+++ /dev/null
-freeswitch (1.0.7-1headgit1) maverick; urgency=low
- * upgrade: Added mod_amrwb
- * upgrade version to 1.0.7 ... 1.0.6 was last realease
-
- -- Michal Bielicki <michal.bielicki@seventhsignal.de> Wed, 13 Oct 2010 22:58:48 -0200
-freeswitch (1.0.6-1ubuntu1) maverick; urgency=low
-
- [ Gabriel Gunderson ]
- * upgrade: Added mod_callcenter and pulled out Python into its own
- package.
-
- [ Mathieu Parent ]
- * Updated Uploaders list
- * Updated Standards-Version to 3.9.1
-
- -- Mathieu Parent <sathieu@debian.org> Thu, 23 Sep 2010 15:34:00 +0200
-
-freeswitch (1.0.4-1ubuntu2) karmic; urgency=low
-
- * upgrade: Add more verbosity when building to make it easier to find build
- errors.
- * upgrade: Remove the requirement for EXACTLY automake1.9 and change it to
- need atleast automake 1.9
- * upgrade: Add the modules (directory, cluechoo, and valet_parking) to the
- build files. These are in the standard build, so they should be here too.
-
- -- William King <quentusrex@gmail.com> Fri, 18 Dec 2009 14:27:42 -0800
-
-freeswitch (1.0.4-1ubuntu1) karmic; urgency=low
-
- * upgrade: Pulling out the sounds into separate source files for easier management.
-
- -- William King <quentusrex@gmail.com> Sun, 15 Nov 2009 16:38:13 -0800
-
-freeswitch (1.0.4-1) unstable; urgency=low
-
- * new
-
- -- Mike Jerris <mike@jerris.com> Mon, 18 Feb 2009 17:39:00 -0500
-
-freeswitch (1.0.3-1) unstable; urgency=low
-
- * build: add targets cd-sounds[-install] and cd-moh[-install] for 48k sounds (r:11151)
- * build: autoconf detect odbc library (FSBUILD-8)
- * build: fix sound install on windows build (r:11635,11638)
- * build: fix configure --sysconfdir (FSBUILD-84)
- * build: fix uclibc build (MODLANG-99)
- * build: fix adduser in debian (FSBUILD-122, FSBUILD-102)
- * core: fix buffering issues (r:11101,11145,11152-11157,11162,11191,11200)
- * core: fix c leg no hangup when converting attended to blind transfer before b leg answers (MODENDP-165/r:11061)
- * core: fix codec and media handling issues (r:11104)
- * core: fix double close of file handles and add recording of native files (r:11108-11113,11482,11483)
- * core: fix file resampling issue (r:11090)
- * core: fix incorrect call progress timestamps in timetable (r:11186-11187/FSCORE-268)
- * core: fix media handling issues (r:11079-11082)
- * core: fix multiple 2833 dtmf handling issues (r:11149,11261,11262,11266,11293,11294,11338/FSCORE-266,FSCORE-273)
- * core: send more event types verbos bridge,unbridge,park,unpark (r:11097-11098)
- * core: Prevent media setup on failed originates (r:11462/FSCORE-279)
- * core: fix recorded soundfiles had random data at end of file (r:11491/MODAPP-205)
- * core: fix user for windows service (r:11538/FSCORE-277)
- * core: modify variable expansion code to expand in more places and to fix potential security issue from injecting variables (r:11569,11570)
- * core: look for soundfiles in more locations based on rate (r:11601/MODFORM-23)
- * core: state machine veto behavior changed (r:11610)
- * core: add enable_file_write_buffering variable (r:11677)
- * core: fix garbled audio on media bug during bridge using stateful codecs (FSCORE-288)
- * core: fix tone detect running multiple bugs when detecting multiple tones
- * core: add {instant_ringback=true} to make ringback not wait for indication to generate ringback
- * core: fix segfault from race condition on multiple reloadxml calls (MOODAPP-211)
- * core: modify xml locking so phrases do not lock the xml for the duration of playing them
- * core: replace resampler with the speexdsp resampler
- * core: fix windows calling convention on threads launched that return a value to fix shutdown segfault (FSCORE-298)
- * core: do not auto-export origination_caller_id_* to avoid confusion (r:12052)
- * core: API visibility support (GCC/SUNCC) (FSCORE-264)
- * core: fix leak in exposed event class serialize method (r:12068)
- * core: add volume as possible return value from input callback on embedded languages (r:12114)
- * core: add resampler to seech handles (r:12141)
- * core: add api.getTime to embedded languages (r:12149)
- * freeswitch: allow you to specify -htdocs dir at runtime. (r:11614)
- * fs_cli: add "debug" command to change the esl debug level at runtime (r:11057)
- * iksemel: update to 1.3 (r:11645)
- * libesl: fix disconnect failure (r:11078,11083)
- * libesl: fix solaris build (r:11067,11068)
- * libesl: add c++ wrapper and swigged wrappers for multiple scripting languages
- * libg722_1: fix dct4.h code generator to include the "f" (r:11188-11189,11367)
- * libilbc: update to new library from Steve Underwood
- * mod_amrwb: add amr wideband passthrough codec (r:11971)
- * mod_cepstral: fix failure return code handling (MODASRTTS-9)
- * mod_conference: add 'conference xml_list' and 'conference [conf_name] xml_list' (r:11062-11063)
- * mod_conference: make conference verbose-events param to control if events have all the channel data or not (r:11073-11077)
- * mod_conference: add MINTWO flag to end conference when down to 1 participant (r:11523)
- * mod_conference: refactor conference record function (r:11626)
- * mod_conference: add conference list summary command (MODAPP-197)
- * mod_conference: fix Deadlock or coredump on conference commands play, transfer (MODAPP-209)
- * mod_dahdi_codec: added (MODCODEC-7)
- * mod_dialplan_xml: make previous auto hunt feature optional and off by defaule use auto_hunt=true session or global variable to enable (r:12144)
- * mod_dptools: Add failure_causes channel variable (r:12058)
- * mod_easyroute: add configuration file example for custom-query (r:11055)
- * mod_easyroute: add custom-query configuration option (r:11054)
- * mod_easyroute: fix build error when not configured for odbc (r:11478)
- * mod_easyroute: fix memory leak (r:11611)
- * mod_erlang_event: add ability to spawn a process (module/function) outbound on a specified node. (r:11460,11477)
- * mod_erlang_event: Fix some issues with standing up a new outbound listener and cleaning up after a failed session (r:11479)
- * mod_erlang_event: Fix setting up a listener for an outbound session if one doesn't already exist (r:11488)
- * mod_erlang_event: add "erlang" fscli command (r:11488)
- * mod_erlang_event: Monitor spawned outbound processes for premature exits (r:11489)
- * mod_erlang_event: Allow the event encoding strategy to be configurable; choices are string or binary (r:11495)
- * mod_erlang_event: Allow certain tuple elements to be binaries or strings, to reduce conversion requirements on the erlang side (r:11496)
- * mod_erlang_event: Support sending a message to a registered process to request a pid (when spawning won't cut it) (r:11499)
- * mod_erlang_event: Ensure events received while a pid session is being created aren't discarded but are queued instead (r:11500)
- * mod_erlang_event: Add freeswitch.erl - An erlang module to make talking to mod_erlang_event more painless (r:11525)
- * mod_erlang_event: use rpc:call instead of spawn and to make the registered process argument to handlecall optional (r:11542)
- * mod_event_socket: add ability to use a comma sep list of events on event-sink create-listener (r:11056)
- * mod_event_socket: add debug logging to event-sink (r:11060)
- * mod_event_socket: fix race condition (r:11680,12146)
- * mod_dptools: add all modifier to break command (r:11557,11558)
- * mod_dptools: add sound_test application (r:11658)
- * mod_fax: Dont hangup after sending/receiving faxes
- * mod_fifo: pause media bugs while not in a bridge (r:11466,11490)
- * mod_fifo: allow unpark during chime list playing (r:11555/MODAPP-206)
- * mod_fifo: fix outbound fifos doesn't check if the consumer is in the fifo in question. (r:11561/MODAPP-207)
- * mod_fifo: Fix segfault when no argument were supplied to fifo_member call (MODAPP-210)
- * mod_lcr: added (r:11180,11184,11532,11609)
- * mod_limit: fix memory corruption caused by race condition when using limit hash (r:11070-11071)
- * mod_limit: Fix transfer bug, fix leak and make the channel hangup if the extension is \!hangup_cause (r:11604,11932)
- * mod_limit: add write different channel variables per realm_id (r:11608)
- * mod_limit: Make max argument optional on the limit app, set the limit_usage variable to current count after inserting call in the db (r:11955)
- * mod_lua: Create empty argv table when no args are passed to a Lua script (r:11559)
- * mod_lua: use dll for lua windows build (FSCORE-299)
- * mod_openmrcp: removed (r:11176-11179)
- * mod_opal: added
- * mod_pocketsphinx: fix leak (r:11974)
- * mod_portaudio: fix stuck channels on outbound calls (r:11160,11470,11471,11472,11475,11476,11485)
- * mod_python: fix build when site dir is not /usr/lib/python2.4 (r:12070)
- * mod_say_en: add short form date/time (MODAPP-180)
- * mod_sofia: add auto-rtp-bugs profile option to make rtp bug compensation configurable (r :11146-11147)
- * mod_sofia: add support in sdp for a=maxptime (r:11103)
- * mod_sofia: fix codec change race condition (r:11143)
- * mod_sofia: fix notify event wasn't allowing content-length 0 (r:11106/MODENDP-167)
- * mod_sofia: fix sending extra sdp in 200 OK when using 100rel in violation of sdp o/a protocol draft-ietf-sipping-sip-offeranswer-10 (r:11088)
- * mod_sofia: fix sip_auto_answer=true (r:11069)
- * mod_sofia: improve outbound registration error message (r:11059)
- * mod_sofia: reset media timeout on re-invite (r:11161)
- * mod_sofia: fix segfault due to missing contact header in invite (r:11463/MODENDP-177)
- * mod_sofia: allow <params> tag in gateways as well as <variables> with direction inbound/outbound (default both) and call counter (r:11468)
- * mod_sofia: add support or SLA, works with Polycom and Snom(Sylantro mode). (r:11562/MODENDP-179)
- * mod_sofia: tolerate missing user in the request uri (r:11636)
- * mod_sofia: Add purpose=gateways and profile=[name] so xml_curl requests make sense (MDXMLINT-46)
- * mod_sofia: Add disable-srv and disable-naptr params to sip profiles (default false) (MODENDP-183)
- * mod_sofia: add outbound-proxy param (MODENDP-184)
- * mod_sofia: fix segfault with stun-enabled=false (SFSIP-120)
- * mod_sofia: Profile Name in Expire Event is incorrect (MODENDP-185)
- * mod_sofia: add "scrooge" mode to "inbound-codec-negotiation" (r:11881)
- * mod_sofia: Add context to reconfig_sofia (r:12080)
- * mod_sofia: fix segfault when calling from a Cisco 7940 using bypass_media (FSCORE-301)
- * mod_sofia: ilbc to default to 30 if no mode= fmtp is defined on the inbound (r:12110)
- * mod_sofia: fix challenge-realm (r:12113)
- * mod_sofia: Segmentation fault when running killgw command on sofia profile without specifying a gateway (MODENDP-189)
- * mod_sofia: gateways will inherit the context from its parent unless manually provided (r:12138)
- * mod_sndfile: Add IMA ADPCM support (MODFORM-22)
- * mod_spidermonkey: fix loading of spidermonkey modules (r:11084-11085)
- * mod_spidermonkey: block some unwanted behaviours in session.originate
- * mod_spidermonkey_socket: fix gc blocking (MODLANG-97)
- * mod_xml_rpc: fixed authentication using @domain syntax (r:11064)
- * mod_xml_rpc: fix http content types sent in responses (r:11099,11148,11150)
- * mod_voicemail: voicemail insert into the proper fields (MODAPP-190)
- * mod_voipcodecs: add G.726 24k (r:12083)
- * sofia-sip: update to current sofia-sip repository
- * spandsp: sync to latest snapshot and fix windows build
- * speex: updated to 1.2rc1
- * sqlite: fix random assert on windows (FSCORE-292)
-
- -- Mike Jerris <mike@jerris.com> Mon, 18 Feb 2009 17:39:00 -0500
-
-freeswitch (1.0.2-1) unstable; urgency=low
-
- * all: don't add module interfaces before returning from error conditions in module load functions (MDXMLINT-36)
- * all: fixed multiple memory leaks
- * all: improved module unloading/reloading support
- * build: add support for --switchconfdir (FSBUILD-84)
- * build: fixed netbsd build
- * build: make freeswitch stop graceflly with /etc/init.d/freeswitch stop on debian add working dir to start-stop-dir so freeswitch dumps core in workdir
- * build: multiple packaging fixes
- * build: user freeswitch not added to audio group on deb install (FSBUILD-95)
- * Configuration: many updates to default configuration
- * core: Add ability to choose uuid from originate string, originate_uuid var (use at your own risk)
- * core: add bridge_generate_comfort_noise option for bridge to generate comfort noise to the A leg when there is no audio on the B leg
- * core: add chan vars to param event during hangup hook
- * core: add exec directive to xml preprocessor (not available on windows)
- * core: add force_transfer_dialplan and force_transfer_context variables
- * core: add hashing to event header lookup
- * core: add hits to tone_detect
- * core: add last_dtmf_duration variable
- * core: add msleep function to swigged languages
- * core: add park_after_bridge variable
- * core: add per leg timeouts and specific cause codes in reject_on_single_fail
- * core: add runtime selection of the module dir (FSCORE-198)
- * core: add scheduler support for heartbeat
- * core: add session heartbeat feature
- * core: add session.destroy psuedo method to sort of destroy a session at least for the sake of FS
- * core: add session.unsetInputCallback
- * core: add strftime format string validation for user supplied values
- * core: add vars param to switch_ivr_originate for recursion (MODAPP-175)
- * core: added a "group" concept to the user directory
- * core: added ability to do dns lookup to find ip with host: like stun: (FSCORE-219)
- * core: added better locking for codec changes during a call
- * core: added current_application and current_application_data variables
- * core: added error/ magic endpoint so modules can return error causes in situations like sofia_contact
- * core: added read_result channel variable
- * core: added support for "F" to indicate flash in dtmf (FSCORE-213)
- * core: allow calls to be stolen from originate
- * core: allow you to get the privacy bits in the caller_profile
- * core: change dso code to load symbols local
- * core: changes core flags to be array based so we have more
- * core: eavesdrop causes the people being eavesdropped on to not hear ach other (MODAPP-140)
- * core: expose time table to variable interface via caller field lookup
- * core: fix 100% cpu when sending parked call to moh (FSCORE-234)
- * core: fix bridge app to make sure both channels are ready for media when one is only in ringing state
- * core: fix buffer overflow (FSCORE-188)
- * core: fix conference dial by allowing multiple braces in originate, fix bad pointer op (FSCORE-208)
- * core: fix double detection of DTMF in IVR (FSCORE-221)
- * core: fix hangup_after_bridge is false on a bridge started with the intercept app
- * core: fix issue where pid file is accidentally truncated
- * core: fix ivr timeout (FSCORE-181)
- * core: fix memory leak in alias tab completion code
- * core: fix min digits in read app
- * core: fix out-of-bounds pointer in variable expansion (FSCORE-171)
- * core: fix segfault in media bugs when in bypass media (FSCORE-193)
- * core: fix segfault on gtalk to sip calls (FSCORE-212)
- * core: fix segfault on reloadxml (FSCORE-176)
- * core: fix segfault on trasfering eavesdopping party (FSCORE-210)
- * core: fix segfault using switch_system function (FSCORE-196)
- * core: fix session.bridge
- * core: fix setting effective_caller_id_name / effective_caller_id_number on bridge dialstring (MODAPP-122)
- * core: fix stream_raw_write (MODAPP-145)
- * core: fix using resampling on ringback file
- * core: fixed performance bottleneck in sqlite db's
- * core: fixed race in reloadxml
- * core: increment app before execute in case it returns to execute it will go to the next item in the list and not the same
- * core: ivr menu max_failures and max_timeouts now default to 3 if not specified or invalid (less than 1) values are specified (FSCORE-244)
- * core: ivr_menu max-timeouts option, result in ivr_menu_status var (FSCORE-183)
- * core: let b legs use park_after_bridge too
- * core: make events less verbose unless verbose_events is set
- * core: parse private events during originate
- * core: pass pdd data to a leg on oubound calls using bridge
- * core: prevent crash in crazy situation with xml interface lookup failures (FSCORE-169)
- * core: reduce cpu requirement for generated comfort noise
- * core: remove interface names from core db on unload
- * core: reworked timing resulting in significant performance increase and better rtp timing
- * core: rewrite switch_play_and_get_digits (MODAPP-166)
- * core: session.recordFile never terminates (MODLANG-79)
- * core: session.transfer make dialplan and context optional
- * core: set_user app now sets domain vars as well as user vars
- * core: tone_detect not triggering app after tone detection (MODAPP-182)
- * core: unprivileged user setting bigger stack for switch_system thread failure (FSCORE-197)
- * core: user_exists returns false when fetching a user from XML Curl or other xml interfaces
- * libesl: added c event socket library and fs_cli
- * libsndfile: fix autoconf 2.62 support (LBSNDF-5)
- * mod commands: add "all" modifier to "break" command
- * mod_celt: added new module
- * mod_commands: Add support for more than 2 variables to uuid_setvar_multi (MODAPP-171)
- * mod_commands: Add support for passing the cause of hangup to the uuid_kill command (FSCORE-217)
- * mod_commands: add attr lookup to user_data
- * mod_commands: add domain_exists fsapi command
- * mod_commands: add eval fsapi command
- * mod_commands: add flush_dtmf app and uuid_flush_dtmf api command
- * mod_commands: add fsctl send_sighup, fsctl shutdown asap, unsched_api commands
- * mod_commands: add fsctl shutdown [elegant|restart|cancel]
- * mod_commands: add new syntax to uuid_setvar to allow you to unset a var. <uuid> <var> [value] (MODAPP-167)
- * mod_commands: add reload fsapi command to reload a module
- * mod_commands: add system fsapi and application (MODAPP-138)
- * mod_commands: added hupall fsapi command
- * mod_commands: added strftime_tz api command
- * mod_commands: break all now stops broadcast too
- * mod_commands: fix api command sent through sched_api was getting the last char lopped off
- * mod_commands: fix race on transfer with -both
- * mod_commands: fix system dialplan app problems (MODAPP-86)
- * mod_commands: only send content-type on status when it really is http.
- * mod_conference: add fsapi to stop async playback too
- * mod_conference: add video caps to mod_conference with video follow audio
- * mod_conference: better sound prefix handling when using say: and allow say: on kick sounds.
- * mod_conference: fix race in record
- * mod_conference: fix runaway thread when floor holder has no video and other people do have video
- * mod_conference: fix seg when kicking many members quickly (MODAPP-129)
- * mod_conference: fix segfault on invalid chat event
- * mod_conference: perpetual sound does not auto-mute, you can do that yourself if you want it
- * mod_dialplan_xml: add Hunt- vars in dialplan lookup after transfer
- * mod_dialplan_xml: fail call on extensions with nested conditions
- * mod_dingaling: (LBDING-7) fix segfault on os x
- * mod_dingaling: end call on ice timeout
- * mod_dingaling: fix presence on jabber to be less protocol ambiguous
- * mod_dingaling: fix segfault (LBDING-10)
- * mod_dingaling: update to support latest client from google
- * mod_dptools: add a mechanism to tell if a file played from sendmsg over event socket
- * mod_dptools: add playback_terminator support to phrase and say app
- * mod_dptools: add playback_terminator_used variable (MODAPP-132)
- * mod_dptools: add presence application
- * mod_dptools: fix originate api not parsing users properly (FSCORE-246)
- * mod_dptools: fix record and record_session to create directory if it does not exist (FSCORE-250)
- * mod_dptools: fixed limit and + parsing bug in record_session app (MODAPP-148)
- * mod_dptools: remove_bugs added to remove all media bugs on a session
- * mod_erlang_event: add new module
- * mod_event_socket: missing : after Content-Length in event socket (MODEVENT-33)
- * mod_event_socket: add event socket listener filters
- * mod_event_socket: add stateful listener fsapi commands for ajax-y type event interface over http
- * mod_event_socket: fix arg parsing errors (MODEVENT-34)
- * mod_event_socket: fix shutdown segfault race (MODEVENT-32)
- * mod_event_socket: inbound connection to event_socket can now take over an existing channel with 'myevents <uuid>' to take on the behaviour of an outbound socket
- * mod_event_socket: let any channel get messages
- * mod_event_socket: make event socket wait for hangup on outbound mode and send disconnect message
- * mod_expr: fix endless loop
- * mod_fax: new module
- * mod_fifo: add fifo_consumer_wrapup_time var (MODAPP-117)
- * mod_fifo: added callback agents
- * mod_fifo: honor keyword silence (MODAPP-118)
- * mod_flite: added windows build
- * mod_fsv: fix in a windows enviroment opening the record file in text mode. (MODAPP-169)
- * mod_http: added new module
- * mod_java: updated to new module api to support read/write locks on interface
- * mod_limit: accept dialplan context for transfer (MODAPP-161)
- * mod_limit: added hashtable based limit functions
- * mod_limit: prevent empty error log message (MODAPP-134)
- * mod_local_stream: add start_local_stream and stop_local_stream fsapi commands to start/stop dynamically (MODFORM-13)
- * mod_local_stream: fix leak and improve error checking
- * mod_local_stream: fix seg when no timer name specified in config file. (MODFORM-16)
- * mod_loopback: add new module
- * mod_lua: add local scripts directory support (MODLANG-86)
- * mod_lua: don't eval blank string
- * mod_lua: fix originate
- * mod_lua: fix segfault (MODLANG-77)
- * mod_lua: update to lua 5.1.4 (MODLANG-87)
- * mod_lumenvox: removed
- * mod_managed: new module replaces mod_mono now supports native .net runtime on windows as well
- * mod_opal: added to trunk (still very beta)
- * mod_perl: fix segfault (MODLANG-77)
- * mod_pocketsphinx: fix rpm build
- * mod_portaudio: fix cpu race on inbound call to pa when no ring file is set
- * mod_radius_cdr: dictionary update for cause code changes (MODEVENT-27)
- * mod_radius_cdr: fix unload (MODEVENT-29)
- * mod_shout: add stereo recording broadcast support
- * mod_shout: added windows build
- * mod_shout: fix segfault when recording mp3's (MODFORM-12)
- * mod_shout: improved stability of mp3 decoding
- * mod_siren: added new module
- * mod_sndfile added support to record 16bit for the various rates including 48kHz
- * mod_sofia: Add filter to "sofia status profile XXX" (MODENDP-138)
- * mod_sofia: Add force-register-db-domain which works in conjunction with force-register-domain.
- * mod_sofia: Add optional <variables> and <params> tag to <gateway> tag.
- * mod_sofia: Challenge the right realm when to_host is outside the users domain. (MODENDP-136)
- * mod_sofia: Improve notify messages through a proxy (MODENDP-147)
- * mod_sofia: MWI for multiple domains (MODAPP-126)
- * mod_sofia: Move "a=sendrecv" from session to media section of SDP (MODENDP-148)
- * mod_sofia: add 200 OK re-invite without sdp
- * mod_sofia: add custom sofia::gateway_state event (MODENDP-112)
- * mod_sofia: add fire events for the refer SIP NOTIFY event package (MODENDP-152)
- * mod_sofia: add more params for xml_curl directory lookup
- * mod_sofia: add new auto vals for challenge-realm param <param name="challenge-realm" value="auto_from|auto_to|<hardcoded_val>"/>
- * mod_sofia: add option to turn of auto_restart of sofia profiles on ip change
- * mod_sofia: add params to use sip callid as uuid on inbound calls and uuid as sip callid on outbound calls
- * mod_sofia: add parsing of Privacy header for privacy info (MODENDP-133)
- * mod_sofia: add proto_specific_hangup_cause to both legs
- * mod_sofia: add proxy 3pcc mode
- * mod_sofia: add redirect variable to channel as well as partner channe (MODENDP-135)
- * mod_sofia: add sip-forbid-register to user params to refuse to let a certian user register
- * mod_sofia: add sip: into register-proxy when it's not specified
- * mod_sofia: add sip_history_info var for inbound invites.
- * mod_sofia: add sip_via_protocol variable
- * mod_sofia: add sofia xmlstatus (MODENDP-156)
- * mod_sofia: add support for params other than Replaces in Refer-To (MODENDP-143)
- * mod_sofia: add support for profiles sharing databases so that you can have a domain that uses multiple profiles for split dns type setups
- * mod_sofia: add support for refer transfer involving multiple machines
- * mod_sofia: add support to send a notify in the invite dialog by specifying the uuid of the call. (SFSIP-92)
- * mod_sofia: add suppress_from_cidname var to not have display name in from header (MODENDP-153)
- * mod_sofia: added sip_hangup_disposition variable
- * mod_sofia: allow send_message and notify events to send a message/notify without a body if needed.
- * mod_sofia: append -1 .. -N postfix after any X-headers as vars that have the same name
- * mod_sofia: cache auth_gateway_name in sofia for challenged bye
- * mod_sofia: cancel proxy or no-media mode if you purposely answer or pre_answer
- * mod_sofia: correct result code mapping for Unallocated Number (MODENDP-124)
- * mod_sofia: disable 100rel by default
- * mod_sofia: don't accept crypto in the RTP/AVP (MODENDP-126)
- * mod_sofia: don't put CN in sdp answer if it was not in the offer.
- * mod_sofia: fix Incorrect IP address shows up in SDP "o" field when multiple external IPs available and FS not bound to first (MODENDP-132)
- * mod_sofia: fix Wrong RTP media port destination after reinvite/UNHOLD (SFSIP-82)
- * mod_sofia: fix bug on linksys where they lie about the ptime and handle linksys transfer problem
- * mod_sofia: fix chat (send an IM) assumes that the user's profile is the same as their domain, which isn't necessarily so (SFSIP-83)
- * mod_sofia: fix dtmf handling of broken info dtmf endpoints
- * mod_sofia: fix eyebeam presence to be RFC compliant (MODENDP-144)
- * mod_sofia: fix ip change detection when in proxy mode
- * mod_sofia: fix register_proxy ignoring the paramaters (MODENDP-121)
- * mod_sofia: fix remote session refresh triggers request glare (MODENDP-131)
- * mod_sofia: fix rtp auto adjust running when it should not
- * mod_sofia: fix rtp sent to wrong port after some re-INVITE scenarios (MODENDP-141)
- * mod_sofia: fix sending of cn packets across bridge when we shouldn't
- * mod_sofia: fix sqlite issue with select of the sip contact
- * mod_sofia: fixed segfault on invalid presence payload
- * mod_sofia: gateway ping needs to look for 501 (SFSIP-78)
- * mod_sofia: handle multi contact register responses and register timeout better
- * mod_sofia: improve gateway resilience
- * mod_sofia: log ip and port you get reply to invite from
- * mod_sofia: make multiple-registations=true use the contact method and call-id option to do it the old way
- * mod_sofia: make proxy mode pull the port from m=image as well
- * mod_sofia: make register-proxy preserve the url composed from proxy but target the packets to desired address (MODENDP-121)
- * mod_sofia: many fixes for sonus rtp issues silence_when_idle=400 chanvar to send generated silence duing sleeps etc
- * mod_sofia: many fixes in presence handling
- * mod_sofia: passthrough t.38 fixes
- * mod_sofia: pick ipv4 or ipv6 based on sipip instead of having mixed in sdp
- * mod_sofia: send NOTIFY on TCP/UDP depending on the SUBSCRIBE (SFSIP-104)
- * mod_sofia: setting profile option multiple-registrations=contact key multi reg off the contact string
- * mod_sofia: wait for a reply on refer
- * mod_soundtouch: fixes and improvements, many options changed (MODAPP-149)
- * mod_soundtouch: updated to new module api
- * mod_spidermonkey: Segmentation fault in check_hangup_hook at mod_spidermonkey.c:1589 (MODLANG-74)
- * mod_spidermonkey: fix bug in apiExecute
- * mod_spidermonkey: fix memory pool handling and leaks
- * mod_spidermonkey: limit recursion busting through the stack (FSCORE-202)
- * mod_spidermonkey: make session.getVariable return blank string not the word false
- * mod_spidermonkey_curl: add optional content-type arg
- * mod_spidermonkey_odbc: fix numRows and add numCols
- * mod_spidermonkey_odbc: fix segfault (MODLANG-75)
- * mod_stress: new module for voice stress analysis
- * mod_syslog: don't log blank lines (FSCORE-163)
- * mod_tone_stream: let silence_stream://0 indicate perpetual silence
- * mod_vmd: add new module to detect voicemail "beep"
- * mod_voicemail: Add vm_alternate_greet_id param to directory entry (MODAPP-174)
- * mod_voicemail: Patch to add voicemail preference controlling date announcement new param 'play-date-announcement' to values 'first' 'last' or 'never' defaults to first to retain previous behavior (MODAPP-121)
- * mod_voicemail: Update mwi light after delete vm via web. (MODAPP-124)
- * mod_voicemail: add ability to get to options without listening to every saved message (MODAPP-115)
- * mod_voicemail: add ability to skip greeting when leaving a voicemail. (MODAPP-181)
- * mod_voicemail: add change-pass-key config file option
- * mod_voicemail: add forwarding support
- * mod_voicemail: add local dtmf driven alternat vm pass
- * mod_voicemail: add proper notification of a vm message being too short
- * mod_voicemail: add support for auth via a1-hash
- * mod_voicemail: add the "storage-dir" parameter to be set on a per-user basis (MODAPP-133)
- * mod_voicemail: add voicemail_greeting_path variable
- * mod_voicemail: added voicemail_alternate_greet_id variable
- * mod_voicemail: allow changing of password from voicemail to update user directory if using non-static config (MODAPP-156)
- * mod_voicemail: created email date (int overflow) (MODAPP-125)
- * mod_voicemail: don't try to deliver vm when no file was recorded. (MODAPP-133)
- * mod_voicemail: fix MWI with xml_curl used for directory (MODAPP-176)
- * mod_voicemail: fix Voicemail messages occasionally lost / stranded (MODAPP-178)
- * mod_voicemail: fix invalid event after message deleted (MODAPP-170)
- * mod_voicemail: fix mwi for phones with multiple registrations problem (MODAPP-153)
- * mod_voicemail: fix voicemail segfault on incorrect password (FSCORE-187)
- * mod_voicemail: fix voicemail_inject error handling (MODAPP-133)
- * mod_voicemail: fix voicemail_inject usage api call
- * mod_voicemail: improve error checking (MODAPP-142)
- * mod_voicemail: localize notification emails (MODAPP-139)
- * mod_voicemail: make more multi-domain friendly (MODAPP-162)
- * mod_voicemail: make playback created file macros optional (MODAPP-150)
- * mod_voicemail: recognize operator key in more places (MODAPP-159)
- * mod_voicemail: web interface displays incorrect created / last heard dates (MODAPP-123)
- * mod_wanpipe: removed
- * mod_xml_cdr: add https support
- * mod_xml_cdr: add optional a-leg prefix to xml cdr filenames (MDXMLINT-39)
- * mod_xml_cdr: add support for fallback webserver for cdr posting (FSCORE-238)
- * mod_xml_curl: Allow specification of HTTP method, and dynamic expansion of variables in URI. (MDXMLINT-41)
- * mod_xml_curl: added redirect following (max 10)
- * mod_xml_ldap: almost a complete rewrite of this module
- * mod_xml_rpc: allow setting of global realm without a global user (MDXMLINT-45)
- * mod_xml_rpc: fix multiple segfaults
- * mod_xml_rpc: fix segfault on originate via http
- * sofia-sip: updated to 1.12.10 (plus a few patches)
-
- -- Mike Jerris <mike@jerris.com> Mon, 29 Dec 2008 14:46:00 -0500
-
-freeswitch (1.0.1-1) unstable; urgency=low
-
- * FIX: prevent intercept race condition that can also be solved with continue_on_fail=originator_cancel
- * FIX: NULL dereference detected by klockwork (www.klockwork.com)
- * FIX: don't open failed local stream (MODFORM-9)
- * FIX: instability in mod_local_stream in failure scenarios
- * FIX: xmlrpc-c build on OS X 10.4 (FSBUILD-47)
- * ENHANCEMENT: Added tab completion on many api commands in console
- * ENHANCEMENT: polycom BLF support
- * FIX: many sip NAT related fixes in mod_sofia
- * FIX: support sip unregister with Contact: *
- * FIX: multiple segfaults in xmlrpc-c
- * FIX: sip unregister event being skipped
- * FIX: hangup properly on malformed sip 3pcc calls being used as a way to ping
- * ADD: enable-3pcc sofia profile param, it is now disabled by default.
- * ADD: presence events to sip proxy mode
- * ADD: legs param to cdr_csv
- * ADD: support for perl as an embedded lanugage
- * ENHANCEMENT: many new api's and functions to the embedded languages including api support, xml interface support, auto start scripts, and many new objects
- * CHANGE: python embedded language api changed to match perl, lua, java
- * FIX: many stability fixes in embedded langauges perl, lua, java, python
- * ADD: failed_xml_cdr magic channel variable
- * FIX: access free memory error in mod_sofia when using respond app
- * ENHNACEMENT: make global_setvar only have 2 fields so you can set foo=bar=blah w/o quotes
- * FIX: mod_spidermonkey keep hangup hook in the session thread
- * ENHANCEMENT: mod_ldap added sasl support and search filters
- * ADD: answered, waitForAnswer and mediaReady methods to embedded language Session object
- * ENHANCEMENT: mod_voicemail param change to allow notification emails using templates
- * ADD: per user acl in sofia
- * FIX: deadlock in mod_portaudio
- * ENHANCEMENT: blank username in sip will trigger a lookup for the user "nobody"
- * ADD: import variable to import variables from a peer channel at time of originate
- * FIX: api type fix for c++ modules when incorrectly using enums
- * FIX: eliminate need for escaped , in [] on originate
- * ADD: NDLB-force-rport option to force behavior as if rport was sent in the via
- * ENHANCEMENT: honor execute_on_answer on outbound legs too
- * ADD: execute_on_ring variable
- * FIX: Seg fault in CoreSession() class destructor
- * ADD: per channel caller id in originate
- * ADD: sip_outgoing_call_id variable
- * FIX: multiple memory leaks in mod_sofia
- * FIX: find_local_ip IPv6 support
- * ADD: variable expansion to on execute vars.(FSCORE-114)
- * ADD: count optional arg to show calls and show channels (MODAPP-103)
- * FIX: MODEVENT-25 (WSAWOULDBLOCK error on socket send in windows) in event socket
- * FIX: multiple fixes to the logic in mod_say_zh
- * ADD: inter digit timeout to swigged embedded languages getDigits method. (MODLANG-65)
- * ADD: Linksys P-RTP-Stat SIP header values (SFSIP-66)
- * FIX: small leak in core
- * ADD: progress_timeout var to originate
- * UPDATE: portaudio library
- * FIX: added timeout to iax read
- * ADD: 'pa rescan' to portaudio to look for new devices
- * FIX: wait for broadcast to start when starting async hold to avoid race
- * FIX: mod_rss, don't always play the first news feed
- * FIX: mod_rss inverval to use the session inteval (audio problems on 30ms channels)
- * ADD: Path: support in mod_sofia on register
- * FIX: mod_shout record stream
- * ENHANCEMENT: mod_voicemail support for effective_caller_id_name/number
- * ADD: url encode/decode api calls
- * FIX: "nua()" in debug information in sofia instead of the real function name
- * FIX: better handling of sips: uris
- * FIX: don't seg when using more than SWITCH_MAX_CODECS and bump SWITCH_MAX_CODECS to 50 (we have more than 30 in tree) (MODFORM-10)
- * ADD: mod_yaml
- * FIX: segfault on freeswitch startup if installed directories are removed
- * FIX: segfault when intercept with inbound_late_negotiation=true set
- * FIX: dont flood logs with eavesdrop messages (MODAPP-101)
- * FIX: don't destroy a codec that has not been created (MODAPP-101)
- * ENHANCEMENT: allows the "eavesdrop_group" variable to contain several groups, comma separated. (MODAPP-101)
- * FIX: cross compile (FSBUILD-53)
- * FIX: add header that Nuaunce considers mandatory (MODASRTTS-5)
- * ADD: write locks to the core and a function to unregister event bindings (adds better ability to unload modules)
- * ENHANCEMENT: make modules unbind events and un-reserve subclasses on module unload
- * ADD: removable xml hook bindings
- * ADD: EventConsumer object to embedded languages so you can make event handlers
- * FIX: sending CN with supress-cng true
- * FIX: segfault in the event system when trying to remove NULL event
- * ADD: flags to turn off srtp auth and rtp auto adj (FSCORE-149 && MODENDP-115)
- * FIX: use lighter math and avoid infinite loop in port allocator (FSCORE-148)
- * ENHANCEMENT: let conference pin entry start during prompt (MODAPP-111)
- * ADD: mod_pocketsphinx
- * FIX: Misuse of SQLRowCount, issues with MSSQL (MODAPP-105)
- * FIX: segfaults in mod_python with dtmf callback
- * ENHANCEMENT: mod_conference auto-record parameter (MODAPP-112)
- * ENHANCEMENT: reload support to many modules
- * FIX: mod_sofia add replaces to supported header
- * ENHANCEMENT: add args callback to sleep so you can process dtmf and events while "sleeping"
- * ADD: mod_flite
- * ENHANCEMENT: switch_xml converted back to c code and support double globs on windows
- * ENHANCEMENT: mod_sofia support for adding and removing gateways without restarting profiles
- * ADD: extract contact header info into A channel when unhandled 3xx response is received (MODENDP-116)
- * FIX: outbound event_socket + late negotiation
- * ADD: copy_xml_cdr variable
- * ADD: silence_stream (like tone_stream but silent)
- * ADD: module_exists api call
- * ADD: emailer implementation for windows
- * ADD: wait_for_silence application
- * FIX: no error message generated if OS is unable to load a module ( due to dependency/installation issues )
- * FIX: segfault in media bugs
- * FIX: acl lists not correctly matching all ip adresses
- * FIX: mod_spidermonkey exit() does not stop script when called from the hangup callback (return "exit" from the callback)
- * FIX: mod_syslog works again
- * FIX: crash on terminal resize
- * FIX: audio problems on big endian
- * ENHANCEMENT: Disable multiple registrations on a per-device basis (MODENDP-117)
- * ADD: fifo_consumer_exit_key variable (MODAPP-100)
- * ADD: cidr based user auth in mod_sofia
- * ADD: uuid_send_dtmf fsapi command (MODAPP-114)
- * ADD: server registration fiels to sip_registration database (MODENDP-118)
- * FIX: use a variable, realm or to host to find gateway when it's not obvious (handles challenged REFER)
- * ADD: timeout to curl run in javascript
- * ADD: voicemail_inject fsapi command
- * ADD: reboot option for sip phones to flush_inboud_reg sofia profile api command
- * FIX: add small padding to end of mp3 to avoid cut off mp3 recording
- * FIX: patch multiple SDP connection lines in sdp for proxy media mode (MODENDP-109)
- * FIX: don't parse ringback varable in proxy situations
- * ADD: per call vm recording ext with vm_message_ext variable
- * ADD: sip_bye_h prefix to add headers to bye
- * ENHANCEMENT: more interfaces available in show fsapi command
- * FIX: don't leak in buffers on realloc fail
- * FIX: fail out of a conference call if write fails
- * ADD: auto ip-change detection
- * ADD: mod_snom
- * FIX: mod_sofia don't send sipfrag on transfer to cisco so they don't hang up the call
-
- -- Mike Jerris <mike@jerris.com> Thu, 24 Jul 2008 07:00:00 -0500
-
-freeswitch (1.0.1~trunk) unstable; urgency=low
-
- * Updated revision number
- * Fixed init problem reported by Jay Binks (FSSCRIPTS-1)
- * Added a patch to the debian build system add more features (thanks to Hadley Rich) (FSBUILD-45)
- - Added en-us-callie sounds and music on hold packages
- - Added recommends and suggests
- - Added mod_say_es and mod_say_nl
- - Updated descriptions
- - Added mod_cdr_csv
- * Fixed typos and some errors in the previus patch.
- * Modified monit script. Now it should work.
- * The debian build system now bootstrap automagically if it's necessary and all scripts are in place.
-
- -- Massimo Cetra <devel@navynet.it> Sun, 6 Jul 2008 16:30:00 +0100
-
-freeswitch (1.0.0-1) unstable; urgency=low
-
- * Enhanced sofia sip nat handling
- * Many fixes found by Klockwork (www.klocwork.com)
- * Added disable_app_log variable
- * Fixed mod_local_stream with rates on windows
- * Fixed finding of files in rate dirs on windows
- * Fixed memory corruption from sofia_contact function
- * Added sofia profile param NDLB-received-in-nat-reg-contact
- * Added sofia profile param aggressive-nat-detection
- * Fixed video sip calls in proxy media mode
- * Added bridge_terminate_key var
- * Update xmlrpc-c lib to trunk revision from upstream, fix windows xmlrpc
- * Enhanced nat handling in proxy media mode in sip
- * Add progress media to timetable so you can calculate pdd
- * Fixed seg when using unicast on socket when call has no read_codec
- * Fixed missed log events on busy box
- * Added -bleg to intercept
- * Enhance configure detection of python
- * Fixed build on solaris and freebsd for several modules
- * Added param "vm-email-only" to make voicemail sent by email only (previously default behavior)
- * Added param "vm-mailto-notify" to allow sending a notification email
- * Fixed mod_java build
- * Fixed mwi failures for some devices that don't subscribe
- * Removed fsapi functions (killchan, transfer, session_displace, reject)
- * Removed fsapi functions (session_record, broadcast, hold, media)
- * Many updates to sofia-sip library including over 100 fixes
-
- -- Michael Jerris <mike@jerris.com> Tue, 27 May 2008 01:30:00 -0400
-
-freeswitch (1.0~rc6-1) unstable; urgency=low
-
- * Changed to not allow pass_2833 on transcoded calls
- (it never worked, now it will tell you)
- * Enhanced sofia sip nat handling
- * Fix libedit build on solaris
- * Fix session timers in mod_sofia
- * Fix conference fire-call
- * Change: add var_event down into the endpoints so chans
- with no parents can still pass options
- * Added enable-post-var param to xml_rpc
- * Fix mod_lua build on solaris
- * Many fixes found by Klockwork (www.klocwork.com)
- * Add unregister event in mod_sofia
- * Enhance python configure detection
- * Add vm_boxcount api func
- * Fixed att_xfer issue
- * Fix sip now includes the Allow-Events header in more places
-
- -- Michael Jerris <mike@jerris.com> Tue, 13 May 2008 02:01:00 -0400
-
-freeswitch (1.0~rc5-1) unstable; urgency=low
-
- * Changed internal state names to avoid confusion
- Fixed video negotiation
- Enhanced accuracy of windows timer
- Fixed mod_ldap build
- Added dialplan and context to sql table for channels
- Multiple fixes to mod_lua and mod_perl
- Fixed logic bug in fifo causing segfault
- internal changes to sip stack so we can remove a hash redundant to the stack
- Fixed multiple memory leaks in mod_sofia
- Fixed event fetch segfault on sip subscribe
- Fixed segfault on timer rollover in sofia on 64bit
- Fixed audio timing issues in mod_portaudio
- Changed names of sip profiles in default config to avoid confusion
- Fixed memory usage leak-like behavior when playing files requiring resampling
- Removed some unused api's
- Fix rtp timeout when playing moh
- Removed some un-needed libraries and files from tree
- Fixed multiple issues in sip stack including multiple segfaults
- Added support for sip transfers on bypass_media and proxy_media calls
- Added say application
- Fixed --disable-debug configure option
- Enhanced switch_cpp wrapper (and perl, python, lua, java)
- Fixed segfault on inavalid stun response
- Fixed configure help output
- Fixed segfault on mp3 playback
- Fixed assert on invalid sdp (missing m= line)
- Added configurable windows service name
- Fixed proxy mode call transition to non proxy call
- Fixed solaris build of voipcodecs
- Fixed sofia seg when call failure edge case
-
- -- Michael Jerris <mike@jerris.com> Tue, 13 May 2008 02:01:00 -0400
-
-freeswitch (1.0~8327) unstable; urgency=low
-
- * Adding perl and lua separate packages
- * Adding mod_voipcodecs
-
- -- root <root@fs.navynet.it> Tue, 6 May 2008 09:46:26 +0000
-
-freeswitch (1.0~rc4-1) unstable; urgency=low
- * Add tab completion in cli
- Add "inline" dialplan
- Fixed segfault in enum
- Enhance enum to fork dial equal priority entries
- Added auto-reload to enum
- Fixed odbc bug is mod_sofia presence handling
- Add presence for conference and dial an eavesdrop
- Fix stack overflow segfault when recursively parking calls
- Fixed race is sofia registration handling
- Enhance sofia registration, unregister on keep-alive OPTIONS failure
- Added internal routing loop detection/avoidance
- Fixed race in bgapi in event socket
- Fixed vars to execute apps before bridge "bridge_pre_execute_aleg_app" and "bridge_pre_execute_bleg_app"
- Fixed re-setting sound prefix to no prefix after a pharse
- Enhanced setting of bracket vars from originate so they show in the CHANNEL_ORIGINATE event
- Add "enable-timer" and "enable-100rel" options to turn off default behaviors in sofia
- Add originate_timeout to originate vars
- Fixed hanging channels in mod_portaudio
- Added auto time sync on vps migration to different hardware
- Fixed seg on transfer when both legs are not sip
- Added configurable dtmf duration defaults
- Enhanced voicemail, allow interruption of hello message
- Fixed voicemail to not light up light on saved messages
- Enhance mod_amr honor disable dtx in fmtp (MODCODEC-3)
- Fixed bootstrap to install automake dependencies so you can use tarball without same version of automake installed
- Fixed MODLANG-56 (bad audio on originate and javascript streamFile)
- Added hold/unhold dialplan apps
- Enhanced sofia error checking to outlaw 0.0.0.0 in sofia ip params
- Backport fixes from sofia-sip tree
- Fixed MSVC build
- Fixed segfault on sip SUBSCRIBE with Expires: 0
- Added mod_say_zh
- Added --with-pyton and --with-pyton-config configure options
- Added mod_lua
- Enhanced switch_cpp wrapper in core and normalized interfaces for perl, python, lua, and java
- Fixed multiple issues in cpp wrapper and the languages perl, python, lua and java
- Added back mod_perl
- Added sofia gateway option ping to adjust options ping frequency
- Added .net event socket lib to contrib
- Fixed passing of exact response codes of sip across a bridge
- Added mod_reference, reference endpoint module
- Enhanced build so you can now make commented out modules using "make mod_name"
-
- -- Michael Jerris <mike@jerris.com> Wed, 23 Apr 2008 12:58:00 -0400
-
-freeswitch (1.0~rc3-1) unstable; urgency=low
- * Enhance xml menu system
- fixes upstream from sofia-sip library
- Enhance mod_fifo
- added close method to ODBC spidermonkey class
- Fix multiple bugs in the cpp wrapper used in mod_java and mod_python
- Fix hung sip channel issue using respond app or on re-invite with bypass media after 1xx or 2xx responses
-
- -- Michael Jerris <mike@jerris.com> Wed, 9 Apr 2008 12:58:22 -0400
-
-freeswitch (1.0~rc2-1) unstable; urgency=low
- * Fixed speex protocol negotiation issues (8k vs 16k)
- Fixed mod_iax race conditions
- Fixed ptime negotiation issues when re-packetizing
- Added ip based acl lists
- *
- -- Michael Jerris <mike@jerris.com> Wed, 9 Apr 2008 12:58:22 -0400
-
-freeswitch (1.0~rc1-1) unstable; urgency=low
- * loads of fixes
- new cdr-csv module
- new spidermonkey-curl module
-
- -- Michal Bielicki <michal.bielicki@voiceworks.pl> Mon, 14 Jan 2008 23:37:04 +0100
-
-freeswitch (1.0~beta3-1) unstable; urgency=low
-
- * Additional scripts for changing the user to freeswitch
- Added Startup Scripts
- Monit integration
- Settings file for integration into init
- init.d file
- added user freeswitch to own and run all off freeswitch
- cleaned up config file control
- new upstream release
- split off codec pakcages
- split off spidermonkey packages
-
- -- Michal Bielicki <michal.bielicki@voiceworks.pl> Tue, 27 Nov 2007 13:20:21 +0100
-
-freeswitch (1.0~beta2-1) unstable; urgency=low
-
- * New upstream release
-
- -- Paul van Genderen <paulvg@member.fsf.org> Wed, 17 Oct 2007 19:32:09 +0200
-
-freeswitch (1.0~beta1-1) unstable; urgency=low
-
- * New packages.
-
- -- Robert McQueen <robot101@debian.org> Sun, 12 Nov 2006 17:32:23 -0500
projects / thirdparty / freeswitch.git / commitdiff
