forked from Mirrors/freeswitch
Update ChangeLog through r16270 (more to come soon)
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@16447 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
parent
78bd05cb65
commit
0fc54805a9
@ -45,6 +45,9 @@ freeswitch (1.0.5)
|
||||
build: add --enable-64 support for os x 10.4 and 10.5 (r:15874)
|
||||
build: add missing modules to modules.conf (FSBUILD-214/r:15879)
|
||||
build: fix mod_managed build on gcc 4.4 (r:15897)
|
||||
build: Disable rpath checking in freeswitch.spec (FSBUILD-217/r:16037)
|
||||
build: Remove libuuid from tree (r:16072)
|
||||
build: Avoid building static version of modules (e.g. mod_enum.a) by adding the "-shared" libtool option. (r:16225)
|
||||
config: improvements to French language handling (MODASRTTS-20/r:14911)
|
||||
config: Add valet_parking to default config (r:15124)
|
||||
config: Add valet macros (r:15156)
|
||||
@ -195,6 +198,17 @@ freeswitch (1.0.5)
|
||||
core: Add tab complete to windows (ESL-24/r:16010)
|
||||
core: switch_play_and_get_digits now returns SWITCH_FALSE if caller doesn't dial anything (DP-10/r:16011)
|
||||
core: adding 12kHz and 24kHz (r:16026)
|
||||
core: Fix dialplain looping over and over again when group_confirm_file is missing (DP-11/r:16063)
|
||||
core: trade some max call count for more accurate timing in full media situations, hint: use 30ms ptime for drastic reduction in resources (r:16081)
|
||||
core: add core param for controling timer (r:16093)
|
||||
core: Incorrect handling when rtp-timer-name=none causes medialess RTP sessions to never time out (FSRTP-10/r:16100)
|
||||
core: add SOCKET_EVENT event (r:16135)
|
||||
core: count runlevel to prevent dup calls to the core init routines (r:16140)
|
||||
core: fix power of 10 in cps counter (r:16144)
|
||||
core: add ;; delim to console (r:16166)
|
||||
core: fix mitm for audio for passing sas with zrtp (r:16204)
|
||||
core: setting jitter buffer keeps dtmf events from firing (FSCORE-523/r:16207)
|
||||
core: don't segfault when -nonat is used and nat_map reinit is called (r:16253)
|
||||
docs: Add large Doxygen update (thanks Muhammed Shahzad) (r:14973)
|
||||
docs: update es phrase file (MODAPP-317/r:15575)
|
||||
embedded_languages: Prevent unloading of embedded languages modules (also fixes MODLANG-121/r:14491)
|
||||
@ -213,6 +227,8 @@ freeswitch (1.0.5)
|
||||
libesl: use event_clone type internally to not confuse people who think they are getting events in response to commands (r:15660)
|
||||
libesl: hook new complete api up to FSAPI and export tab completion down to fs_cli (r:15956)
|
||||
libesl: change execute and executeAsync to return the last event instead of status since it will almost always be 0 (r:15973)
|
||||
libesl: add help banner on connection failure (ESL-25/r:16040)
|
||||
libesl: add support for user level auth to esl and fs_cli (r:16161)
|
||||
libiksemel: fix iksemel build against gnutls 2.x (FSBUILD-219/r:16019)
|
||||
libjs: pass ldflags to dso builds for libjs and nspr (r:16549)
|
||||
libportaudio: Dither code in mod_portaudio doesn't properly compile on 64 bit systems (r:15422)
|
||||
@ -261,6 +277,9 @@ freeswitch (1.0.5)
|
||||
mod_commands: fix db_cache's auto-complete (r:15969)
|
||||
mod_commands: add uuid_audio cli cmd (r:15989)
|
||||
mod_commands: check for zstr(cmd) in escape_function (r:16029)
|
||||
mod_commands: Fix group_call_function not loading group-dial-string and dial-string from user type pointer (MODAPP-335/r:16034)
|
||||
mod_commands: add timer_test cli app (r:16079)
|
||||
mod_commands: fsctl add shutdown now for debugging (r:16220)
|
||||
mod_conference: Fix conference floor ownership being ceded too easily (MODAPP-323/r:14703)
|
||||
mod_conference: Display callers' rates (r:14992)
|
||||
mod_conference: Don't start conf auto record until a second party arrives (MODAPP-348/r:15029)
|
||||
@ -271,6 +290,7 @@ freeswitch (1.0.5)
|
||||
mod_conference: fix video thread (r:15590)
|
||||
mod_conference: add mute on/mute off actions in addition to toggle mute action (MODAPP-370/r:15845)
|
||||
mod_conference: add conference member data into the play-file-member-done event (r:15909)
|
||||
mod_conference: add waste-bandwidth flag to conference (r:16110)
|
||||
mod_console: Improved tab completions, and more description usage informations (LOGGER-2/r:15103)
|
||||
mod_curl: don't include response code in response data as it has its own var (MODAPP-369/r:15591)
|
||||
mod_dialplan_xml: Fix anti-action not being supported for time-based conditions (DP-6/r:14901)
|
||||
@ -303,6 +323,7 @@ freeswitch (1.0.5)
|
||||
mod_event_socket: fix event_sink race (r:15389)
|
||||
mod_event_socket: fix expires seg in event_sink (r:15594)
|
||||
mod_event_socket: only kill event socket on 100 consecutive errors (r:15671)
|
||||
mod_event_socket: add userauth <user>@<domain>:<pass> to event_socket to auth against user directory uses esl-password esl-allowed-api esl-allowed-events and esl-allowed-log to control resource access (r:16160)
|
||||
mod_fifo: fix mod_fifo not honoring member_timeout (MODAPP-322/r:14552)
|
||||
mod_fifo: add fifo_position var (r:14806)
|
||||
mod_fifo: add API: fifo_add_outbound to add outbound members to a FIFO (r:14809)
|
||||
@ -353,6 +374,8 @@ freeswitch (1.0.5)
|
||||
mod_portaudio: add context param (r:14737)
|
||||
mod_portaudio: make the ringfile configurable via new API command 'pa ringfile' (r:15633)
|
||||
mod_portaudio: fix mod_portaudio linux build with alsa (MODAPP-377/r:15899)
|
||||
mod_portaudio: Set a new channel variable with the call_id (MODAPP-277/r:16115)
|
||||
mod_portaudio: fix pa play bug (r:16188)
|
||||
mod_python: Fix segfault on multiple calls to Python (MODLANG-133/r:14847)
|
||||
mod_python: Fix infinite recursion in python script causing crash (MODLANG-134/r:14898)
|
||||
mod_python: fix memory leak (MODLANG-136/r:15432)
|
||||
@ -460,12 +483,21 @@ freeswitch (1.0.5)
|
||||
mod_sofia: move NUTAG_MIN_SE to sofia_glue_do_invite so it only appears on requests. (MODSOFIA-47/r:15943)
|
||||
mod_sofia: Fix redirect contacts not set correctly on 300 Mutliple Choices (SFSIP-190/r:15994)
|
||||
mod_sofia: Add fix for silly endpoints who use G729a improperly (MODSOFIA-48/r:16004)
|
||||
mod_sofia: Add ping hysteresis for SIP peer qualification (MODSOFIA-40/r:16039)
|
||||
mod_sofia: Major presence update (r:16053)
|
||||
mod_sofia: Fix sofia loglevel 9 (MODSOFIA-50/r:16058)
|
||||
mod_sofia: don't segfault on notify message with no contact (r:16130)
|
||||
mod_sofia: honor disabled soa flag in re-invite (r:16193)
|
||||
mod_sofia: share and share alike, only nothing is alike in sip =/ (r:16194)
|
||||
mod_sofia: Get sofia "auth-calls" parameter and directory "auth-acl" parameter to work through a proxy. (BOUNTY-12/r:16200)
|
||||
mod_sofia: add command-line completion for sofia xmlstatus (MODSOFIA-52/r:16260)
|
||||
mod_spidermonkey: allow inline javascript, use a ~ as first script character (r:15598)
|
||||
mod_spidermonkey: fix mod_spidermonkey on OSX 10.6 (lets see if this breaks any other platforms) (r:15650)
|
||||
mod_spidermonkey_core_db: Allow to bind value to parameters in prepared statements (MODLANG-139/r:15632)
|
||||
mod_syslog: Enable mod_syslog to log to a specific facility (LOGGER-3/r:15162)
|
||||
mod_syslog: mod_syslog does not respect facility setting, always logs to user.* (LOGGER-4/r:15447)
|
||||
mod_syslog: fixed mod_syslog.c on solaris since there is no LOF_FTP and LOG_AUTHPRIV in solaris syslog (r:15606)
|
||||
mod_syslog: add uuid logging support (r:16187)
|
||||
mod_tts_commandline: Add new module (r:14827)
|
||||
mod_tts_commandline: introduce the rate parameter, decrease useless verbosity (r:14885)
|
||||
mod_tts_commandline: cut samples in half (suggested by anthm), adjust and clean log levels (r:14886)
|
||||
@ -497,7 +529,6 @@ freeswitch (1.0.5)
|
||||
mod_voicemail: Add new api call to get user mailbox settings (MODAPP-325/r:14982)
|
||||
mod_voicemail: Add ability to skip greeting and instructions when leaving voicemail (MODAPP-331/r:14990)
|
||||
mod_voicemail: Fix VM disk quota (MODAPP-353/r:15161)
|
||||
mod_voipcodecs: move mod_voipcodecs to use spandsp instead of libvoipcodecs (windows build to follow) (r:15036)
|
||||
mod_voicemail: add missing switch_event_destroy in profile config function (r:15262)
|
||||
mod_voicemail: default samplerate to 0 instead of 8000 so that we record using the channel's native rate, thx bkw (r:15394)
|
||||
mod_voicemail: prevent obscure divide by zero code path (r:15413)
|
||||
@ -506,7 +537,11 @@ freeswitch (1.0.5)
|
||||
mod_voicemail: Fix r15636 so New are New, and Old are Old messages (r:15639)
|
||||
mod_voicemail: mod_voicemail: Change so total_new_messages and total_saved_messages include the count of new/saved urgent messages (Issue caused by MODAPP-359) (r:15670)
|
||||
mod_voicemail: Fix segfault (FSCORE-511/r:15876)
|
||||
mod_voicemail: passing originator's account name to cc'd accounts or groups to let them see who forwarded it (BOUNTY-13/r:16270)
|
||||
mod_voipcodecs: move mod_voipcodecs to use spandsp instead of libvoipcodecs (windows build to follow) (r:15036)
|
||||
mod_voipcodecs: move mod_voipcodecs to use spandsp instead of libvoipcodecs - windows (r:15088)
|
||||
mod_voipcodecs: rearrange codecs so we don't have crazy overlap and remove G723-32 and move it up to dynamic cuz people have nothing better to do then write stupid RFC's (r:16251)
|
||||
mod_xml_cdr: Allow specifying auth-scheme in config (MDXMLINT-56/r:16247)
|
||||
mod_xml_curl: Fix crash when using use-dynamic-url (XML-10/r:14850)
|
||||
mod_xml_curl: Allow choice between HTTP Basic and Digest authentication (r:15107)
|
||||
mod_xml_curl: Don't use signals when a timeout is specified (XMLINT-13/r:15997)
|
||||
|
Loading…
Reference in New Issue
Block a user