2007) | 3 lines Backport
the change to chan_iax2 to return NULL instead of a "null frame"
from its read callback. See revision 59341 to the 1.4 branch for
more info. ........
* 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.) ........
* /: Blocked revisions 59299 via svnmerge ........ r59299 |
tilghman | 2007-03-29 10:33:10 -0500 (Thu, 29 Mar 2007) | 2 lines
Change ENV section to use setenv, instead of putenv (Alexandru
Pirvulescu <sigxcpu@gmail.com>, reported via -dev list) ........
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 16:52 +0000 [r58833] Joshua Colp <jcolp@digium.com>
* /: Blocked revisions 58832 via svnmerge ........ r58832 | file |
2007-03-12 12:49:49 -0400 (Mon, 12 Mar 2007) | 2 lines We can't
use the assembler version of fetchadd_int under Intel Macs.
(issue #9254 reported by darrell budic) ........
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>
* /: Blocked revisions 58167 via svnmerge ........ r58167 | russell
| 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines Fix a
misplaced block of code in the 1.2 version of the patch to fix
issue #8977 ........
* 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.
* /: Blocked revisions 56729 via svnmerge ........ r56729 | russell
| 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines Ensure
that lock.h is included in utils.c with AST_API_MODULE defined so
that the implementations will be properly included when the
AST_INLINE_API functions are not going to be inlined. (issue
#9124, festr) ........
2007-02-25 14:46 +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 23:25 +0000 [r56280] Joshua Colp <jcolp@digium.com>
* /: Blocked revisions 56279 via svnmerge ........ r56279 | file |
2007-02-22 18:19:25 -0500 (Thu, 22 Feb 2007) | 2 lines Always
defer Agent logoff if any channels are up until they hang up.
(issue #9123 reported by arbrandes) ........
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)
* /: Blocked revisions 55750 via svnmerge ........ r55750 | russell
| 2007-02-20 18:19:14 -0600 (Tue, 20 Feb 2007) | 9 lines Fix
random crashes when using the MeetMe application. This patch
converts list handling to use the linked list macros and most
importantly, implements reference counting on the ast_conference
objects. The reference counting was first backported from 1.4.
However, that code has some problems that caused the reference
count to never hit zero. Those problems are fixed in this patch
and will be resolved in 1.4 and trunk next, with a different
patch. (issues #7647, #9073, #9106, BE-115). ........
2007-02-21 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.
* /: Blocked revisions 55588 via svnmerge ........ r55588 | russell
| 2007-02-20 13:49:50 -0600 (Tue, 20 Feb 2007) | 3 lines Convert
a tab to spaces so that the documentation is printed out properly
aligned. ........
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) ........
* /: Blocked revisions 53134 via svnmerge ........ r53134 | russell
| 2007-02-03 14:39:45 -0600 (Sat, 03 Feb 2007) | 2 lines Revert
some changes that accidentally got committed as a part of another
fix. ........
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 23:16 +0000 [r53108] Jason Parker <jparker@digium.com>
* /: Blocked revisions 53107 via svnmerge ........ r53107 | qwell |
2007-02-01 17:14:09 -0600 (Thu, 01 Feb 2007) | 2 lines Fix a
small typo. Synopsis lines shouldn't have a newline ........
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)
* /: Blocked revisions 52954 via svnmerge ........ r52954 | russell
| 2007-01-30 13:41:52 -0600 (Tue, 30 Jan 2007) | 4 lines Don't
print a message indicating that we don't know what to do with a
proceeding control frame in ast_request_and_dial(). We just need
to ignore it. (reported by JerJer on #asterisk-dev) ........
* 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. ........
* /: Blocked revisions 52857 via svnmerge ........ r52857 | russell
| 2007-01-30 09:35:23 -0600 (Tue, 30 Jan 2007) | 5 lines Comment
out the parts in the Makefile that make codec_zap get built. It
will not yet build against zaptel 1.2, so I am disabling it to
prevent further bug reports until it gets merged. (issue #8940)
........
2007-01-30 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)
* /: Blocked revisions 52137 via svnmerge ........ r52137 | russell
| 2007-01-24 18:39:50 -0600 (Wed, 24 Jan 2007) | 3 lines Fix a
seg fault when running this application with no arguments from
AGI. (issue #8905, junky) ........
* 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 19:41 +0000 [r51411] Russell Bryant <russell@digium.com>
* /: Blocked revisions 51410 via svnmerge ........ r51410 | russell
| 2007-01-22 13:39:30 -0600 (Mon, 22 Jan 2007) | 3 lines Merge
codec_zap support for the transcoder card. This is a standalone
codec module so it will not affect anything else. ........
2007-01-22 19:28 +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) ........
* /: Blocked revisions 51359 via svnmerge ........ r51359 | file |
2007-01-22 11:23:03 -0500 (Mon, 22 Jan 2007) | 2 lines Explicitly
declare what codecs are supported by default globally since using
a bitmask for all may include ones we don't need. (issue #8357
reported by gknispel_proformatique) ........
2007-01-20 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 23:10 +0000 [r50988] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /: Blocked revisions 50987 via svnmerge ........ r50987 |
tilghman | 2007-01-15 17:09:02 -0600 (Mon, 15 Jan 2007) | 2 lines
Check return value before dereferencing (Bug 8822) ........
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; so the
source files need to be changed instead, and the generated files
regenerated.
2007-01-02 19:58 +0000 [r49212] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Small cleanup of add_t38sdp - it's always
enabled at that point in the code
2007-01-02 17:33 +0000 [r49189] Jason Parker <jparker@digium.com>
* main/pbx.c: Allow fractions of a second in the Wait()
application, like it says it allows.
2007-01-02 13:59 +0000 [r49165] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c: remove comment that is unrelated to this
function
2007-01-02 12:08 +0000 [r49145] Olle Johansson <oej@edvina.net>
* configs/features.conf.sample: Adding note on effect of
applicationmap features on re-invites
2007-01-01 23:34 +0000 [r49098-49102] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c, build_tools/menuselect-deps.in, configure,
configure.ac, codecs/codec_zap.c: check specifically for VLDTMF
and transcoding support in the system's Zaptel installation, and
make only the modules that need those features dependent on them
(this will allow building the other Zaptel-using parts of
Asterisk against older versions of Zaptel or those on other
platforms that haven't caught up yet to the Linux version)
* Makefile: use a simpler (and portable) method to ensure that
menuselect is built as a host binary
* Makefile: revert this change until a better solution can be
found... 'env -i' was not being used properly, but even when
changed to do so, this process fails during cross-compilation
because the menuselect build still sees 'CC' as set to the
cross-compiler
2007-01-01 20:14 +0000 [r49096] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: remove incomplete implementation of dnsmgr.
Let's fix this in trunk.
2006-12-30 18:31 +0000 [r49063-49073] Joshua Colp <jcolp@digium.com>
* pbx/pbx_config.c: IAX has been deprecated for quite some time so
we had better use IAX2 when creating the dial string for users.
(issue #8697 reported by ssokol)
* channels/chan_zap.c: Use asprintf to build the channel names
instead of custom function. I believe the custom function is
doing some things that are not portable across all
implementations. (issue #8570 reported by hterag & issue #8692
reported by nicolasg)
* main/rtp.c: If the Packet2Packet bridge is being broken because
of a masquerade then attempt to read a frame in so the masquerade
actually happens. Otherwise weirdness will occur. (issue #8696
reported by kjotte)
* channels/chan_iax2.c: Initialize the packet queue in load_module
instead of just declaring the list with the default value. (issue
#8695 reported by ssokol)
2006-12-30 00:40 +0000 [r49061] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have
comma args converted to vertical bars. I hope this change does
little harm.
2006-12-29 00:50 +0000 [r49042-49048] Kevin P. Fleming <kpfleming@digium.com>
* /: put this value into the correct property
* /, BUGS: Merged revisions 49045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006)
| 2 lines location of the bug posting guidelines has changed
........
* sample.call: simple commit to test CIA integration
2006-12-28 21:26 +0000 [r49032-49035] Jason Parker <jparker@digium.com>
* main/cli.c: Fix some deprecated commands. Issue 8682, patch by me
* main/http.c: saw this in passing... fix a small typo
2006-12-28 20:08 +0000 [r49028] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile: new versions of sounds
2006-12-28 19:52 +0000 [r49024] Jason Parker <jparker@digium.com>
* main/http.c: make the uris_lock a rwlock instead of a mutex lock
- needs to be forward ported to trunk
2006-12-28 19:43 +0000 [r49022] Joshua Colp <jcolp@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/lock.h: Backport support for read/write locks.
2006-12-28 19:21 +0000 [r49020] Steve Murphy <murf@digium.com>
* main/ast_expr2.fl, main/ast_expr2.c, main/frame.c,
pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c,
pbx/ael/ael_lex.c, include/asterisk/ael_structs.h,
pbx/ael/ael.tab.h, utils/ael_main.c: removed <err.h> as in trunk
from the ael stuff. Also, threw in a minor fix to frame.c to
avoid build-killing compiler warnings.
2006-12-27 22:28 +0000 [r49009] Joshua Colp <jcolp@digium.com>
* main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not
available when LOW_MEMORY is used and things are being built in
the utils directory, so we need to resort to the old method of
strncpy. (issue #8579 reported by mottano)
2006-12-27 22:06 +0000 [r48998-49006] Kevin P. Fleming <kpfleming@digium.com>
* main/enum.c, main/asterisk.c, main/rtp.c, main/term.c,
main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c,
main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c,
main/http.c, main/logger.c: since these variables all have static
duration, none of them need initializers (they default to zero
anyway)
* include/asterisk/options.h, main/asterisk.c, main/file.c: move
extern declaration for this option to a header file where it
belongs provide an initial value for 'languageprefix' option,
instead of relying on randomness to provide a useful value
2006-12-27 21:06 +0000 [r48993-48997] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Only include acl.h and lock.h once
* channels/chan_sip.c: Only set rfc2833compensate flag once
(handle_common_options)
* channels/chan_sip.c: - Remove checking for T38 options twice.
Keeping them in handle_common_options
2006-12-27 18:33 +0000 [r48987-48988] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: make the option actually match the
documentation
* channels/iax2-parser.c, include/asterisk/utils.h,
include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show
memory' and 'show memory summary' to distinguish memory
allocations that were done for caching purposes, so they don't
look like memory leaks
2006-12-27 17:59 +0000 [r48975-48985] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c, configs/sip.conf.sample: Be a bit more
politically correct
* channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy
cisco MWI support. Normally we try not to change our software for
bugs in other devices. But in this case, the Cisco phones are so
widespread so we try to implement a fix while waiting for a
bugfix from Cisco.
* channels/chan_sip.c: - Make sure handle_common_options return 1
when we found a common option - Move uncommon (only global)
option away from handle_common_options Reported by rizzo. Thanks!
* channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before
re-sending invite with auth.
* /, channels/chan_sip.c: Fix bogus content-length in t38 sdp.
(rizzo, #8600)
2006-12-26 05:20 +0000 [r48960-48966] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Get rid of a needless memory allocation and
only create a conference structure in find_conf_realtime if data
was read from realtime. (issue #8669 reported by robl)
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an
API call that initializes an RTP structure. We need this because
chan_sip is cheeky and uses a temporary RTP structure for codec
purposes, and the API calls that are used rely on the lock.
(Pointed out on asterisk-dev by Andy Wang)
* configure, configure.ac: Clean up autoconf file (gets rid of
warnings seen when rebuilding configure) and rebuild configure.
2006-12-25 05:21 +0000 [r48931-48956] Russell Bryant <russell@digium.com>
* /, funcs/func_math.c: Merged revisions 48955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) |
6 lines Fix an error introduced by copying and pasting the
handling of the >= operator for the MATH function. If a single
equal sign was used as an operator, the function would treat it
is as if it were the >= operator. Now, it properly handles it as
an invalid operator. (issue #8665, patch by tempest1) ........
* channels/chan_oss.c: Fix a typo in an error message that
indicated that the MGCP channel type could not be registered,
instead of the correct type, OSS.
* /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) |
3 lines Check for the proper return value on an error in a call
to mmap(). This was reported by Andy Wang on the asterisk-dev
list. Thanks! ........
* /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) |
3 lines Remove a couple of misplaced dots in log messages. This
was reported by Andrea Spadaccini on the asterisk-dev mailing
list. ........
* main/http.c: Implement locking for the list of URI handlers to
make it thread-safe.
2006-12-23 Kevin P. Fleming <kpfleming@digium.com>
* Asterisk 1.4.0 released.
2006-12-22 22:33 +0000 [r48870-48906] Jason Parker <jparker@digium.com>
* Makefile, main/stdtime/localtime.c: Minor fixes for Solaris.
* channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia
2006-12-21 20:26 +0000 [r48783] Joshua Colp <jcolp@digium.com>
* /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2
lines Add new silence sound files to the spec for Redhat. (issue
#8652 reported by alvaro_palma_aste) ........
2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: vms doesn't exist on non-IMAP storage
builds.
* apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so
it is then passed to the IMAP store file function. (issue #8614
reported by punknow)
* doc/snmp.txt: find is not the same as bind when it comes to
documentation. (issue #8626 reported by johann8384)
2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming <kpfleming@digium.com>
* channels/Makefile: suppress compiler warnings in this module
until it can be improved
2006-12-19 21:12 +0000 [r48585] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 48584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2
lines Free localuser structure when we fail to dial (issue #8612
reported by rizzo) ........
2006-12-19 21:03 +0000 [r48583] Luigi Rizzo <rizzo@icir.org>
* apps/app_sms.c: fix a bogus datalen in the frames generated by
app_sms (causing noisy output if you listen to the output!) This
affects trunk as well, whereas 1.2 is ok.
2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming <kpfleming@digium.com>
* res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable
type for these unixODBC API calls, eliminating warnings on 64-bit
platforms that use the 'new' 64-bit types for ODBC API calls
2006-12-19 03:46 +0000 [r48571] Joshua Colp <jcolp@digium.com>
* Makefile: Use env -i to start a fresh environment when going to
build menuselect. This is more portable then using unset. (issue
#8543 reported by jtodd)
2006-12-18 17:23 +0000 [r48566] Luigi Rizzo <rizzo@icir.org>
* include/asterisk/channel.h: unbreak the macro used for
incrementing the frame counters. I don't know when the bug was
introduced, but with the typical usage c->fin =
FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects
trunk as well (fix coming).
2006-12-18 17:15 +0000 [r48564] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Put thread into proper list if we abort
handling due to an error, and also hold the lock while putting it
back into the proper idle list so we don't prematurely get a
signal. (issue #8604 reported by arkadia)
2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming <kpfleming@digium.com>
* codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile,
utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile,
utils/ael_main.c: remove some now-unnecessary explicit includes
of autoconfig.h clean up per-file dependencies during 'make
clean'
* build_tools/prep_tarball: need an additional argument here to
make the downloads actually occur
* configure, include/asterisk/autoconfig.h.in, configure.ac,
acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep
these calls from thinking they have multiple arguments
* codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile,
funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast,
main, codecs/gsm, pbx, res, channels, codecs, utils, agi,
main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr:
simplify dependency tracking system, using the compiler's
built-in method for generating them, and only doing dependency
tracking if developer mode is enabled via the configure script
* Makefile, include/asterisk.h, main/stdtime/localtime.c: since we
really, really have to have autoconfig.h included before all
other headers (especially system headers), the Makefile will now
force it to happen (this will fix build problems with files like
ast_expr2f.c, where we can't control the inclusion order in the
file itself)
* funcs/func_curl.c: instead of initializing the curl library every
time the CURL() function is invoked, do it only once per thread
(this allows multiple calls to CURL() in the dialplan for a
channel to run much more quickly, and also to re-use connections
to the server) (thanks to JerJer for frequently complaining about
this performance problem)
2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Turn payload_lock into bridge_lock and make it
encompass all RTP structure contents that may relate to bridge
information, including who we are bridged to.
* channels/chan_iax2.c: Hold call structure lock in places where a
qualify or peer action can destroy it.
* channels/chan_iax2.c: Lock network retransmission queue in all
places that it is used.
2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported
from 1.2)
* channels/chan_sip.c: Update to latest IANA spec
2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Use a wakeup variable so that we don't wait
on IO indefinitely if packets need to be retransmitted.
* main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP
structure can change AFTER a bridge has started. This comes from
the packet handling of the SIP response when indication that it
was answered has been sent. Therefore we need to protect this
data with a lock when we read/write. (issue #8232 reported by
tgrman)
* main/rtp.c: Remove direct RTCP bridging. I've come to the
conclusion that we should handle this through the core and not
just forward it on. Should solve a few bugs.
2006-12-12 Kevin P. Fleming <kpfleming@digium.com>
* Asterisk 1.4.0-beta4 released.
2006-12-12 04:13 +0000 [r48401] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This
is the way it should have been done.
2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman <mogorman@digium.com>
* sounds/Makefile: new sounds package with 100% more silence
* /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge
from https://svn.digium.com/svn/asterisk/branches/1.2 ........
r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
| 4 lines app_externalivr needs a real silence file, and
additional changes to add silence files into core instead of
extra patch provided by bug 8177 with minor additions. ........
2006-12-11 21:31 +0000 [r48391] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Return non-existant callerid handling to
that which it was before. In 1.4 and trunk callerid can be
allocated but not have any contents so we have to use
ast_strlen_zero before passing it to the relevant functions.
(issue #8567 reported by pabelanger)
2006-12-11 05:37 +0000 [r48382] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* funcs/func_strings.c: STRFTIME() does not actually require an
argument (issue 8540)
2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Merge in my latest RTP changes. Break out RTP and
RTCP callback functions so they no longer share a common one.
* apps/app_meetme.c: Use the correct API call to say a device state
changed. (Yes, I'm a nub.)
* apps/app_meetme.c: Don't access the conference structure after it
has been freed.
2006-12-11 00:47 +0000 [r48375] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006)
| 5 lines When doing a fork() and exec(), two problems existed
(Issue 8086): 1) Ignored signals stayed ignored after the exec().
2) Signals could possibly fire between the fork() and exec(),
causing Asterisk signal handlers within the child to execute,
which caused nasty race conditions. ........
2006-12-10 03:04 +0000 [r48372] Steve Murphy <murf@digium.com>
* channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
line This version applies the patch suggested by stevens in bug
7836 (make inbound channel RINGING state consistent with other
channels). ........
2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Use locking when accessing the
registrations list. This list is not actually used very often, so
the likelihood of there being a problem is pretty small, but
still possible. For example, if the CLI command to list the
registrations was called at the same time that a reload was
occurring and the registrations list was getting destroyed and
rebuilt, a crash could occur. In passing, go ahead and convert
this list to use the linked list macros.
* /: Blocked revisions 48361 via svnmerge ........ r48361 | russell
| 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use
locking when accessing the registrations list. This list is not
actually used very often, so the likelihood of there being a
problem is pretty small, but still possible. For example, if the
CLI command to list the registrations was called at the same time
that a reload was occurring and the registrations list was
getting destroyed and rebuilt, a crash could occur. ........
2006-12-07 18:17 +0000 [r48357] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
Dec 2006) | 3 lines Ensure that the file position is not
incremented beyond the total number of files available for
playback. (issue #8539, ulogic) ........
2006-12-07 15:33 +0000 [r48349] Steve Murphy <murf@digium.com>
* main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that
killed bug 8423 -- OriginateSuccess and OriginateError incomplete
channel name. May it rest in peace.
2006-12-06 16:25 +0000 [r48326] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being
retransmitted to Asterisk
2006-12-06 16:15 +0000 [r48323] Russell Bryant <russell@digium.com>
* configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
in the sample configuration file. (issue #8526, arkadia) ........
2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Don't send Contact on MESSAGE
2006-12-05 20:42 +0000 [r48279] Jason Parker <jparker@digium.com>
* configure.ac: Fix curl version number testing to be much more
friendly to non-bash shells. Issue 8508, patch by me. This
*SHOULD* be POSIX compliant now..
2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Merging the invitestate-1.4 branch after
successful testing. Will check if I can solve this with less
changes in 1.2.
* configs/sip.conf.sample: Add missing s from another repository.
(thanks jcmoore!)
* configs/sip.conf.sample: Updating sip.conf.sample with
information about T38 not working when chan_local or chan_agent
is involved in the call. I don't know how big a fix that would be
to solve, but this is the current state of affairs. (Chan_sip
currently checks if the other side of the bridge has a SIP tech.
We could/should implement another check, possibly for udptl_write
or some flag in the ast_channel structure).
2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Oops, forgot to release the odbc handle
* apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006)
| 6 lines If the recording in the database is too large, it will
fail to retrieve with an mmap error. Not too sure why this
doesn't happen when we put it in the database, also, but since
that doesn't seem to be broken, I'm not going to fix it (at least
until someone reports it). Solution is to ask for the file in
smaller chunks. (Bug 8385) ........
2006-12-04 21:48 +0000 [r48237-48248] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c: Fix an issue which didn't allow
unavail/greet/busy/etc messages from being saved into ODBC (and
probably IMAP).
* /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell |
2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert
change from 8016 - this breaks other stuff... Needs further
review. Tip: When you've reported a bug about something and
somebody has put up a patch for it.. It's not a good idea to open
a completely new bug and say that something is broken because of
the patch in the other bug - PLEASE mention something in the bug
where the patch was actually created. ........
* /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell |
2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an
issue where a message isn't saved correctly when using ODBC
storage and reviewing a message. Issue 8016 - patch by sokhapkin.
........
2006-12-04 18:16 +0000 [r48234] Joshua Colp <jcolp@digium.com>
* /: Blocked revisions 48233 via svnmerge ........ r48233 | file |
2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the
generic bridge tells us not to retry, and we have a frame to spit
out then break the bridge. Props to markit in #asterisk-bugs for
bringing this up. ........
2006-12-04 17:54 +0000 [r48228-48230] Jason Parker <jparker@digium.com>
* configs/voicemail.conf.sample: Add documentation to
voicemail.conf.sample for ODBC storage. Issue 8499 - patch by
blitzrage.
* doc/snmp.txt: Attempt to document some of the dependencies that
are needed for net-snmp Issue 8499 - initial patch by blitzrage.
2006-12-03 06:34 +0000 [r48223] Russell Bryant <russell@digium.com>
* sounds/Makefile: When "fetch" is in use, instead of "wget",
--continue is not a valid option. (issue #8451)
2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: - Removing one of two pieces of code to
handle 481 response on INVITE - Move handling of REFER response
to handle_response_refer()
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax
transmission happens - Encapsulate RTP timers in the rtp
structure so we have one for video and one for audio The video
one is not used in 1.4, really. Will be used for RTP keepalives
when we can send something that video phones support in the RTP
stream. I now this is a big architectual change at this stage for
1.4, but decided it was needed to avoid future bug reports. -
Document the RTP NAT keepalive option in sip.conf.sample Issue
7679 in the bug tracker. Please test.
2006-12-02 03:50 +0000 [r48195] Russell Bryant <russell@digium.com>
* include/asterisk/utils.h: Backport the comment containing the
warning regarding the limitations on the usage of this function.
It is thread safe, but not technically reentrant.
2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_dial.c, /: Merged revisions 48192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006)
| 2 lines if Dial() is going to send music-on-hold to the calling
party, it has to send PROGRESS first to ensure that the reverse
audio path has been setup first (BE-106) ........
2006-12-01 23:16 +0000 [r48190] Russell Bryant <russell@digium.com>
* Makefile, configure, configure.ac, makeopts.in, sounds/Makefile:
FreeBSD 6.1 does not include wget by default. However, it has
fetch which will work just fine for our purposes of downloading
the sounds packages. So, check for both wget and fetch and the
configure script and use what was found to download them. If
neither one was found, and sound packages are selected that must
be downloaded, the install process will print out an informative
error message indicating the situation. Also, fix a couple places
where "make" was hard coded into some output messages by
replacing them with the $(MAKE) variable. (issue #8451, initial
patch by pabelanger, with additional modifications by me)
2006-12-01 20:25 +0000 [r48184-48186] Jason Parker <jparker@digium.com>
* configs/extensions.conf.sample, /: Merged revisions 48183 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
lines Fix a small typo - issue 8848, reported by pabelanger
........
2006-12-01 19:38 +0000 [r48179] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/cli.c: Double-unlock error (reported by blitzrage on IRC)
2006-12-01 17:41 +0000 [r48177] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c, configs/sip.conf.sample: - Backport of the
"limitonpeers" patch from trunk, to fix a lot of issues with
queues and SIP device states - Remove support for T.38 early
media, since it's impossible. (Two patches in one - extra friday
evening offer due to being off line from svn today... :-)
2006-11-30 21:18 +0000 [r48168] Joshua Colp <jcolp@digium.com>
* main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not
do a partial bridge for Google Talk since we need to handle STUN.
(issue #8448 reported by phsultan)
2006-11-30 20:51 +0000 [r48166] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Issue 8319 - change noncecount before
using it.
2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp <jcolp@digium.com>
* /: Blocked revisions 48161 via svnmerge ........ r48161 | file |
2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't
write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel
driver. (issue #8390 reported by hselasky) ........
* /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
lines Only print out debug message if bridged channel is not
NULL. (issue #8412 reported by jubilex) ........
* /, res/res_features.c: Merged revisions 48154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
lines Do not listen for DTMF on the bridge that comes into
existence when ParkedCall is executed. This means native bridging
can now occur for this. (issue #8406 reported by kebl0155)
........
* main/cdr.c, /: Merged revisions 48151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
lines Print certain CDR messages out at the NOTICE level versus
WARNING since they can occur when used with the CDR applications
and are perfectly fine. (issue #8367 reported by dartvader)
........
* /: Blocked revisions 48146 via svnmerge ........ r48146 | file |
2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember
the pointer to the allocated block of memory so that we can free
it and not cause a memory leak. (issue #8449 reported by arkadia)
........
* /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov
2006) | 2 lines Document 'port' for SIP peers, came up because of
the current mailing list thread. (issue #8450 reported by
blitzrage) ........
2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson <oej@edvina.net>
* doc/manager.txt: Explain status reports and make codefreeze more
happy :-)
* /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by
GS 487 adapter without CSEQ on separate line in the REGISTER
request. Imported from 1.2.
2006-11-29 21:05 +0000 [r48115] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in
mm_login. (issue #8420 reported by slimey)
2006-11-29 19:56 +0000 [r48113] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Explain the use device status system
implemented in SIP for subscriptions, queues and manager a bit
better. Like in 1.2, you will get more detailed information if
you set a call limit for a device. When the call limit is
reached, the status system will report a device as busy. For
queues, setting a call limit per SIP device is propably a
requirement. In most cases, it will work much better if you only
use type=peer and not type=friend. We might decide to backport
the new setting from trunk to apply all call limits to the peer
part of a friend only.
2006-11-29 16:50 +0000 [r48107] Joshua Colp <jcolp@digium.com>
* main/rtp.c, /: Merged revisions 48106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
lines If the frame was duplicated before writing out then we need
to free it. (issue #8429 reported by edguy3) ........
2006-11-29 08:03 +0000 [r48105] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma.
2006-11-29 04:26 +0000 [r48101] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Don't crash if the mailstream was not
created.
2006-11-28 18:26 +0000 [r48095] Jason Parker <jparker@digium.com>
* Makefile: Export several more variables in top level Makefile.
Inspired by issue 8438.
2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp <jcolp@digium.com>
* channels/chan_phone.c, /: Merged revisions 48087 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov
2006) | 2 lines According to the research I have done we never
needed to include compiler.h in the first place so let's not!
(issue #8430 reported by edguy3) ........
* apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2
lines Use the proper function to get the new message count
instead of always using the filesystem. (issue #8421 reported by
slimey) ........
2006-11-27 17:20 +0000 [r48049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27
Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381)
........
2006-11-27 17:17 +0000 [r48046] Russell Bryant <russell@digium.com>
* main/manager.c: Remove a couple of unused variables (issue #8380,
casper)
2006-11-27 15:32 +0000 [r48038] Joshua Colp <jcolp@digium.com>
* pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2
lines Do not reference the freed outgoing structure in the debug
message. (issue #8425 reported by arkadia) ........
2006-11-27 06:41 +0000 [r48031] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Change logging message
2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy <murf@digium.com>
* funcs/func_cdr.c: might as well also document the raw values of
the flag vars
* /, funcs/func_cdr.c: A little bit of func_cdr documentation
upgrade-- no bug# involved, although 8221 may have inspired it.
2006-11-25 09:28 +0000 [r48002] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4
and future releases, you can disable subscription support totally
or per peer in sip.conf with allowsubscribe = yes | no
2006-11-24 17:17 +0000 [r47992] Steve Murphy <murf@digium.com>
* main/translate.c: bug 8189 posted this fix for main/translate.c
for PLC
2006-11-24 15:46 +0000 [r47989] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23
Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE.
beatufied some logs, changed some loglevels. changed the default
value of block_on_alarm ........
2006-11-23 11:01 +0000 [r47959] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Don't allocate unused variable.
2006-11-22 21:47 +0000 [r47944] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Video will never reach Packet2Packet bridging and can
do more harm then good.
2006-11-21 17:32 +0000 [r47897] Joshua Colp <jcolp@digium.com>
* main/rtp.c: If we have the non standard G726-32 setting turned on
we want to return G726-32 to the SDP, not our AAL2 string. (issue
#8330 reported by voipgate)
2006-11-21 15:20 +0000 [r47892] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Apparently Exosip sends a 101 after a 100
provisional response. Let's not treat that as early media.
(discovered at the AVTF meeting in Paris).
2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Oops, merge missed release of odbc object
* apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006)
| 2 lines Failing to trap -1 error from mmap causes segfault
(Issue 8385) ........
2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp <jcolp@digium.com>
* main/frame.c, /: Merged revisions 47859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2
lines Don't forget to byte swap if we are exiting the smoother
feed early. (issue #8287 reported by arturs) ........
* /: Blocked revisions 47855 via svnmerge ........ r47855 | file |
2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines Free
history items at the end of use of the temporary SIP pvt
structure. (issue #8383 reported by benh) ........
* main/rtp.c: Only remove/destroy the RTCP I/O item if it exists.
* .cleancount, apps/app_dial.c, apps/app_directed_pickup.c,
include/asterisk/channel.h: Use a separate variable in the
channel structure to store the context that the channel was
dialed from. (issue #8382 reported by jiddings)
2006-11-20 11:45 +0000 [r47843-47845] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Explain properly how videosupport works.
Committ from Asterisk Video Task Force meeting in Paris!
* /, channels/chan_sip.c: Make sure we destroy scheduled items and
not use them ever again after destruction (rizzo)
2006-11-18 17:59 +0000 [r47823] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: fix bug 7450 - Parsing fails if From header
contains angle brackets (the bug was only in a corner case where
the < was right after the opening quote, and the fix is trivial).
2006-11-16 23:19 +0000 [r47781-47782] Jason Parker <jparker@digium.com>
* apps/app_db.c, apps/app_dial.c: Fix a couple of typos. Initially
pointed out by mrobinson.
* /: Blocked revisions 47780 via svnmerge ........ r47780 | qwell |
2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a
couple of typos in applications.. Initially spotted by mrobinson.
........
2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming <kpfleming@digium.com>
* /, doc/billing.txt: update documentation regarding IAX2 transfers
and CDRs Merged revisions 47776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006)
| 2 lines update clearly wrong documentation regarding cdr_custom
........
2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Compare technology using the pointers
instead of a straight comparison based on name. (issue #8228
reported by dean bath)
* /: Blocked revisions 47761 via svnmerge ........ r47761 | file |
2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines Look for
the header file specifically in all cases, not just the existence
of the directory. (issue #8358 reported by mrness) ........
2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming <kpfleming@digium.com>
* configure, configure.ac: check for pre-1.4 versions of Zaptel and
abort the configure script if found with an appropriate error
message
2006-11-16 19:24 +0000 [r47755] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD
notification optional, in order to avoid a lot of extra database
lookups for all those realtime users out there.
2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c, /: Merged revisions 47750 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov
2006) | 2 lines Because of the way chan_local is written we
should be extra careful and make sure our callback functions have
a tech_pvt. (issue #8275 reported by mflorell) ........
* apps/app_meetme.c: Don't unreference the SLA object if there is
no SLA object in the devicestate callback. (issue #8354 reported
by loloski)
2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Don't fixup if there's nothing to fixup
* UPGRADE.txt: Warn users about change in canreinvite
* channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never
authenticated (according to the RFC) - Update docs on
canreinvite. "nonat" is the recommended setting for most users
with phones behind a NAT.
2006-11-15 22:31 +0000 [r47712] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c, /: Merged revisions 47711 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov
2006) | 2 lines Make sure that the pvt structure exists before
trying to do fixup on Local channels. (issue #7937 reported by
mada123, fix by alamantia with mods by me) ........
2006-11-15 21:56 +0000 [r47709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL
2006-11-15 21:33 +0000 [r47707] Joshua Colp <jcolp@digium.com>
* main/channel.c: We need to ensure timelimit stuff is included as
well so warnings get played. (issue #8050 reported by KNK)
2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming <kpfleming@digium.com>
* main/file.c: don't try to call fclose() if fopen() failed
2006-11-15 20:31 +0000 [r47698] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: - Improve SIP history - Never send reply to
ACK (again...)
2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006)
| 4 lines ensure that message duration is included in email
notifications for forwarded messages (BE-96, fix by me after
corydon used his clue-bat on me) ensure that duration in the
message metadata is updated if prepending is done during
forwarding (related to BE-96) remove prototype for API call that
does not exist ........
* main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15
Nov 2006) | 2 lines clear the category's variable tail pointer as
well when variables are detached from it ........ r47688 |
kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2
lines when appending a list of variable to a category, ensure the
tail pointer points to the last variable in the list ........
r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006)
| 2 lines when re-writing the config file, don't repeat the path
if it hasn't changed ........
* main/config.c, /: Merged revisions 47682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006)
| 2 lines ouch... don't use printf, use ast_log/ast_verbose
........
2006-11-15 17:46 +0000 [r47672] Luigi Rizzo <rizzo@icir.org>
* main/cli.c: fix longest match search in find_cli. Trunk already
fixed. 1.2 not affected (well, i have no idea, the code is
totally different there).
2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Send error message when we can't allocate
SIP dialog, possibly due to limitation of file descriptors.
(imported from 1.2)
2006-11-15 04:45 +0000 [r47645] Joshua Colp <jcolp@digium.com>
* main/rtp.c: If NAT detection is turned on or already detected
then say NAT is active when setting the remote RTP peer when
doing early bridging. (issue #8365 reported by marcelbarbulescu)
2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming <kpfleming@digium.com>
* main/term.c: more formatting cleanup, and avoid running off the
end of the string
2006-11-15 00:14 +0000 [r47639] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Turn notice about unknown RTCP packet type into a
debug message instead.
2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming <kpfleming@digium.com>
* channels/misdn/isdn_lib.c: silence compiler warning on 64-bit
platforms (this variable is an 'int' anyway, comparing it to
'signed long' is not useful)
2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2
lines Update copyright information in the ADSI logo blob.
........
* channels/chan_sip.c: Only keep the video RTP structure around if
1. Video support is enabled and 2. A video codec is enabled on
the dialog
* funcs/func_uri.c: Small documentation clarification for
URIENCODE. (issue #8294 reported by salaud)
2006-11-14 18:54 +0000 [r47621] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Conversion of res_odbc API to include ast_
prefix did not completely transition app_voicemail when
ODBC_STORAGE is used (reported on IRC by caio1982, not in
bugtracker)
2006-11-14 16:45 +0000 [r47617] Joshua Colp <jcolp@digium.com>
* apps/app_amd.c: Use LOG_DEBUG to print out the indication that
app_amd is using default settings instead of using LOG_NOTICE.
This stops needless logging of this information under normal
circumstances. (issue #8361 reported by Seb7)
2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Update documentation to fit the
implementation...
* /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in
retransmission system if it's an OPTION packet from peerpoke
2006-11-13 21:28 +0000 [r47584] Joshua Colp <jcolp@digium.com>
* /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2
lines Initialize global pointers for connection and result to
NULL. (issue #8356 reported by james) ........
2006-11-13 20:20 +0000 [r47581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006)
| 2 lines Having more than 255 old messages caused corruption in
the new/old count ........
2006-11-13 19:15 +0000 [r47576] Steve Murphy <murf@digium.com>
* main/config.c: This solves bug 8342, whereby a crash occurs under
certain circumstances while reading a config file with comments--
a call to CB_ADD shouldn't happen if withcomments is zero
2006-11-13 19:11 +0000 [r47573] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/cli.c, channels/chan_sip.c: Re-enable old deprecated
commands
2006-11-13 19:10 +0000 [r47572] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: - Don't reply to INVITE already replied
to when we get BYE - Declare errmsg as int. Oops.
2006-11-13 18:18 +0000 [r47564] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing
the messed if, but we all forgot to update the regressions. Until
now.
2006-11-13 17:13 +0000 [r47553] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: AEL need not complain about parkedcalls not being
found... just confuses users
2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp <jcolp@digium.com>
* /, apps/app_sms.c: Merged revisions 47549 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2
lines When sending an SMS with a user data header properly set
the UDH flag in the first byte. (issue #8347 reported by
hoffmeis) ........
* main/cli.c: Free full command string upon unregistering of CLI
command. Backported from revision 47536 from rizzo.
2006-11-13 16:00 +0000 [r47540] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Only produce error message about sip history
once
2006-11-13 05:48 +0000 [r47527] Russell Bryant <russell@digium.com>
* configure, acinclude.m4: AC_PROG_SED is included in autoconf
2.60, but apparently it is not included in 2.59. So, to maintain
compatability with 2.59 since it is a small change, copy this
macro into acinclude.m4 and rename it to AST_PROG_SED. (issue
#8345)
2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_odbc.c, /: Merged revisions 47525 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006)
| 2 lines If the execute fails a second time, make sure that we
don't pass back a stale handle ........
* channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006)
| 2 lines Don't play dialtone if the seizing the channel fails
(Bug 7754) ........
2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks
DEA!!!)
* channels/chan_sip.c: Part of issue 8078 - parse even if udptl is
UDPTL in sdp...
* channels/chan_sip.c: - Don't destroy SIP dialog because of a
failed T.38 re-invite. Wait for a bye. Final response to a
re-invite does not mean that the session dies, only that the
re-invite fails. - Keep RTP active during processing of T.38
re-invite. If the re-invite fails, RTP needs to remain as before
the re-invite. Issue 8338 - darren1713. Please test.
* channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp
-Add some comments to t.38 code
2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) |
4 lines Only do the check to determine whether the channel
calling this function is an IAX2 channel when getting the IP
address using the special argument, CURRENTCHANNEL. (issue #8341,
jcovert) ........
* Makefile: Add the target "menuconfig" as an alias for the
"menuselect" target. This is just a favor to users so that if you
accidentally type "make menuconfig" instead of "make menuselect",
it still works. (inspired by a comment on IRC from wangster
calling me an "especially devious asterisk developer" for having
it be menuselect instead of menuconfig. :) )
* main/term.c: Tweak the formatting of this new function to better
conform to coding guidelines.
2006-11-11 02:04 +0000 [r47490] Matt O'Gorman <mogorman@digium.com>
* main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo
safe output!
2006-11-10 22:23 +0000 [r47480] Matt Frederickson <creslin@digium.com>
* channels/chan_zap.c: Make sure we don't use 32 bits when we only
need one bit.
2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: ...and make sure that the dialog is
destroyed, even if we don't get any answer on the bye... This is
the channel that remains dead after the SIP transfer
* channels/chan_sip.c: Add debug output while trying to trace bug
in bug report
* channels/chan_sip.c: Make sure we destroy dialog...
* /, channels/chan_sip.c: Small cleanup of handle_request_invite()
- imported from 1.2 with changes
2006-11-10 19:47 +0000 [r47462] Matt Frederickson <creslin@digium.com>
* channels/chan_zap.c: Fix for #7321. Be able to explicitly hide
callerid name for switches that bork on it.
2006-11-10 18:56 +0000 [r47454] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Issue 8010 - Fix support for multipart
SDP (alphaque)
2006-11-10 17:13 +0000 [r47444] Luigi Rizzo <rizzo@icir.org>
* build_tools/prep_moduledeps: grep -m is not available on BSD, so
use head -1 instead
2006-11-10 16:53 +0000 [r47437] Joshua Colp <jcolp@digium.com>
* apps/app_chanspy.c: Only split up extension and context if a
value exists. (issue #8332 reported by loloski)
2006-11-10 16:51 +0000 [r47436] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c,
channels/chan_skinny.c, channels/chan_h323.c,
channels/chan_iax2.c: Discussion of these CLI changes resulted in
more consistency (Bug 8236)
2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_queue.c: if adding a queue member is LOG_NOTICE, then
removing them should be LOG_NOTICE, not LOG_DEBUG
* apps/app_queue.c: reflect addition/removal of dynamic queue
members in queue_log, so that people using dialplan replacement
for AgentCallbackLogin can still track login/logout (issue #7736,
reported/patched by whoiswes but this commit was written by me
and covers all three paths for AQM/RQM)
2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Rip out half implementation of 491 response
support, since it wasn't implemented properly and caused memory
leaks in the case of us getting 491's, which Asterisk actually
sends... Since it is a bit too complicated to fix this, I'll rip
it out of 1.4 and put it on the to-do-list for future releases.
Now, we handle this as congestion, which it really is. Issue
#8331
* channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD.
Thanks fenlander!
2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp <jcolp@digium.com>
* channels/chan_h323.c: Fix building of chan_h323 by completeing
some structure definitions. (issue #8327 reported by Mithraen)
* apps/app_voicemail.c: Do conversion in a more easier to read and
working way for \r, \n, and \t. (issue #8324 reported by
johnlange)
2006-11-09 21:26 +0000 [r47391] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c, channels/chan_zap.c,
build_tools/prep_moduledeps: Work around an issue that caused
menuselect to display a bogus description for app_voicemail and
chan_zap. These modules use some preprocessor directives to
determine what it will report to Asterisk as its description.
However, the way we extract this information from the source
files for menuselect is not smart enough to figure this out.
(issue #8326, #8328)
2006-11-09 16:53 +0000 [r47380] Joshua Colp <jcolp@digium.com>
* channels/chan_phone.c, /: Merged revisions 47379 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov
2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and
higher as, well, it's apparently going to be removed. This should
make all you FC6 fans happy as your Asterisk will now build
without any mods. ........
2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant <russell@digium.com>
* main/cli.c: fix tab completion for "core debug channel" and "core
no debug channel"
* main/cli.c: Fix "core show channel". Also, fix tab completion for
both "core show channel" and "core show channels".
* main/cli.c: Fix "core debug channel <whatever>". I guess someone
needs to go through and audit every CLI command that changed
number of arguments ...
* main/asterisk.c: revert the previous change, which actually
modified the deprecated command, "show profile". Now, actually
apply the change to "core show profile".
* main/asterisk.c: Fix argument parsing for the "core show profile"
CLI command (fixed by rizzo in his branch, team/rizzo/astobj2)
* main/cli.c: Fix another CLI command, "core show uptime" ...
(issue #8323, reported by johnlange, fixed by myself)
* main/asterisk.c: fix "core show version" to reflect the new
number of arguments for this CLI command (issue #8316, kshumard)
2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy <murf@digium.com>
* main/channel.c: This update fixes 7531
* channels/chan_skinny.c: Committed in behalf of 8190.
2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming <kpfleming@digium.com>
* main/frame.c: the battle over CLI command formats has broken
stuff...
* channels/chan_sip.c: add simple fix for SDP to report proper
sample rate for G.722 media sessions
2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant <russell@digium.com>
* utils/streamplayer.c: I occasionally get email from users that
are trying to figure out what this does, or due to some
misunderstanding as to what it is supposed to do, can't get it to
work. So, I have added some text here to hopefully explain what
this application does and does not do.
* channels/chan_gtalk.c: Make this module build again
* configure, configure.ac, acinclude.m4: Copy the macros from
libtool.m4 to our own acinclude.m4 such that libtool is no longer
required to be installed to be able to generated the configure
script.
2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo)
2006-11-07 23:46 +0000 [r47303] Steve Murphy <murf@digium.com>
* channels/chan_oss.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_nbs.c, channels/chan_mgcp.c,
include/asterisk/stringfields.h, 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, main/utils.c, include/asterisk/channel.h,
channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to
solve the problem in bug 7506. It's a lot of rework to solve a
fairly small problem... such is life.
2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c: Make MOH work as it did before in
chan_local, without this then it can go funky when transfers and
MOH are involved. (issue #7671 reported by jmls)
2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming <kpfleming@digium.com>
* configs/musiconhold.conf.sample: clean up sample config, and make
native file playback the more obvious default choice
2006-11-07 18:38 +0000 [r47275] Matt O'Gorman <mogorman@digium.com>
* apps/app_voicemail.c: large overhaul to voicemail imap support.
Allows support for more imap servers, also a better
implementation of several parts of the original work. patch
provided by 8033 with major upgrades.
2006-11-07 17:30 +0000 [r47268] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of
continue.
2006-11-07 13:13 +0000 [r47250] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Fixing the attack shield so it doesn't
produce attacks... Issue 8265 - never reply to an ACK
2006-11-07 01:25 +0000 [r47239] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06
Nov 2006) | 5 lines If random order is enabled for files mode
music on hold, set a random initial position, instead of always
starting at the first file, and doing the random operation only
when switching to the next file. (bug reported by John Lange on
the asterisk-dev mailing list) ........
2006-11-04 18:32 +0000 [r47199] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and
transfer from "john" Thank you!
2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant <russell@digium.com>
* main/cli.c: Fix another bug in "core set debug" ...
* main/asterisk.c, main/cli.c: Really fix the "core set debug" and
"core set verbose" CLI commands.
* main/cli.c: fix the "atleast" option to the "core set verbose"
and "core set debug" CLI commands
2006-11-03 23:17 +0000 [r47176] Steve Murphy <murf@digium.com>
* channels/chan_sip.c: This fix introduced via bug 8233
2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo <rizzo@icir.org>
* bootstrap.sh: align bootstrap.sh with the version in trunk (needs
to be blocked as it is already in trunk)
* configure.ac: add proper environment vars to detect modules on
freebsd. (already applied to trunk so it needs to be blocked
there)
2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c,
channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More
changes making the CLI more consistent with "category verb
arguments" (continuation of issue 8236)
* main/config.c, main/cli.c, main/channel.c, main/manager.c,
channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c,
main/http.c, main/file.c, main/logger.c, main/image.c,
res/res_indications.c, main/asterisk.c, res/res_odbc.c,
channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
channels/chan_local.c, main/frame.c, channels/chan_sip.c,
res/res_features.c, channels/chan_agent.c, res/res_crypto.c,
res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c:
Reverse change of "show" to "list" and make several other
commands more consistent with "category verb arguments"
2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Move check for codec translation to
sip_call() instead of in add_sdp. No one bothers with the result
of add_sdp anyway... Yet...
* channels/chan_sip.c: Disable code for T38 over TCP and RTP since
there's no trace of actual functionality for it :-)
2006-11-02 17:49 +0000 [r46965] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02
Nov 2006) | 3 lines ignore files in a music on hold directory
that begin with '.' (issue #8249, cboie) ........
2006-11-02 17:17 +0000 [r46963] Nadi Sarrar <ns@beronet.com>
* channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix
2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: don't send INVITE when we have determined
that we can't offer any audio formats due to lack of transcoding
support (or incorrect configuration)
2006-11-02 16:06 +0000 [r46930] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2
lines Repeat after me oej: I will at least make sure my code
compiles before I commit it. ........
2006-11-02 15:24 +0000 [r46901] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2)
2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant <russell@digium.com>
* /, main/callerid.c: Add the missing call to free described in
issue #8268. Also, add a bunch of missing calls to free in
callerid_feed_jp().
* main/say.c: fix saying one hundred and two hundred in hebrew
(issue #7810, eldadran)
* Makefile, configure, codecs/gsm/Makefile, configure.ac,
build_tools/strip_nonapi, makeopts.in: Fixes for
cross-compilation on mips (issue #8058, ywalther, with some
modifications)
* aclocal.m4, build_tools/menuselect-deps.in, configure,
build_tools/embed_modules.xml, configure.ac: Add a check in the
configure script to determine whether ld is GNU ld or not. This
is needed because module embedding only works for gnu ld. GNU ld
is now listed as a dependency for all of the module embedding
options in menuselect. (issue #8143)
2006-11-01 20:35 +0000 [r46822] Matt O'Gorman <mogorman@digium.com>
* channels/chan_gtalk.c: bind address support from bug 8164
2006-11-01 19:49 +0000 [r46802] Steve Murphy <murf@digium.com>
* res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
accept longer strings or mass confusion and a lot of lost time is
the result
2006-11-01 18:39 +0000 [r46780] Joshua Colp <jcolp@digium.com>
* main/Makefile: Force poll() emulation for Darwin to always be on.
It's too broken to consider being used. This resolves the console
issue OSX users have been seeing. I would have liked to autoconf
this but I haven't been able to come up with a test case that
works. Que sera.
2006-11-01 18:26 +0000 [r46778] Russell Bryant <russell@digium.com>
* res/res_monitor.c, /: Merged revisions 46776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) |
9 lines soxmix and Asterisk expect different file extensions for
certain formats. This was already handled for the wav49 format.
However, it was not handled for ulaw and alaw. I fixed this in
such a way that using the alternate extensions for ulaw and alaw
will only happen if we know we're calling soxmix, and not a
custom script defined using the MONITOR_EXEC variable. The wav49
processing was left alone so that external scripts will see no
behavior change. (issue #7550, reported by mnicholson, proposed
patch by junky, committed fix is a bit different) ........
2006-11-01 18:21 +0000 [r46775] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: It's another round of chan_iax2 fixes!
Should hopefully fix the deadlock issues people have been
reporting. IAXtel now has qualify turned on for 800 peers and it
is handling it fine.
2006-11-01 17:48 +0000 [r46760] Steve Murphy <murf@digium.com>
* main/config.c: Cleanups suggested by Russell.
2006-11-01 16:39 +0000 [r46744] Russell Bryant <russell@digium.com>
* channels/chan_zap.c: Prevent an infinite loop when config
processing gets to a jitterbuffer option
2006-10-31 22:02 +0000 [r46716] Jason Parker <jparker@digium.com>
* main/translate.c: Fix "core show translation" output. Issue
#8243, patch by Damin.
2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/translate.h, main/translate.c: add an API so
that translators can activate/deactivate themselves when needed
* include/asterisk/translate.h, main/translate.c: revert changes
that were the wrong way to address this... proper fix coming
* main/translate.c: let's set the seen flag early enough to
actually make a difference...
* include/asterisk/translate.h, main/translate.c: don't re-do setup
operations for translators that can dynamically register
themselves
2006-10-31 15:49 +0000 [r46663] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /: Blocked revisions 46662 via svnmerge ........ r46662 |
tilghman | 2006-10-31 09:46:04 -0600 (Tue, 31 Oct 2006) | 3 lines
Move thread-unsafe initializer to the module loading code; add
the corresponding function to the module unload to fix a memory
leak. ........
2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson <oej@edvina.net>
* main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue
#8089 - Fix the ENUM support (picking one record by number).
Thanks otmar!
* /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport
when we're supposed to support ;rport. Issue #7473.
* /, channels/chan_sip.c: If peer fails ACL check, fail peer at
REGISTER
* channels/chan_sip.c: Fix T38 too. Thanks, tgrman !
2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant <russell@digium.com>
* contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the
boot process to ensure it starts after stuff like MySQL (issue
#8253, Alric)
* /, main/utils.c: Merged revisions 46560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) |
3 lines When handling the case where the hostname is just an IPV4
numeric address, be sure to set the address type. (issue #8247,
alexr) ........
* /, res/res_agi.c: Merged revisions 46557 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) |
3 lines fix some copy/paste bugs in the checking of arguments for
the "control stream file" AGI command (issue #8255, mnicholson)
........
* main/translate.c: Add a small tweak to the code that checks to
see whether destination formats are translatable based on the
source format. If we have already determined that there is no
translation path in one direction, don't bother checking the
other direction.
2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming <kpfleming@digium.com>
* main/translate.c: when unregistering a translator, don't rebuild
the translation matrix unless needed when filtering formats out
of an offer, ensure we check for translation ability in both
directions
* include/asterisk/linkedlists.h: ensure that items removed from a
list are always unlinked from the list (next pointer set to NULL)
2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp <jcolp@digium.com>
* configure, configure.ac: Don't explicitly link in crypt as it is
not used on some platforms.
* channels/chan_iax2.c: We need to lock the pvt structure during
retransmission as another worker thread may be doing something as
well.
2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson <oej@edvina.net>
* main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h,
include/asterisk/doxyref.h, channels/chan_sip.c,
main/ast_expr2f.c, include/asterisk/module.h,
formats/format_ogg_vorbis.c, main/app.c,
include/asterisk/channel.h, include/asterisk/lock.h,
include/asterisk/frame.h: Issue #8246 - Doxygen fixes from
kshumard. An extra big thankyou is given to everyone that
contributes to doxygen! THANK YOU!
* main/rtp.c, /: Bind RTCP to the same IP as RTP
* /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302
redirects (imported from 1.2)
* /, channels/chan_sip.c: Issue #7608 - Notifications sent with
wrong content-type (imported from 1.2, modified)
* channels/chan_sip.c, CHANGES: Backport of patch for #7828 that
was reported for trunk, but obviously exists in 1.4 too.
* channels/chan_sip.c: Restoring the old logic, since working
around it and fixing it seemed too complicated. - The
SIP_OUTGOING flag indicates the direction of the last transaction
in the dialog. - The initreq stores the last request in the
dialog, the request that opened the latest transaction. Please
now retry all the 1.4 bug reports with mixed to/from headers,
tags etc in ACK, BYE, CANCEL. Thanks!
* channels/chan_sip.c: Accepting a message twice may be
misinterpreted...
* channels/chan_sip.c: - 183 is not reliable message... - Error
should not have SDP
2006-10-28 16:37 +0000 [r46377] Joshua Colp <jcolp@digium.com>
* utils/Makefile: Don't build muted on OpenBSD, it is not
supported.
2006-10-27 19:03 +0000 [r46370] Russell Bryant <russell@digium.com>
* channels/chan_zap.c: move the copy of the default settings to the
global settings back out of process_zap, so that they aren't
overwritten when process_zap is called multiple times
2006-10-27 18:29 +0000 [r46367] Olle Johansson <oej@edvina.net>
* contrib/asterisk-ng-doxygen: Put some doxygen pressure on
Christian :-)
2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant <russell@digium.com>
* main/asterisk.c, res/res_agi.c, apps/app_externalivr.c,
res/res_musiconhold.c: We should always be using _exit() after a
fork() or vfork() instead of exit(). This is because exit() does
some extra cleanup which in some implementations of vfork(), for
example, can actually modify the state of the parent process,
causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
* /: Blocked revisions 46361 via svnmerge ........ r46361 | russell
| 2006-10-27 12:36:07 -0500 (Fri, 27 Oct 2006) | 5 lines We
should always be using _exit() after a fork() or vfork() instead
of exit(). This is because exit() does some extra cleanup which
in some implementations of vfork(), for example, can actually
modify the state of the parent process, causing very weird bugs
or crashes. (issue #7971, Nick Gavrikov) ........
* channels/chan_zap.c: Instead of iterating all of the options once
to look for jitterbuffer options, and then again for everything
else, move the processing of jitterbuffer options into the main
loop so that there are no erroneous messages about ignoring
unknown options. (issue #8226)
2006-10-27 10:03 +0000 [r46351-46353] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
Merged revisions 46350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) |
1 line fixed a bug which caused chan_misdn to try to allocate 2
times the same channel on high load, which then caused
instability of mISDN. removed a useless function from isdn_lib.c
........
* channels/misdn_config.c: fixed not compile issue, which was just
introduced
* channels/misdn_config.c, channels/chan_misdn.c, /,
channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
Merged revisions 46176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) |
1 line added nttimeout option to configure wether we disconnect
calls on NT timeouts or not during an overlapdial session
........
2006-10-26 17:57 +0000 [r46335-46340] Jason Parker <jparker@digium.com>
* /, contrib/scripts/astgenkey.8: Merged revisions 46337 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2
lines oops - somebody forgot to change this - long ago, probably.
........
* CHANGES: grammar check
2006-10-26 16:38 +0000 [r46331] Olle Johansson <oej@edvina.net>
* CHANGES: Corrections to changes (Multiparking is not included)
2006-10-26 16:31 +0000 [r46329] Russell Bryant <russell@digium.com>
* main/translate.c: - If the source has no audio or no video
portion, do not call powerof() to get the format index. - Don't
run through the audio and video loops if there is no audio or
video portion of the source If 0 is passed to powerof, it will
return -1. This value of -1 was then being used as an array index
in these loops, which caused a crash on some systems. Other than
this issue, this code works as we expected it to. If a format is
not in the source, and we have to translation path to it, it is
not offered in the list of acceptable destination formats. (fixes
issue #8231)
2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming <kpfleming@digium.com>
* CHANGES: update to reflect G.722 addition
2006-10-26 04:18 +0000 [r46298] Russell Bryant <russell@digium.com>
* doc/backtrace.txt: update backtrace documentation to reflect
changes in 1.4 (issue #8230, kshumard)
2006-10-26 01:37 +0000 [r46287] Mark Spencer <markster@digium.com>
* main/config.c, main/manager.c: Fix config comment code
preservation code (thanks murf!)
2006-10-25 20:14 +0000 [r46276] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Old todo note - Don't add Contact header on
BYE and Cancel
2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant <russell@digium.com>
* configure.ac: fix error output when checking for openh323 to
refer to openh323 instead of pwlib (issue #8222, misaksen)
2006-10-25 19:16 +0000 [r46252] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Somewhat ugly code to try to fix issue
#7608. Since the problem was not very well defined, the fix is a
bit fuzzy too... Thanks to Luigi for accidentally spotting the
possible problem!
2006-10-25 19:08 +0000 [r46249] Russell Bryant <russell@digium.com>
* apps/app_queue.c: update warning message to include "agi" option
(issue #8225, jmls)
2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile: use 1.4.3 extra sounds with corrected silence
files
* sounds/sounds.xml, sounds/Makefile: add support for prebuilt
G.722 prompts and music on hold files
2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: show settings doesn't produce a list of
similar objects, it should stay a "show"
2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming <kpfleming@digium.com>
* main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c,
channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c,
pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c,
main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c,
cdr/cdr_custom.c, channels/chan_mgcp.c,
apps/app_parkandannounce.c, apps/app_voicemail.c,
channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c,
res/res_adsi.c, main/utils.c, apps/app_ices.c,
pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c,
apps/app_getcpeid.c: apparently developers are still not aware
that they should be use ast_copy_string instead of strncpy... fix
up many more users, and fix some bugs in the process
2006-10-25 04:58 +0000 [r46165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/pbx.c: WaitExten truncates decimals of times to wait,
instead of accepting them (Bug 8208)
2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming <kpfleming@digium.com>
* main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c,
channels/chan_h323.c, channels/chan_iax2.c,
include/asterisk/frame.h: add passthrough and file format support
for G.722 16KHz audio (issue #5084, original patch by andrew,
updated by mithraen)
* channels/chan_sip.c, main/translate.c: code zone experiment:
don't offer formats in the outbound INVITE that aren't either
passthrough or translatable
* main/translate.c: if multiple translators are registered for the
same source/dest combination, ensure that the lowest-cost one is
always inserted earlier in the list
2006-10-24 20:30 +0000 [r46142] Mark Spencer <markster@digium.com>
* res/res_agi.c: Fix FastAGI when there is no pid (bug #7628,
#8147)
2006-10-24 19:29 +0000 [r46130] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: We need to initialize our scheduler pthread
condition... yes.
2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo <rizzo@icir.org>
* main/http.c: merge 45152 don't leak descriptors in http.c
* channels/chan_sip.c: merge 45966 refer_to_domain potentially
containing options
* channels/chan_sip.c: merge 46026 improper checks on get_header()
return values
* channels/chan_sip.c: merge 46045 prevent NULL args to
ast_strdupa() in chan_sip.c
2006-10-24 05:23 +0000 [r46093] Russell Bryant <russell@digium.com>
* Makefile: Restore the ability to remove the firmware directory
without causing the installation to fail (issue #8111)
2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming <kpfleming@digium.com>
* main/translate.c: ensure that the translation matrix is properly
lock-protected every place it is used
* include/asterisk/translate.h, main/translate.c: add an API call
to allow channel drivers to determine which media formats are
compatible (passthrough or transcode) with the format an existing
channel is already using
* doc/imapstorage.txt: simplify and correct voicemail IMAP storage
build instructions
2006-10-24 03:01 +0000 [r46078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/channel.c: Pass through a frame if we don't know what it is,
rather than trying to pass a NULL, which will segfault a channel
driver (Bug 8149)
2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant <russell@digium.com>
* utils/muted.c, utils/ael_main.c: In muted.c, check the return
value of strdup. In ael_main.c, check the return value of calloc.
(issue #8157) In passing fix a few minor bugs in ael_main.c. The
last argument to strncpy() was a hard-coded 100, where it should
have been 99. I changed this to use sizeof() - 1.
* apps/app_meetme.c: Fix the descriptions of some of the
MeetMeAdmin options (issue #8098, mflorell)
* res/res_jabber.c: don't crash when an incoming message has no
"from" (issue #8205, jmls)
2006-10-23 00:27 +0000 [r45928] Joshua Colp <jcolp@digium.com>
* /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2
lines Don't leak memory mmmk? ........
2006-10-22 21:44 +0000 [r45916] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21
Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and
couldn't be initialized it would cause a segfault after 'reload'.
Reported by Drew/Matt thx. ........
2006-10-21 18:49 +0000 [r45818] Russell Bryant <russell@digium.com>
* res/res_monitor.c: Add a couple missing unregistrations of
manager actions and remove duplicate unregistrations of
applications. (issue #8194, jmls)
2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp <jcolp@digium.com>
* main/loader.c: Don't use promotion on Darwin because it doesn't
seem to work quite right in all cases, this should solve the
unresolved symbol issue people have been seeing.
* Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get
installed in the proper location (reported on asterisk-dev
mailing list)
2006-10-20 07:44 +0000 [r45741] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Let's understand SIP: - REFER can create
dialog, Asterisk does not support it yet - NOTIFY can create
dialog in Asterisk's implementation (voicemail) even though we
don't support the server side of it. In this case, the standard
is a side issue ;-) - Added extened functionality for unsupported
methods (PING, PUBLISH) so we don't create PVT's for those
either. Russellb needs to judge what to do with this in 1.2, but
I think the current implementation n 1.2 is a bug since we're
sending bad replies to NOTIFY and REFER outside of dialogs
2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp <jcolp@digium.com>
* res/res_jabber.c: Let's remember to unregister JabberStatus too
(issue #8184 reported by jmls)
* /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct
2006) | 2 lines Respect language selection when seeing if the
file exists (issue #8178 reported by mnicholson) ........
* channels/chan_sip.c: If the jitterbuffer is forced on then we
can't partially bridge (reported by wangster on #asterisk-dev)
2006-10-19 00:59 +0000 [r45622] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Don't leak the actual thread-specific
sip_pvt struct
2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: don't leak memory when a chan_sip thread is
destroyed that has a thread-local temp_pvt allocated
2006-10-18 21:03 +0000 [r45595] Joshua Colp <jcolp@digium.com>
* main/asterisk.c: Don't modify things if we are using vfork as
this is very bad and may cause unexpected behavior (issue #7970
reported by Nick Gavrikov)
2006-10-18 11:54 +0000 [r45517] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: remove duplicate declarations
2006-10-18 04:09 +0000 [r45464] Luigi Rizzo <rizzo@icir.org>
* main/http.c: merge from trunk: move ast_variables_destroy() to a
better place in handle_uri() to avoid leaking memory on non
existing files.
2006-10-18 03:02 +0000 [r45452] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Don't segfault if you're using a channel driver that
doesn't turn RTCP on
2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant <russell@digium.com>
* main/channel.c: Don't attempt to access private data members of
the pthread_mutex_t object, because this does not work on all
linux systems. Instead, just access the reentrancy field in the
ast_mutex_info struct when DEBUG_THREADS is enabled. If
DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
DEBUG_THREADS on as well. (issue #8139, me)
* configs/sip_notify.conf.sample: update entry to reboot a snom
phone (issue #7850, pnlarsson)
2006-10-17 Kevin P. Fleming <kpfleming@digium.com>
* Asterisk 1.4.0-beta3 released.
2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/stringfields.h, main/ast_expr2.c,
main/channel.c, channels/chan_sip.c, channels/chan_iax2.c:
optimize the 'quick response' code a bit more... no more malloc()
or memset() for each response expand stringfields API a bit to
allow reusing the stringfield pool on a structure when needed,
and remove some unnecessary code when the structure was being
freed
2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Don't create a "real" pvt structure for
requests that shouldn't be able to create one. Instead use a
temporary pvt and fill it with enough information so we can send
a reply.
2006-10-17 17:39 +0000 [r45329] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Adding information about Marks
direct-RTP hack to the docs...
2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming <kpfleming@digium.com>
* LICENSE: provide licensing language for IAXy firmware file
2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, apps/app_directed_pickup.c: Backport of new
directed pickup (BE-85).
2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson <oej@edvina.net>
* CREDITS: Adding Inotel to credits for SIP transfers. Thanks for
your support!
* channels/chan_sip.c: Don't destroy dialog for unexpected REFER
response...
2006-10-14 04:38 +0000 [r45143] Steve Murphy <murf@digium.com>
* funcs/func_rand.c: update the doc string for both AEL and
extensions.conf users.
2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming <kpfleming@digium.com>
* main/acl.c don't drop the entire permit/deny list when an attempt
is made to add an invalid entry (BE-92)
2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp <jcolp@digium.com>
* res/res_speech.c: Clear the quiet flag too since we are
restarting a recognition again (reported on -dev by Stephan
Edelman)
* res/res_speech.c: Check return value from engine in case of
failure (ie: out of licenses) (reported on -dev mailing list)
2006-10-13 20:52 +0000 [r45103] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-vtest17 (added),
pbx/ael/ael-test/ael-vtest17/extensions.ael (added),
pbx/ael/ael-test/ael-vtest17 (added),
pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in
this release via these changes
2006-10-13 19:19 +0000 [r45088] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: avoiding warning, fixing potential bug
2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp <jcolp@digium.com>
* codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c,
codecs/lpc10/decode.c, codecs/lpc10/dcbias.c,
codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c,
codecs/lpc10/difmag.c, codecs/lpc10/hp100.c,
codecs/lpc10/synths.c, codecs/lpc10/preemp.c,
codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c,
codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c,
codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c,
codecs/lpc10/lpcini.c, codecs/lpc10/random.c,
codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c,
codecs/lpc10/placea.c, codecs/lpc10/tbdm.c,
codecs/lpc10/analys.c, codecs/lpc10/onset.c,
codecs/lpc10/energy.c, codecs/lpc10/deemp.c,
codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c,
codecs/lpc10/median.c, codecs/lpc10/encode.c,
codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c,
codecs/lpc10/invert.c: And file said... let the compiler warnings
STOP!
* apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136
reported by mnicholson)
* apps/app_playback.c: Move say.conf existence check to do_say
function since it is called from multiple places (issue #8144
reported by kshumard)
2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c: when sending a call to a peer, use the proper socket if
we have multiple bindings (reported on asterisk-dev)
2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Complete merging in RPID screen changes
(issue #8101 reported by hristo, patch by oej in revision 44757)
* main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add
the background refresh item back into the scheduler if enabled
since it is deleted during reload. (issue #8142 reported by
p_lindheimer)
2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming <kpfleming@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/utils.c: use a configure script test for PMTU discovery
control instead of just assuming it's available on Linux
2006-10-13 14:45 +0000 [r44994-45026] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some
echocandisable issues when bridged. this caused a kernel panic
sometimes.. also some minor formatting fixes
* channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause
got a wrong isdn cause at RELEASE_COMPLETE
2006-10-12 22:07 +0000 [r44992] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: merge formatting and minor code
simplifications from trunk
2006-10-12 20:34 +0000 [r44982] Matt O'Gorman <mogorman@digium.com>
* channels/chan_gtalk.c: fix for bug 7764.
2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: we can only send one 'a=ptime' attribute per
media session, not one for each format
* main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c,
main/utils.c: ensure that IAX2 and SIP sockets allow UDP
fragmentation when running on Linux (thanks to Brian Candler on
the asterisk-dev list for the tip)
2006-10-12 16:56 +0000 [r44945] Russell Bryant <russell@digium.com>
* main/manager.c: fix a silly typo in a comment that I saw while
reading the commit list
2006-10-12 16:08 +0000 [r44942] Joshua Colp <jcolp@digium.com>
* Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue
#8135 reported by ssokol)
2006-10-12 12:55 +0000 [r44921] Nadi Sarrar <ns@beronet.com>
* main/manager.c: append_event must be called while holding the
session lock
2006-10-12 10:24 +0000 [r44911] Russell Bryant <russell@digium.com>
* res/res_jabber.c: change some debug output to use LOG_DEBUG
instead of verbose output
2006-10-11 16:57 +0000 [r44888] Jason Parker <jparker@digium.com>
* main/db1-ast/Makefile: These are already set by the parent
Makefile.. There is no need to have this here (it doesn't
actually work anyways).
2006-10-11 09:18 +0000 [r44854] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c: removed warning because of missing
prototype declaration
2006-10-10 19:23 +0000 [r44830] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Do not set default/global values in the
variable declaration, set it in reload_config()
2006-10-10 17:21 +0000 [r44819] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Move some stuff around so that a NOTIFY
dialog won't hang around until the end of the world under certain
circumstances
2006-10-10 16:44 +0000 [r44809] Paul Cadach <paul@odt.east.telecom.kz>
* main/channel.c, funcs/func_channel.c, include/asterisk/channel.h:
CHANNEL() function sometime mix parameter and value
2006-10-10 16:42 +0000 [r44808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* funcs/func_logic.c: Lost of a bit of logic when this was
simplified between 1.2 and 1.4 (Bug 8117)
2006-10-10 16:30 +0000 [r44806] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Bail out if we have no refer structure and
we get a refer response
2006-10-10 16:21 +0000 [r44805] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: more merge from trunk (comments and change a
static function name)
2006-10-10 15:23 +0000 [r44788] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Only set DTMF information if an RTP
structure exists
2006-10-10 13:50 +0000 [r44786] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added
support of dynamically enabling hdlc on bchannels
2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: whitespace changes related to previous
commit
* channels/chan_sip.c: merge a few code simplifications that have
gone into trunk during last week, to reduce differences between
the two branches and make porting fixes easier.
2006-10-09 16:12 +0000 [r44764] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Fix a problem where phones that go
"missing" never got unregistered. Issue #8067, reported by pj,
patch by Anthony LaMantia (with minor whitespace modifications)
2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid
the deadlock
* channels/chan_iax2.c: Properly avoid a collision with iax2_hangup
(issue #8115 reported by vazir)
2006-10-08 14:14 +0000 [r44746] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: do not dereference p if we
know it is NULL
2006-10-07 14:39 +0000 [r44684] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx, channels/chan_h323.c,
channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate
caller's transfer capability too
2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: put common code in a
function to avoid repetitions.
* channels/chan_sip.c: remove hardwired usage of 5060, use
DEFAULT_SIP_PORT instead
* channels/chan_sip.c: option_debug checking
before printing to debug channel.
* channels/chan_sip.c: backport simplifications on sip_register,
usage of ast_set2_flag(), and fixes to the handling of failed
module loading.
* channels/chan_sip.c: improve and document function
get_in_brackets(), introducing a helper function
find_closing_quote() of more general use.
2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/linkedlists.h: ensure that mutex locks inside
list heads are initialized properly on platforms that require
constructor initialization (issue #8029, patch from timrobbins)
* CHANGES: remove Jingle as per mog
2006-10-06 21:08 +0000 [r44628] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Remove the seqno check for RFC2833, the handler is
smart enough to not need it.
2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming <kpfleming@digium.com>
* CHANGES: various cleanups
2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp <jcolp@digium.com>
* main/rtp.c: When the sequence number rolls over then reset the
recorded sequence number for DTMF (issue #8106 reported by
bungalow)
* main/file.c: Even more frames to treat as though the remote side
disappeared (issue #8097 reported by eldadran)
2006-10-06 15:59 +0000 [r44567] Luigi Rizzo <rizzo@icir.org>
* main/manager.c, main/http.c: make sure sockets are blocking when
they should be blocking.
2006-10-06 12:53 +0000 [r44559-44563] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: fixed segfault which happens during
hold/transfer action
* channels/chan_misdn.c: if INFORMATION Message come with keypad
instead of called party number, we just use the keypad as called
party number.
* 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:
added the option 'reject_cause' to make it possible to set
the RELEASE_COMPLETE - cause on the 3. incoming PMP channel,
which is automatically rejected because chan_misdn does not
support that kind of callwaiting. Therefore chan_misdn supports
now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc
now gets the info if the requested channel is incoming or
outgoing to make the 3. channel possible
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c: fixed the hold/retrieve/transfer issues,
removed a useless bc field, added setting of frame.delivery fields,
some minor code cleanups
2006-10-05 19:57 +0000 [r44502] Joshua Colp <jcolp@digium.com>
* main/file.c: Treat busy control frames as hangup in the file streaming
core (issue #8097 reported by eldadran)
2006-10-05 18:21 +0000 [r44488] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang.
Many thanks to Doug!
2006-10-05 18:01 +0000 [r44486] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite
hanging by a thread if the other side is already setup with T.38
2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming <kpfleming@digium.com>
* main/app.c: don't segfault when an argument without a close
parenthesis is found stop parsing as soon as that situation
occurs
2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy <murf@digium.com>
* CHANGES: I put the accumulated changes from the commit logs and
inspection, into CHANGES. Hope everyone approves!
* configs/muted.conf.sample, utils/muted.c: Hang on a minute, the
install process sticks muted.conf in /etc/asterisk, so that's
where muted should look for it, right?
2006-10-05 02:40 +0000 [r44450] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Don't totally bail out if T.38 was
negotiated
2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: fix Polycom presence notification again
2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo <rizzo@icir.org>
* utils/Makefile: as far as i can tell astman only uses newt...
* Makefile: put linker flags in ASTLDFLAGS where they belong
2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE
requests add workaround for new Polycom firmware SUBSCRIBE
requests (bug is known to exist in 2.0.1 firmware)
* include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually
work
2006-10-04 19:57 +0000 [r44380] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c,
pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6,
pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8,
pbx/ael/ael-test/ael-test16/extensions.ael (added),
pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y,
pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the
problems reported in bug 8090
2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_oss.c, main/cdr.c, channels/chan_phone.c,
main/manager.c, pbx/pbx_spool.c, res/res_smdi.c,
channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c,
main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c,
include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c,
channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c,
main/devicestate.c, main/utils.c, res/res_musiconhold.c,
channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update
thread creation code a bit reduce standard thread stack size
slightly to allow the pthreads library to allocate the stack+data
and not overflow a power-of-2 allocation in the kernel and waste
memory/address space add a new stack size for 'background'
threads (those that don't handle PBX calls) when LOW_MEMORY is
defined
2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy <murf@digium.com>
* configs/muted.conf.sample: I've been meaning to add some
explanation about muted... here it is
* configs/manager.conf.sample: CLI reverbification update to this
config file
* apps/app_macro.c: In response to bug 7776, a Warning has been
added to the doc string for Macro().
2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming <kpfleming@digium.com>
* main/asterisk.c, main/loader.c, main/term.c, Makefile,
include/asterisk.h: ensure that local include files are always
used avoid a duplicate function name (term_init())
2006-10-03 22:35 +0000 [r44312] Matt O'Gorman <mogorman@digium.com>
* channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing
client without resource.
2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_queue.c: fix a logic error in my previous fix to the queue
reload code
2006-10-03 18:42 +0000 [r44286] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Change default presentation indicator
to "user provided not screened" if octet 3a missed in
CallingPartyNumber IE
2006-10-03 18:35 +0000 [r44284] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Use VideoSupport instead so it is considered
a valid XML attribute name. (issue #8075 reported by renemendoza)
2006-10-03 18:30 +0000 [r44283] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Fix preparation of type and
presentation of calling number
2006-10-03 00:01 +0000 [r44240] Matt O'Gorman <mogorman@digium.com>
* doc/jingle.txt, channels/chan_jingle.c (removed),
include/asterisk/jabber.h, configs/jingle.conf.sample (removed),
res/res_jabber.c: updated res_jabber for even better component
support, soon will be jep-0100 compliant. also removed
chan_jingle and infromed info from jingle.txt, chan_gtalk still
works and should be used in this version.
2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Change the fd on the I/O context in case it
changed during the reload, which is indeed possible. (issue #7943
reported by eclubb)
* contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
instead of hardcoding the path for the error message (issue #7942
reported by eclubb)
2006-10-02 18:52 +0000 [r44186] Paul Cadach <paul@odt.east.telecom.kz>
* configs/users.conf.sample, pbx/pbx_config.c: Missed part of
userconf functionality for chan_h323
2006-10-02 17:25 +0000 [r44169] Joshua Colp <jcolp@digium.com>
* main/io.c: Shrink when current_ioc is unused. It is set to -1 when
unused, not 0. (issue #7941 reported by eclubb)
2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach <paul@odt.east.telecom.kz>
* doc/realtime.txt: Typo fix
* channels/chan_h323.c: Optimization of oh323_indicate(): less
locks - less problems, plus single exit point
2006-10-02 02:38 +0000 [r44146] Mark Spencer <markster@digium.com>
* channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when
you're not talking about a channel :)
2006-10-01 19:32 +0000 [r44135] Paul Cadach <paul@odt.east.telecom.kz>
* channels/chan_h323.c: Do not simulate any audio tones if we got
PROGRESS message
2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant <russell@digium.com>
* Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to
be empty. The cause is that since ASTDATADIR is explicitly
exported using "export ASTDATADIR" at the top of the Makefile,
make no longer considers the variable "undefined", so the
Makefile can't use ?= to set ASTDATADIR if not yet set. (issue
#8063, reported by akohlsmith, fixed by me)
* configs/queues.conf.sample: Fix the name of the "eventmemberstatus"
option in the sample queues.conf (issue #8065, adamg)
2006-10-01 15:01 +0000 [r44109] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: sync with trunk - move variable declarations
to the beginning of a block.
2006-09-30 19:20 +0000 [r44090] Paul Cadach <paul@odt.east.telecom.kz>
* main/rtp.c: Allow one-way RTP streams (device->Asterisk)
2006-09-30 16:28 +0000 [r44080] Luigi Rizzo <rizzo@icir.org>
* codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent
build problems: - with AST_DEVMODE, building codecs/lpc10 fails
because of lots of warnings, and the configure step in editline
fails as well. Fix this by removing the -Werror in these steps. -
on FreeBSD (but probably on other platforms as well), the final
link of asterisk fails because AST_LIBS was not exported to the
subdirs Makefiles. Add a proper fix in the top-level Makefile (a
possible alternative way is to add "export AST_LIBS" near the
beginning of the file). With this fix, i believe that some of the
platform-specific conditionals in main/Makefile are redundant
(because they should be already dealt with in the top level
Makefile) but i don't have a platform to check. Merging to head
will happen in a moment.
2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <paul@odt.east.telecom.kz>
* channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment
of previous fix: Issue #7928 - Don't send both 404 and 503. Fix
by phsultan with a small fix by me, myself or I. Thanks,
Philippe! (This was caused by my changes to the transaction
handling)
* channels/chan_sip.c: Found some buggy SIP clients (phones Planet
VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
sends ACK not on OK message only (when remote party answers) but
on RINGING message too, so when we send 200 OK message, we get
unidentified ACK message (because INVITE acknowledged on RINGING
message already), so 200 OK retransmits within its retransmission
interval then call gets dropped. If someone else knows how to
provide workaround for such cases, please, fix it in correct way.
Thanks to ssh from #asteriskru for provide access to his box to
study and fix this case.
2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming <kpfleming@digium.com>
* agi, utils: ignore temporary files made by the Makefiles during a
build
* codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile,
codecs/Makefile, utils/Makefile, configure,
build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac,
Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile,
pbx/Makefile, res/Makefile, channels/Makefile: fix a few build
system bugs, and convert Makefiles to be compatible with GNU make
3.80
2006-09-29 22:35 +0000 [r44053] Jason Parker <jparker@digium.com>
* main/asterisk.c, main/cli.c: Fix a bug with the removal of
'atleast' argument to 'core verbose' and 'core debug'. Add that
argument back in.
2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more
carefully when no CallingNumber IE available
* channels/h323/ast_h323.cxx: Fake display name by called number on
incoming calls (until passing connected number/connected name is
not implemented)
* channels/h323/ast_h323.cxx: Ported code refers to H.450 - add
includes
* channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly
pass TON/PRESENTATION information - original
H323Connection::SendSignalSetup() destroys Q.931 fields.
2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming <kpfleming@digium.com>
* main/Makefile: yet another place where we were not using the
correct CFLAGS by default
* main/Makefile: missed one conversion to ASTCFLAGS
2006-09-29 18:30 +0000 [r44009] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx, channels/chan_h323.c,
channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass
TON/PRESENTATION information too
2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming <kpfleming@digium.com>
* main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile,
main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse
CFLAGS and LDFLAGS for build of Asterisk components, because they
are also then used for non-Asterisk components (like menuselect);
use our own variables instead
* configure, configure.ac: support --without-curl in configure
script
* Makefile.rules: another cross-compile fix
* Makefile: a couple more environment settings that can't leak into
the menuselect build
* main/cli.c: proper fix for ast_group_t change
* include/asterisk/lock.h: eliminate compiler warning when
DEBUG_CHANNEL_LOCKS is enabled and users of this header file
don't also include channel.h
2006-09-28 20:11 +0000 [r43944] Jason Parker <jparker@digium.com>
* apps/app_queue.c: Fix incorrect argument order for member names,
on persisted members. Issue 8047, patch by jmls.
2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp <jcolp@digium.com>
* apps/app_playback.c, res/res_monitor.c,
include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c,
channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c,
main/udptl.c, main/frame.c, funcs/func_timeout.c,
channels/chan_sip.c, apps/app_festival.c,
channels/iax2-provision.c, apps/app_alarmreceiver.c,
res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c:
Put in missing \ns on the end of ast_logs (issue #7936 reported
by wojtekka)
2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_queue.c: fix buggy (and overly complex) loop used during reload
of app_queue for static member list updating
2006-09-28 17:34 +0000 [r43918] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Extend call establishment timeout
2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Make sure the pvt exists before accessing
it again as it may have gone away (issue #7562 reported by Seb7
and issue #7939 reported by sorg)
* main/cli.c: Warning be gone!
2006-09-28 16:41 +0000 [r43899] BJ Weschke <bweschke@btwtech.com>
* apps/app_queue.c: app_queue is comparing the device names incorrectly
while checking their statuses. It's internal list of interfaces
includes the dial string, while the argument passed to this
function does not have the dial string (/n for a local channel).
This causes it to ignore the device state changes because it
thinks it belongs to none of its members. (#8040 reported and
patch by tim_ringenbach)
2006-09-28 16:17 +0000 [r43893] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Stop the stream after waitstream returns so that our
formats get restored. (issue #7370 reported by kryptolus)
2006-09-28 15:56 +0000 [r43877] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Fix compiler warning
2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke <bweschke@btwtech.com>
* apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 -
tim_ringenbach reported and patched)
* apps/app_queue.c: Autopause not working for queue members. (#8042
- jmls reported and patch)
2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force
remote side to start media on outgoing PROGRESS message
* include/asterisk/compiler.h: Put attribute tag at correct place
2006-09-28 11:03 +0000 [r43852] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c: fixed a bug which led to chan_list zombies,
when the call could not be properly established in misdn_call.
also removed the ACK_HDLC stuff which is not really needed.
2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Do not open transmit channel until
TCS is received
* main/file.c: Don't warn on HOLD/UNHOLD control frames
* main/file.c: Don't treat unknown control frames as voice
2006-09-27 20:21 +0000 [r43816] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Avoid inability to lock directory log message by
creating the directory ahead of time. (Issue 7631)
2006-09-27 19:44 +0000 [r43801-43803] Jason Parker <jparker@digium.com>
* apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS
not being set under certain circumstances. Fix a minor issue, to
make it use the filenames that were parsed, instead of the entire
argument string. Fix Background() to return -1 like Playback(),
if no args are specified.
2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Compensate for out of order packets better if RFC2833
compensation is turned on.
* channels/chan_iax2.c: Get rid of two functions from a time now
past (we THINK these are from pre-recursive lock time) that may
be contributing to two open issues on the bug tracker (7562/7939)
and that has the potential to just make bad things happen if the
timing is right.
2006-09-27 16:55 +0000 [r43779] Russell Bryant <russell@digium.com>
* main/channel.c,res/res_features.c: Fix a problem that occurred if
a user entered a digit
that matched a bridge feature that was configured using multiple
digits, and the digit that was pressed timed out in the feature
digit timeout period. For example, if blind transfer is
configured as '##', and a user presses just '#'. In this
situation, the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by
michaels and valuable input provided by mneuhauser and kuj. Fixed
by me, with testing help and peer review from Joshua Colp). There
are a couple of issues involved in this fix: 1) When
ast_generic_bridge determines that there has been a timeout, it
returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets
this result, it calls ast_generic_bridge over again with the same
timestamp for the next event. This results in an endless loop of
nothing until the call is terminated. This is resolved by simply
changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it
sees a timeout. 2) I also changed ast_channel_bridge such that if
in the process of calculating the time until the next event, it
knows a timeout has already occured, to immediately return
AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
anyway. 3) In the process of testing the previous two changes, I
ran into a problem in res_features where ast_channel_bridge would
return because it determined that there was a timeout. However,
ast_bridge_call in res_features would then determine by its own
calculation that there was still 1 ms before the timeout really
occurs. It would then proceed, and since the bridge broke out and
did *not* return a frame, it interpreted this as the call was
over and hung up the channels. The reason for this was because
ast_bridge_call in res_features and ast_channel_bridge in
channel.c were using different times for their calculations.
channel.c uses the start_time on the bridge config, which is the
time that the feature digit was recieved. However, res_features
had another time, 'start', which was set right before calling
ast_channel_bridge. 'start' will always be slightly after
start_time in the bridge config, and sometimes enough to round up
to one ms. This is fixed by making ast_bridge_call use the same
time as ast_channel_bridge for the timeout calculation. ........
2006-09-27 16:24 +0000 [r43775] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, channels/Makefile: removed the chan_misdn
versioning, since Asterisk has it's own
2006-09-27 16:23 +0000 [r43774] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Make rfc2833compensate a global option.
2006-09-27 04:35 +0000 [r43756] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: Backport revision 43754 from the trunk,
which removes an unused buffer from mm_login to close bug 8038,
as well as addresses some formatting and coding guidelines issues
in passing. Originally, I did not commit this to 1.4 since it is
not necessarily fixing a bug. However, since the IMAP storage
code is brand new, I decided it would be better to make the
change here as well, in case someone has to work on this code to
address issues in the very near future. I don't want to make
unnecessary merge problems going to the trunk.
2006-09-27 02:32 +0000 [r43739] Steve Murphy <murf@digium.com>
* configs/extensions.ael.sample: This change to extensions.ael was
to fix bug 8031; the install scripts are causing it to be copied
to /etc/asterisk/extensions.ael, and because it is a fairly
direct conversion of the original extensions.conf, the macro and
context names clash with the existing extensions.conf. So, I put
an ael- in front of all macros and contexts, and checked every
goto and macro call. Also, this file compiles under aelparse.
2006-09-26 20:56 +0000 [r43710] Russell Bryant <russell@digium.com>
* main/asterisk.c: Back in revision 4798, this message was changed from
using ast_cli() to directly calling write(). During this change,
checking if this was a remote console was removed. This caused
this message about using "exit" or "quit" to exit an Asterisk
console to come up in times where it did not make sense. This
change restores the check to see if this is a remote console
before printing the message. (fixes BE-65)
2006-09-26 20:47 +0000 [r43707] Joshua Colp <jcolp@digium.com>
* .cleancount, main/cli.c, channels/chan_sip.c,
include/asterisk/channel.h: Use proper type to represent the group variable
(issue #8025 reported by makoto)
2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Add missing newline character in the warning
message about deprecated TOS values in configuration.
* apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain
mailbox definitions, don't introduce a length limit on the
definition by using a 256 byte temporary storage buffer. Instead,
make the temporary buffer just as big as it needs to be to hold
the entire mailbox definition. (fixes BE-68)
2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c: Strip options off the argument passed for
devicestate in chan_local. (issue #8034 reported by pcardozo)
* apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight
overhaul of the whisper support. 1. We need to duplicate the
frame from ast_translate 2. We need to ensure we always have
signed linear coming in for signed linear combining. 3. We need
to ensure we are always feeding signed linear out. 4. Properly
store and restore write format when beeping on the channel we are
whispering on. 5. Properly discontinue the stream on the channel
for the beep. (issue #8019 reported by timkelly1980)
2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile: update to use 1.4.3 core sounds, with corrected
beep/beeperr/tt-monkeys files
2006-09-26 18:08 +0000 [r43650-43674] Jason Parker <jparker@digium.com>
* doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by
Dan Austin. Maximum values were incorrect, which is why this is
being put in 1.4
* channels/chan_skinny.c: Add proper codec support to chan_skinny.
Works with at least ulaw, alaw, and g729a. This is technically a
"new feature", but there are justifications for it. I found a bug
with the recent rtp packetization changes, which caused the media
setup to fail under certain circumstances, particularly when
using allow=all, or having no allow= statements (globally or on
the device). I could have either removed the rtp packetization
features, or I could add proper codec support (which, without, I
think most people would consider to be a bug anyways).
2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Should have moved these lines up in the
merge, instead of removing them
* apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1)
delete=yes was ignored 2) maxmessages was ignored
2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h,
channels/h323/cisco-h225.asn: Fix ASN1 description of
non-standard Cisco extensions
* channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport
changes of trunk: 1) r43540: Avoid possible deadlock on channel
destruction 2) r43590: Disable fastStart if requested by remote
side
2006-09-25 15:23 +0000 [r43616] Jason Parker <jparker@digium.com>
* sounds/Makefile: One more fix for sounds installation - this time
for portability. Reported to asterisk-dev mailing list.
2006-09-25 14:52 +0000 [r43605] Steve Murphy <murf@digium.com>
* formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from
crashing if trying to play an OGG moh file.
2006-09-25 06:15 +0000 [r43582] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/caps_h323.cxx, channels/h323/compat_h323.h,
channels/chan_h323.c: Merged revisions 43472,43495 from trunk
2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant <russell@digium.com>
* channels/iax2-provision.c: Fix a CLI command registration issue
where an erroneous message claiming that "iax2 show provisioning"
was already registered. This was because this command was
registering itself as both the command, as well as the command it
is deprecating. (issue #8022, reported by bjweeks, fixed by
myself)
* channels/chan_iax2.c:Check to see if the channel that is activating the
IAXPEER function is actually an IAX2 channel before proceeding to
process it to avoid crashing. (issue #8017, reported by admott,
fixed by myself)
2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming <kpfleming@digium.com>
* Makefile: don't output the 'build complete' message when the
target being run is already going to do an installation
2006-09-22 22:12 +0000 [r43518] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Allow chan_skinny.so to be unloaded
properly. Remove reload support, since it doesn't
actually...work.
2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: This commits a change to return
MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all
goes well for bug 8004
* pbx/pbx_ael.c: If the extensions.ael file not found, or
unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004.
2006-09-22 17:25 +0000 [r43492] Jason Parker <jparker@digium.com>
* main/cli.c: Make sure we explicitly set the CLI command to not be
deprecated, if it isn't.
2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile: use rebuilt extra sounds
* main/channel.c: all the Linux systems I have don't use
'__m_count' for this field, so I don't know where this came
from...
2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant <russell@digium.com>
* include/asterisk/threadstorage.h: backport the compatability fix
to use attribute_malloc instaed of __attribute__ ((malloc))
* channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN
could not be configured (issue #8006, Mithraen)
* main/frame.c: Suppress a compiler warning about the use of a
potentially uninitialized variable. It couldn't actually happen,
though.
2006-09-22 03:01 +0000 [r43469] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: First shot at unload_module in
chan_skinny.. More to come.
2006-09-21 23:50 +0000 [r43466] Matt O'Gorman <mogorman@digium.com>
* include/asterisk/jabber.h, channels/chan_gtalk.c,
res/res_jabber.c: updates for better compontent support
2006-09-21 23:24 +0000 [r43464] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we
actually documented how the new features in res_odbc actually
work. (Oops)
2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp <jcolp@digium.com>
* channels/chan_oss.c: Some more clean up in the load function for
chan_oss (issue #8002 reported by Mithraen with minor mods by
moi)
* channels/chan_mgcp.c: Clean up chan_mgcp's module load function
(issue #8001 reported by Mithraen with mods by moi)
2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming <kpfleming@digium.com>
* main/Makefile, build_tools/strip_nonapi (added): add another
attempt to strip non-API symbols from the final binary... script
will need to be extended to work on non-Linux systems
2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_url.c: Fix documentation to reflect how Url() really
works
* cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates
2006-09-21 Kevin P. Fleming <kpfleming@digium.com>
* Asterisk 1.4.0-beta2 released.
2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming <kpfleming@digium.com>
* main/Makefile: remove this change... it requires binutils 2.17
2006-09-20 23:19 +0000 [r43396] Jason Parker <jparker@digium.com>
* build_tools/make_version: fix minor typo in the way version is
handled
2006-09-20 Kevin P. Fleming <kpfleming@digium.com>
* Asterisk 1.4.0-beta1 released.
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