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asterisk/ChangeLog
Kevin P. Fleming 91c932cb93 importing files for 1.4.0-beta3 release 
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.0-beta3@45427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-17 23:20:24 +00:00

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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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