[svn-commits] kpfleming: tag 1.4.0-beta3 r45427 - /tags/1.4.0-beta3/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Oct 17 16:20:25 MST 2006
Author: kpfleming
Date: Tue Oct 17 18:20:24 2006
New Revision: 45427
URL: http://svn.digium.com/view/asterisk?rev=45427&view=rev
Log:
importing files for 1.4.0-beta3 release
Added:
tags/1.4.0-beta3/.lastclean (with props)
tags/1.4.0-beta3/.version (with props)
tags/1.4.0-beta3/ChangeLog (with props)
Added: tags/1.4.0-beta3/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.0-beta3/.lastclean?rev=45427&view=auto
==============================================================================
--- tags/1.4.0-beta3/.lastclean (added)
+++ tags/1.4.0-beta3/.lastclean Tue Oct 17 18:20:24 2006
@@ -1,0 +1,1 @@
+25
Propchange: tags/1.4.0-beta3/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.4.0-beta3/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.4.0-beta3/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.4.0-beta3/.version
URL: http://svn.digium.com/view/asterisk/tags/1.4.0-beta3/.version?rev=45427&view=auto
==============================================================================
--- tags/1.4.0-beta3/.version (added)
+++ tags/1.4.0-beta3/.version Tue Oct 17 18:20:24 2006
@@ -1,0 +1,1 @@
+1.4.0-beta3
Propchange: tags/1.4.0-beta3/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.4.0-beta3/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.4.0-beta3/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.4.0-beta3/ChangeLog
URL: http://svn.digium.com/view/asterisk/tags/1.4.0-beta3/ChangeLog?rev=45427&view=auto
==============================================================================
--- tags/1.4.0-beta3/ChangeLog (added)
+++ tags/1.4.0-beta3/ChangeLog Tue Oct 17 18:20:24 2006
@@ -1,0 +1,1024 @@
+2006-10-17 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.4.0-beta3 released.
+
+2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming <kpfleming at 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 at 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 at 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 at digium.com>
+
+ * LICENSE: provide licensing language for IAXy firmware file
+
+2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at beronet.com>
+
+ * channels/chan_misdn.c: avoiding warning, fixing potential bug
+
+2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * channels/chan_gtalk.c: fix for bug 7764.
+
+2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming <kpfleming at 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 at 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 at 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 at 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 at 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 at 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 at beronet.com>
+
+ * channels/misdn/isdn_lib.c: removed warning because of missing
+ prototype declaration
+
+2006-10-10 19:23 +0000 [r44830] Olle Johansson <oej at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * CHANGES: various cleanups
+
+2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * channels/chan_sip.c: fix Polycom presence notification again
+
+2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo <rizzo at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at odt.east.telecom.kz>
+
+ * main/rtp.c: Allow one-way RTP streams (device->Asterisk)
+
+2006-09-30 16:28 +0000 [r44080] Luigi Rizzo <rizzo at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx: Extend call establishment timeout
+
+2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp <jcolp at 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 at 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 at 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 at odt.east.telecom.kz>
+
+ * channels/h323/ast_h323.cxx: Fix compiler warning
+
+2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke <bweschke at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * channels/chan_sip.c: Make rfc2833compensate a global option.
+
+2006-09-27 04:35 +0000 [r43756] Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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
+
[... 146 lines stripped ...]
More information about the svn-commits
mailing list