diff options
author | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-10-17 23:20:24 +0000 |
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committer | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-10-17 23:20:24 +0000 |
commit | 0ade7fc1a758a046139d8c08da1113be3eba568c (patch) | |
tree | 43cb43f7a8053a062b339e0db48a40c188b536ce | |
parent | 80b59145da82176c62686b2768b18f9b80452b6c (diff) |
importing files for 1.4.0-beta3 release
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.0-beta3@45427 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | .lastclean | 1 | ||||
-rw-r--r-- | .version | 1 | ||||
-rw-r--r-- | ChangeLog | 1024 |
3 files changed, 1026 insertions, 0 deletions
diff --git a/.lastclean b/.lastclean new file mode 100644 index 000000000..7273c0fa8 --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +25 diff --git a/.version b/.version new file mode 100644 index 000000000..951c84835 --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.4.0-beta3 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..f418aaa5b --- /dev/null +++ b/ChangeLog @@ -0,0 +1,1024 @@ +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. |